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FreeBSD/Linux Kernel Cross Reference
sys/dev/audio.c
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1 /* $NetBSD: audio.c,v 1.243.6.4 2010/12/21 22:25:56 riz 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 1.243.6.4 2010/12/21 22:25:56 riz 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 #include <sys/intr.h> 85 86 #include <dev/audio_if.h> 87 #include <dev/audiovar.h> 88 89 #include <machine/endian.h> 90 91 /* #define AUDIO_DEBUG 1 */ 92 #ifdef AUDIO_DEBUG 93 #define DPRINTF(x) if (audiodebug) printf x 94 #define DPRINTFN(n,x) if (audiodebug>(n)) printf x 95 int audiodebug = AUDIO_DEBUG; 96 #else 97 #define DPRINTF(x) 98 #define DPRINTFN(n,x) 99 #endif 100 101 #define ROUNDSIZE(x) x &= -16 /* round to nice boundary */ 102 #define SPECIFIED(x) (x != ~0) 103 #define SPECIFIED_CH(x) (x != (u_char)~0) 104 105 /* #define AUDIO_PM_IDLE */ 106 #ifdef AUDIO_PM_IDLE 107 int audio_idle_timeout = 30; 108 #endif 109 110 int audio_blk_ms = AUDIO_BLK_MS; 111 112 int audiosetinfo(struct audio_softc *, struct audio_info *); 113 int audiogetinfo(struct audio_softc *, struct audio_info *, int); 114 115 int audio_open(dev_t, struct audio_softc *, int, int, struct lwp *); 116 int audio_close(struct audio_softc *, int, int, struct lwp *); 117 int audio_read(struct audio_softc *, struct uio *, int); 118 int audio_write(struct audio_softc *, struct uio *, int); 119 int audio_ioctl(struct audio_softc *, u_long, void *, int, struct lwp *); 120 int audio_poll(struct audio_softc *, int, struct lwp *); 121 int audio_kqfilter(struct audio_softc *, struct knote *); 122 paddr_t audio_mmap(struct audio_softc *, off_t, int); 123 124 int mixer_open(dev_t, struct audio_softc *, int, int, struct lwp *); 125 int mixer_close(struct audio_softc *, int, int, struct lwp *); 126 int mixer_ioctl(struct audio_softc *, u_long, void *, int, struct lwp *); 127 static void mixer_remove(struct audio_softc *, struct lwp *); 128 static void mixer_signal(struct audio_softc *); 129 130 void audio_init_record(struct audio_softc *); 131 void audio_init_play(struct audio_softc *); 132 int audiostartr(struct audio_softc *); 133 int audiostartp(struct audio_softc *); 134 void audio_rint(void *); 135 void audio_pint(void *); 136 int audio_check_params(struct audio_params *); 137 138 void audio_calc_blksize(struct audio_softc *, int); 139 void audio_fill_silence(struct audio_params *, uint8_t *, int); 140 int audio_silence_copyout(struct audio_softc *, int, struct uio *); 141 142 void audio_init_ringbuffer(struct audio_softc *, 143 struct audio_ringbuffer *, int); 144 int audio_initbufs(struct audio_softc *); 145 void audio_calcwater(struct audio_softc *); 146 static inline int audio_sleep_timo(int *, const char *, int); 147 static inline int audio_sleep(int *, const char *); 148 static inline void audio_wakeup(int *); 149 int audio_drain(struct audio_softc *); 150 void audio_clear(struct audio_softc *); 151 static inline void audio_pint_silence 152 (struct audio_softc *, struct audio_ringbuffer *, uint8_t *, int); 153 154 int audio_alloc_ring 155 (struct audio_softc *, struct audio_ringbuffer *, int, size_t); 156 void audio_free_ring(struct audio_softc *, struct audio_ringbuffer *); 157 static int audio_setup_pfilters(struct audio_softc *, const audio_params_t *, 158 stream_filter_list_t *); 159 static int audio_setup_rfilters(struct audio_softc *, const audio_params_t *, 160 stream_filter_list_t *); 161 static void audio_destruct_pfilters(struct audio_softc *); 162 static void audio_destruct_rfilters(struct audio_softc *); 163 static void audio_stream_dtor(audio_stream_t *); 164 static int audio_stream_ctor(audio_stream_t *, const audio_params_t *, int); 165 static void stream_filter_list_append 166 (stream_filter_list_t *, stream_filter_factory_t, 167 const audio_params_t *); 168 static void stream_filter_list_prepend 169 (stream_filter_list_t *, stream_filter_factory_t, 170 const audio_params_t *); 171 static void stream_filter_list_set 172 (stream_filter_list_t *, int, stream_filter_factory_t, 173 const audio_params_t *); 174 int audio_set_defaults(struct audio_softc *, u_int); 175 176 int audioprobe(device_t, cfdata_t, void *); 177 void audioattach(device_t, device_t, void *); 178 int audiodetach(device_t, int); 179 int audioactivate(device_t, enum devact); 180 181 #ifdef AUDIO_PM_IDLE 182 static void audio_idle(void *); 183 static void audio_activity(device_t, devactive_t); 184 #endif 185 186 static bool audio_suspend(device_t dv PMF_FN_PROTO); 187 static bool audio_resume(device_t dv PMF_FN_PROTO); 188 static void audio_volume_down(device_t); 189 static void audio_volume_up(device_t); 190 static void audio_volume_toggle(device_t); 191 192 static void audio_mixer_capture(struct audio_softc *); 193 static void audio_mixer_restore(struct audio_softc *); 194 195 static int audio_get_props(struct audio_softc *); 196 static bool audio_can_playback(struct audio_softc *); 197 static bool audio_can_capture(struct audio_softc *); 198 199 static void audio_softintr_rd(void *); 200 static void audio_softintr_wr(void *); 201 202 struct portname { 203 const char *name; 204 int mask; 205 }; 206 static const struct portname itable[] = { 207 { AudioNmicrophone, AUDIO_MICROPHONE }, 208 { AudioNline, AUDIO_LINE_IN }, 209 { AudioNcd, AUDIO_CD }, 210 { 0, 0 } 211 }; 212 static const struct portname otable[] = { 213 { AudioNspeaker, AUDIO_SPEAKER }, 214 { AudioNheadphone, AUDIO_HEADPHONE }, 215 { AudioNline, AUDIO_LINE_OUT }, 216 { 0, 0 } 217 }; 218 void au_setup_ports(struct audio_softc *, struct au_mixer_ports *, 219 mixer_devinfo_t *, const struct portname *); 220 int au_set_gain(struct audio_softc *, struct au_mixer_ports *, 221 int, int); 222 void au_get_gain(struct audio_softc *, struct au_mixer_ports *, 223 u_int *, u_char *); 224 int au_set_port(struct audio_softc *, struct au_mixer_ports *, 225 u_int); 226 int au_get_port(struct audio_softc *, struct au_mixer_ports *); 227 int au_get_lr_value(struct audio_softc *, mixer_ctrl_t *, 228 int *, int *); 229 int au_set_lr_value(struct audio_softc *, mixer_ctrl_t *, 230 int, int); 231 int au_portof(struct audio_softc *, char *, int); 232 233 typedef struct uio_fetcher { 234 stream_fetcher_t base; 235 struct uio *uio; 236 int usedhigh; 237 int last_used; 238 } uio_fetcher_t; 239 240 static void uio_fetcher_ctor(uio_fetcher_t *, struct uio *, int); 241 static int uio_fetcher_fetch_to(stream_fetcher_t *, 242 audio_stream_t *, int); 243 static int null_fetcher_fetch_to(stream_fetcher_t *, 244 audio_stream_t *, int); 245 246 dev_type_open(audioopen); 247 dev_type_close(audioclose); 248 dev_type_read(audioread); 249 dev_type_write(audiowrite); 250 dev_type_ioctl(audioioctl); 251 dev_type_poll(audiopoll); 252 dev_type_mmap(audiommap); 253 dev_type_kqfilter(audiokqfilter); 254 255 const struct cdevsw audio_cdevsw = { 256 audioopen, audioclose, audioread, audiowrite, audioioctl, 257 nostop, notty, audiopoll, audiommap, audiokqfilter, D_OTHER 258 }; 259 260 /* The default audio mode: 8 kHz mono mu-law */ 261 const struct audio_params audio_default = { 262 .sample_rate = 8000, 263 .encoding = AUDIO_ENCODING_ULAW, 264 .precision = 8, 265 .validbits = 8, 266 .channels = 1, 267 }; 268 269 CFATTACH_DECL_NEW(audio, sizeof(struct audio_softc), 270 audioprobe, audioattach, audiodetach, audioactivate); 271 272 extern struct cfdriver audio_cd; 273 274 int 275 audioprobe(device_t parent, cfdata_t match, void *aux) 276 { 277 struct audio_attach_args *sa; 278 279 sa = aux; 280 DPRINTF(("audioprobe: type=%d sa=%p hw=%p\n", 281 sa->type, sa, sa->hwif)); 282 return (sa->type == AUDIODEV_TYPE_AUDIO) ? 1 : 0; 283 } 284 285 void 286 audioattach(device_t parent, device_t self, void *aux) 287 { 288 struct audio_softc *sc; 289 struct audio_attach_args *sa; 290 const struct audio_hw_if *hwp; 291 void *hdlp; 292 int error; 293 mixer_devinfo_t mi; 294 int iclass, mclass, oclass, rclass, props; 295 int record_master_found, record_source_found; 296 297 sc = device_private(self); 298 sc->dev = self; 299 sa = aux; 300 hwp = sa->hwif; 301 hdlp = sa->hdl; 302 #ifdef DIAGNOSTIC 303 if (hwp == 0 || 304 hwp->query_encoding == 0 || 305 hwp->set_params == 0 || 306 (hwp->start_output == 0 && hwp->trigger_output == 0) || 307 (hwp->start_input == 0 && hwp->trigger_input == 0) || 308 hwp->halt_output == 0 || 309 hwp->halt_input == 0 || 310 hwp->getdev == 0 || 311 hwp->set_port == 0 || 312 hwp->get_port == 0 || 313 hwp->query_devinfo == 0 || 314 hwp->get_props == 0) { 315 printf(": missing method\n"); 316 sc->hw_if = 0; 317 return; 318 } 319 #endif 320 321 sc->hw_if = hwp; 322 sc->hw_hdl = hdlp; 323 sc->sc_dev = parent; 324 sc->sc_opencnt = 0; 325 sc->sc_writing = sc->sc_waitcomp = 0; 326 sc->sc_lastinfovalid = false; 327 328 props = audio_get_props(sc); 329 330 if (props & AUDIO_PROP_FULLDUPLEX) 331 aprint_normal(": full duplex"); 332 else 333 aprint_normal(": half duplex"); 334 335 if (props & AUDIO_PROP_PLAYBACK) 336 aprint_normal(", playback"); 337 if (props & AUDIO_PROP_CAPTURE) 338 aprint_normal(", capture"); 339 if (props & AUDIO_PROP_MMAP) 340 aprint_normal(", mmap"); 341 if (props & AUDIO_PROP_INDEPENDENT) 342 aprint_normal(", independent"); 343 344 aprint_naive("\n"); 345 aprint_normal("\n"); 346 347 if (audio_can_playback(sc)) { 348 error = audio_alloc_ring(sc, &sc->sc_pr, 349 AUMODE_PLAY, AU_RING_SIZE); 350 if (error) { 351 sc->hw_if = NULL; 352 aprint_error("audio: could not allocate play buffer\n"); 353 return; 354 } 355 } 356 if (audio_can_capture(sc)) { 357 error = audio_alloc_ring(sc, &sc->sc_rr, 358 AUMODE_RECORD, AU_RING_SIZE); 359 if (error) { 360 if (sc->sc_pr.s.start != 0) 361 audio_free_ring(sc, &sc->sc_pr); 362 sc->hw_if = NULL; 363 aprint_error("audio: could not allocate record buffer\n"); 364 return; 365 } 366 } 367 368 sc->sc_lastgain = 128; 369 370 if ((error = audio_set_defaults(sc, 0))) { 371 aprint_error("audioattach: audio_set_defaults() failed\n"); 372 sc->hw_if = NULL; 373 return; 374 } 375 376 sc->sc_sih_rd = softint_establish(SOFTINT_SERIAL, 377 audio_softintr_rd, sc); 378 sc->sc_sih_wr = softint_establish(SOFTINT_SERIAL, 379 audio_softintr_wr, sc); 380 381 iclass = mclass = oclass = rclass = -1; 382 sc->sc_inports.index = -1; 383 sc->sc_inports.master = -1; 384 sc->sc_inports.nports = 0; 385 sc->sc_inports.isenum = false; 386 sc->sc_inports.allports = 0; 387 sc->sc_inports.isdual = false; 388 sc->sc_inports.mixerout = -1; 389 sc->sc_inports.cur_port = -1; 390 sc->sc_outports.index = -1; 391 sc->sc_outports.master = -1; 392 sc->sc_outports.nports = 0; 393 sc->sc_outports.isenum = false; 394 sc->sc_outports.allports = 0; 395 sc->sc_outports.isdual = false; 396 sc->sc_outports.mixerout = -1; 397 sc->sc_outports.cur_port = -1; 398 sc->sc_monitor_port = -1; 399 /* 400 * Read through the underlying driver's list, picking out the class 401 * names from the mixer descriptions. We'll need them to decode the 402 * mixer descriptions on the next pass through the loop. 403 */ 404 for(mi.index = 0; ; mi.index++) { 405 if (hwp->query_devinfo(hdlp, &mi) != 0) 406 break; 407 /* 408 * The type of AUDIO_MIXER_CLASS merely introduces a class. 409 * All the other types describe an actual mixer. 410 */ 411 if (mi.type == AUDIO_MIXER_CLASS) { 412 if (strcmp(mi.label.name, AudioCinputs) == 0) 413 iclass = mi.mixer_class; 414 if (strcmp(mi.label.name, AudioCmonitor) == 0) 415 mclass = mi.mixer_class; 416 if (strcmp(mi.label.name, AudioCoutputs) == 0) 417 oclass = mi.mixer_class; 418 if (strcmp(mi.label.name, AudioCrecord) == 0) 419 rclass = mi.mixer_class; 420 } 421 } 422 /* 423 * This is where we assign each control in the "audio" model, to the 424 * underlying "mixer" control. We walk through the whole list once, 425 * assigning likely candidates as we come across them. 426 */ 427 record_master_found = 0; 428 record_source_found = 0; 429 for(mi.index = 0; ; mi.index++) { 430 if (hwp->query_devinfo(hdlp, &mi) != 0) 431 break; 432 if (mi.type == AUDIO_MIXER_CLASS) 433 continue; 434 if (mi.mixer_class == iclass) { 435 /* 436 * AudioCinputs is only a fallback, when we don't 437 * find what we're looking for in AudioCrecord, so 438 * check the flags before accepting one of these. 439 */ 440 if (strcmp(mi.label.name, AudioNmaster) == 0 441 && record_master_found == 0) 442 sc->sc_inports.master = mi.index; 443 if (strcmp(mi.label.name, AudioNsource) == 0 444 && record_source_found == 0) { 445 if (mi.type == AUDIO_MIXER_ENUM) { 446 int i; 447 for(i = 0; i < mi.un.e.num_mem; i++) 448 if (strcmp(mi.un.e.member[i].label.name, 449 AudioNmixerout) == 0) 450 sc->sc_inports.mixerout = 451 mi.un.e.member[i].ord; 452 } 453 au_setup_ports(sc, &sc->sc_inports, &mi, 454 itable); 455 } 456 if (strcmp(mi.label.name, AudioNdac) == 0 && 457 sc->sc_outports.master == -1) 458 sc->sc_outports.master = mi.index; 459 } else if (mi.mixer_class == mclass) { 460 if (strcmp(mi.label.name, AudioNmonitor) == 0) 461 sc->sc_monitor_port = mi.index; 462 } else if (mi.mixer_class == oclass) { 463 if (strcmp(mi.label.name, AudioNmaster) == 0) 464 sc->sc_outports.master = mi.index; 465 if (strcmp(mi.label.name, AudioNselect) == 0) 466 au_setup_ports(sc, &sc->sc_outports, &mi, 467 otable); 468 } else if (mi.mixer_class == rclass) { 469 /* 470 * These are the preferred mixers for the audio record 471 * controls, so set the flags here, but don't check. 472 */ 473 if (strcmp(mi.label.name, AudioNmaster) == 0) { 474 sc->sc_inports.master = mi.index; 475 record_master_found = 1; 476 } 477 #if 1 /* Deprecated. Use AudioNmaster. */ 478 if (strcmp(mi.label.name, AudioNrecord) == 0) { 479 sc->sc_inports.master = mi.index; 480 record_master_found = 1; 481 } 482 if (strcmp(mi.label.name, AudioNvolume) == 0) { 483 sc->sc_inports.master = mi.index; 484 record_master_found = 1; 485 } 486 #endif 487 if (strcmp(mi.label.name, AudioNsource) == 0) { 488 if (mi.type == AUDIO_MIXER_ENUM) { 489 int i; 490 for(i = 0; i < mi.un.e.num_mem; i++) 491 if (strcmp(mi.un.e.member[i].label.name, 492 AudioNmixerout) == 0) 493 sc->sc_inports.mixerout = 494 mi.un.e.member[i].ord; 495 } 496 au_setup_ports(sc, &sc->sc_inports, &mi, 497 itable); 498 record_source_found = 1; 499 } 500 } 501 } 502 DPRINTF(("audio_attach: inputs ports=0x%x, input master=%d, " 503 "output ports=0x%x, output master=%d\n", 504 sc->sc_inports.allports, sc->sc_inports.master, 505 sc->sc_outports.allports, sc->sc_outports.master)); 506 507 selinit(&sc->sc_rsel); 508 selinit(&sc->sc_wsel); 509 510 #ifdef AUDIO_PM_IDLE 511 callout_init(&sc->sc_idle_counter, 0); 512 callout_setfunc(&sc->sc_idle_counter, audio_idle, self); 513 #endif 514 515 if (!pmf_device_register(self, audio_suspend, audio_resume)) 516 aprint_error_dev(self, "couldn't establish power handler\n"); 517 #ifdef AUDIO_PM_IDLE 518 if (!device_active_register(self, audio_activity)) 519 aprint_error_dev(self, "couldn't register activity handler\n"); 520 #endif 521 522 if (!pmf_event_register(self, PMFE_AUDIO_VOLUME_DOWN, 523 audio_volume_down, true)) 524 aprint_error_dev(self, "couldn't add volume down handler\n"); 525 if (!pmf_event_register(self, PMFE_AUDIO_VOLUME_UP, 526 audio_volume_up, true)) 527 aprint_error_dev(self, "couldn't add volume up handler\n"); 528 if (!pmf_event_register(self, PMFE_AUDIO_VOLUME_TOGGLE, 529 audio_volume_toggle, true)) 530 aprint_error_dev(self, "couldn't add volume toggle handler\n"); 531 532 #ifdef AUDIO_PM_IDLE 533 callout_schedule(&sc->sc_idle_counter, audio_idle_timeout * hz); 534 #endif 535 } 536 537 int 538 audioactivate(device_t self, enum devact act) 539 { 540 struct audio_softc *sc; 541 542 sc = device_private(self); 543 switch (act) { 544 case DVACT_ACTIVATE: 545 return EOPNOTSUPP; 546 547 case DVACT_DEACTIVATE: 548 sc->sc_dying = true; 549 break; 550 } 551 return 0; 552 } 553 554 int 555 audiodetach(device_t self, int flags) 556 { 557 struct audio_softc *sc; 558 int maj, mn; 559 int s; 560 561 sc = device_private(self); 562 DPRINTF(("audio_detach: sc=%p flags=%d\n", sc, flags)); 563 564 sc->sc_dying = true; 565 566 pmf_event_deregister(self, PMFE_AUDIO_VOLUME_DOWN, 567 audio_volume_down, true); 568 pmf_event_deregister(self, PMFE_AUDIO_VOLUME_UP, 569 audio_volume_up, true); 570 pmf_event_deregister(self, PMFE_AUDIO_VOLUME_TOGGLE, 571 audio_volume_toggle, true); 572 573 #ifdef AUDIO_PM_IDLE 574 callout_stop(&sc->sc_idle_counter); 575 576 device_active_deregister(self, audio_activity); 577 #endif 578 579 pmf_device_deregister(self); 580 581 wakeup(&sc->sc_wchan); 582 wakeup(&sc->sc_rchan); 583 s = splaudio(); 584 if (--sc->sc_refcnt >= 0) { 585 if (tsleep(&sc->sc_refcnt, PZERO, "auddet", hz * 120)) 586 printf("audiodetach: %s didn't detach\n", 587 device_xname(sc->dev)); 588 } 589 splx(s); 590 591 /* free resources */ 592 audio_free_ring(sc, &sc->sc_pr); 593 audio_free_ring(sc, &sc->sc_rr); 594 audio_destruct_pfilters(sc); 595 audio_destruct_rfilters(sc); 596 597 /* locate the major number */ 598 maj = cdevsw_lookup_major(&audio_cdevsw); 599 600 /* Nuke the vnodes for any open instances (calls close). */ 601 mn = device_unit(self); 602 vdevgone(maj, mn | SOUND_DEVICE, mn | SOUND_DEVICE, VCHR); 603 vdevgone(maj, mn | AUDIO_DEVICE, mn | AUDIO_DEVICE, VCHR); 604 vdevgone(maj, mn | AUDIOCTL_DEVICE, mn | AUDIOCTL_DEVICE, VCHR); 605 vdevgone(maj, mn | MIXER_DEVICE, mn | MIXER_DEVICE, VCHR); 606 607 if (sc->sc_sih_rd) { 608 softint_disestablish(sc->sc_sih_rd); 609 sc->sc_sih_rd = NULL; 610 } 611 if (sc->sc_sih_wr) { 612 softint_disestablish(sc->sc_sih_wr); 613 sc->sc_sih_wr = NULL; 614 } 615 616 #ifdef AUDIO_PM_IDLE 617 callout_destroy(&sc->sc_idle_counter); 618 #endif 619 seldestroy(&sc->sc_rsel); 620 seldestroy(&sc->sc_wsel); 621 622 return 0; 623 } 624 625 int 626 au_portof(struct audio_softc *sc, char *name, int class) 627 { 628 mixer_devinfo_t mi; 629 630 for(mi.index = 0; 631 sc->hw_if->query_devinfo(sc->hw_hdl, &mi) == 0; 632 mi.index++) 633 if (mi.mixer_class == class && strcmp(mi.label.name, name) == 0) 634 return mi.index; 635 return -1; 636 } 637 638 void 639 au_setup_ports(struct audio_softc *sc, struct au_mixer_ports *ports, 640 mixer_devinfo_t *mi, const struct portname *tbl) 641 { 642 int i, j; 643 644 ports->index = mi->index; 645 if (mi->type == AUDIO_MIXER_ENUM) { 646 ports->isenum = true; 647 for(i = 0; tbl[i].name; i++) 648 for(j = 0; j < mi->un.e.num_mem; j++) 649 if (strcmp(mi->un.e.member[j].label.name, 650 tbl[i].name) == 0) { 651 ports->allports |= tbl[i].mask; 652 ports->aumask[ports->nports] = tbl[i].mask; 653 ports->misel[ports->nports] = 654 mi->un.e.member[j].ord; 655 ports->miport[ports->nports] = 656 au_portof(sc, mi->un.e.member[j].label.name, 657 mi->mixer_class); 658 if (ports->mixerout != -1 && 659 ports->miport[ports->nports] != -1) 660 ports->isdual = true; 661 ++ports->nports; 662 } 663 } else if (mi->type == AUDIO_MIXER_SET) { 664 for(i = 0; tbl[i].name; i++) 665 for(j = 0; j < mi->un.s.num_mem; j++) 666 if (strcmp(mi->un.s.member[j].label.name, 667 tbl[i].name) == 0) { 668 ports->allports |= tbl[i].mask; 669 ports->aumask[ports->nports] = tbl[i].mask; 670 ports->misel[ports->nports] = 671 mi->un.s.member[j].mask; 672 ports->miport[ports->nports] = 673 au_portof(sc, mi->un.s.member[j].label.name, 674 mi->mixer_class); 675 ++ports->nports; 676 } 677 } 678 } 679 680 /* 681 * Called from hardware driver. This is where the MI audio driver gets 682 * probed/attached to the hardware driver. 683 */ 684 device_t 685 audio_attach_mi(const struct audio_hw_if *ahwp, void *hdlp, device_t dev) 686 { 687 struct audio_attach_args arg; 688 689 #ifdef DIAGNOSTIC 690 if (ahwp == NULL) { 691 aprint_error("audio_attach_mi: NULL\n"); 692 return 0; 693 } 694 #endif 695 arg.type = AUDIODEV_TYPE_AUDIO; 696 arg.hwif = ahwp; 697 arg.hdl = hdlp; 698 return config_found(dev, &arg, audioprint); 699 } 700 701 #ifdef AUDIO_DEBUG 702 void audio_printsc(struct audio_softc *); 703 void audio_print_params(const char *, struct audio_params *); 704 705 void 706 audio_printsc(struct audio_softc *sc) 707 { 708 printf("hwhandle %p hw_if %p ", sc->hw_hdl, sc->hw_if); 709 printf("open 0x%x mode 0x%x\n", sc->sc_open, sc->sc_mode); 710 printf("rchan 0x%x wchan 0x%x ", sc->sc_rchan, sc->sc_wchan); 711 printf("rring used 0x%x pring used=%d\n", 712 audio_stream_get_used(&sc->sc_rr.s), 713 audio_stream_get_used(&sc->sc_pr.s)); 714 printf("rbus 0x%x pbus 0x%x ", sc->sc_rbus, sc->sc_pbus); 715 printf("blksize %d", sc->sc_pr.blksize); 716 printf("hiwat %d lowat %d\n", sc->sc_pr.usedhigh, sc->sc_pr.usedlow); 717 } 718 719 void 720 audio_print_params(const char *s, struct audio_params *p) 721 { 722 printf("%s enc=%u %uch %u/%ubit %uHz\n", s, p->encoding, p->channels, 723 p->validbits, p->precision, p->sample_rate); 724 } 725 #endif 726 727 int 728 audio_alloc_ring(struct audio_softc *sc, struct audio_ringbuffer *r, 729 int direction, size_t bufsize) 730 { 731 const struct audio_hw_if *hw; 732 void *hdl; 733 734 hw = sc->hw_if; 735 hdl = sc->hw_hdl; 736 /* 737 * Alloc DMA play and record buffers 738 */ 739 if (bufsize < AUMINBUF) 740 bufsize = AUMINBUF; 741 ROUNDSIZE(bufsize); 742 if (hw->round_buffersize) 743 bufsize = hw->round_buffersize(hdl, direction, bufsize); 744 if (hw->allocm) 745 r->s.start = hw->allocm(hdl, direction, bufsize, 746 M_DEVBUF, M_WAITOK); 747 else 748 r->s.start = malloc(bufsize, M_DEVBUF, M_WAITOK); 749 if (r->s.start == 0) 750 return ENOMEM; 751 r->s.bufsize = bufsize; 752 return 0; 753 } 754 755 void 756 audio_free_ring(struct audio_softc *sc, struct audio_ringbuffer *r) 757 { 758 if (r->s.start == 0) 759 return; 760 761 if (sc->hw_if->freem) 762 sc->hw_if->freem(sc->hw_hdl, r->s.start, M_DEVBUF); 763 else 764 free(r->s.start, M_DEVBUF); 765 r->s.start = 0; 766 } 767 768 static int 769 audio_setup_pfilters(struct audio_softc *sc, const audio_params_t *pp, 770 stream_filter_list_t *pfilters) 771 { 772 stream_filter_t *pf[AUDIO_MAX_FILTERS]; 773 audio_stream_t ps[AUDIO_MAX_FILTERS]; 774 const audio_params_t *from_param; 775 audio_params_t *to_param; 776 int i, n; 777 778 while (sc->sc_writing) { 779 sc->sc_waitcomp = 1; 780 (void)tsleep(sc, 0, "audioch", 10*hz); 781 } 782 783 memset(pf, 0, sizeof(pf)); 784 memset(ps, 0, sizeof(ps)); 785 from_param = pp; 786 for (i = 0; i < pfilters->req_size; i++) { 787 n = pfilters->req_size - i - 1; 788 to_param = &pfilters->filters[n].param; 789 audio_check_params(to_param); 790 pf[i] = pfilters->filters[n].factory(sc, from_param, to_param); 791 if (pf[i] == NULL) 792 break; 793 if (audio_stream_ctor(&ps[i], from_param, AU_RING_SIZE)) 794 break; 795 if (i > 0) 796 pf[i]->set_fetcher(pf[i], &pf[i - 1]->base); 797 from_param = to_param; 798 } 799 if (i < pfilters->req_size) { /* failure */ 800 DPRINTF(("%s: pfilters failure\n", __func__)); 801 for (; i >= 0; i--) { 802 if (pf[i] != NULL) 803 pf[i]->dtor(pf[i]); 804 audio_stream_dtor(&ps[i]); 805 } 806 sc->sc_waitcomp = 0; 807 return EINVAL; 808 } 809 810 audio_destruct_pfilters(sc); 811 memcpy(sc->sc_pfilters, pf, sizeof(pf)); 812 memcpy(sc->sc_pstreams, ps, sizeof(ps)); 813 sc->sc_npfilters = pfilters->req_size; 814 for (i = 0; i < pfilters->req_size; i++) { 815 pf[i]->set_inputbuffer(pf[i], &sc->sc_pstreams[i]); 816 } 817 818 /* hardware format and the buffer near to userland */ 819 if (pfilters->req_size <= 0) { 820 sc->sc_pr.s.param = *pp; 821 sc->sc_pustream = &sc->sc_pr.s; 822 } else { 823 sc->sc_pr.s.param = pfilters->filters[0].param; 824 sc->sc_pustream = &sc->sc_pstreams[0]; 825 } 826 #ifdef AUDIO_DEBUG 827 printf("%s: HW-buffer=%p pustream=%p\n", 828 __func__, &sc->sc_pr.s, sc->sc_pustream); 829 for (i = 0; i < pfilters->req_size; i++) { 830 char num[100]; 831 snprintf(num, 100, "[%d]", i); 832 audio_print_params(num, &sc->sc_pstreams[i].param); 833 } 834 audio_print_params("[HW]", &sc->sc_pr.s.param); 835 #endif /* AUDIO_DEBUG */ 836 837 sc->sc_waitcomp = 0; 838 return 0; 839 } 840 841 static int 842 audio_setup_rfilters(struct audio_softc *sc, const audio_params_t *rp, 843 stream_filter_list_t *rfilters) 844 { 845 stream_filter_t *rf[AUDIO_MAX_FILTERS]; 846 audio_stream_t rs[AUDIO_MAX_FILTERS]; 847 const audio_params_t *to_param; 848 audio_params_t *from_param; 849 int i; 850 851 memset(rf, 0, sizeof(rf)); 852 memset(rs, 0, sizeof(rs)); 853 for (i = 0; i < rfilters->req_size; i++) { 854 from_param = &rfilters->filters[i].param; 855 audio_check_params(from_param); 856 to_param = i + 1 < rfilters->req_size 857 ? &rfilters->filters[i + 1].param : rp; 858 rf[i] = rfilters->filters[i].factory(sc, from_param, to_param); 859 if (rf[i] == NULL) 860 break; 861 if (audio_stream_ctor(&rs[i], to_param, AU_RING_SIZE)) 862 break; 863 if (i > 0) { 864 rf[i]->set_fetcher(rf[i], &rf[i - 1]->base); 865 } else { 866 /* rf[0] has no previous fetcher because 867 * the audio hardware fills data to the 868 * input buffer. */ 869 rf[0]->set_inputbuffer(rf[0], &sc->sc_rr.s); 870 } 871 } 872 if (i < rfilters->req_size) { /* failure */ 873 DPRINTF(("%s: rfilters failure\n", __func__)); 874 for (; i >= 0; i--) { 875 if (rf[i] != NULL) 876 rf[i]->dtor(rf[i]); 877 audio_stream_dtor(&rs[i]); 878 } 879 return EINVAL; 880 } 881 882 audio_destruct_rfilters(sc); 883 memcpy(sc->sc_rfilters, rf, sizeof(rf)); 884 memcpy(sc->sc_rstreams, rs, sizeof(rs)); 885 sc->sc_nrfilters = rfilters->req_size; 886 for (i = 1; i < rfilters->req_size; i++) { 887 rf[i]->set_inputbuffer(rf[i], &sc->sc_rstreams[i - 1]); 888 } 889 890 /* hardware format and the buffer near to userland */ 891 if (rfilters->req_size <= 0) { 892 sc->sc_rr.s.param = *rp; 893 sc->sc_rustream = &sc->sc_rr.s; 894 } else { 895 sc->sc_rr.s.param = rfilters->filters[0].param; 896 sc->sc_rustream = &sc->sc_rstreams[rfilters->req_size - 1]; 897 } 898 #ifdef AUDIO_DEBUG 899 printf("%s: HW-buffer=%p pustream=%p\n", 900 __func__, &sc->sc_rr.s, sc->sc_rustream); 901 audio_print_params("[HW]", &sc->sc_rr.s.param); 902 for (i = 0; i < rfilters->req_size; i++) { 903 char num[100]; 904 snprintf(num, 100, "[%d]", i); 905 audio_print_params(num, &sc->sc_rstreams[i].param); 906 } 907 #endif /* AUDIO_DEBUG */ 908 return 0; 909 } 910 911 static void 912 audio_destruct_pfilters(struct audio_softc *sc) 913 { 914 int i; 915 916 for (i = 0; i < sc->sc_npfilters; i++) { 917 sc->sc_pfilters[i]->dtor(sc->sc_pfilters[i]); 918 sc->sc_pfilters[i] = NULL; 919 audio_stream_dtor(&sc->sc_pstreams[i]); 920 } 921 sc->sc_npfilters = 0; 922 } 923 924 static void 925 audio_destruct_rfilters(struct audio_softc *sc) 926 { 927 int i; 928 929 for (i = 0; i < sc->sc_nrfilters; i++) { 930 sc->sc_rfilters[i]->dtor(sc->sc_rfilters[i]); 931 sc->sc_rfilters[i] = NULL; 932 audio_stream_dtor(&sc->sc_rstreams[i]); 933 } 934 sc->sc_nrfilters = 0; 935 } 936 937 static void 938 audio_stream_dtor(audio_stream_t *stream) 939 { 940 941 if (stream->start != NULL) 942 free(stream->start, M_DEVBUF); 943 memset(stream, 0, sizeof(audio_stream_t)); 944 } 945 946 static int 947 audio_stream_ctor(audio_stream_t *stream, const audio_params_t *param, int size) 948 { 949 int frame_size; 950 951 size = min(size, AU_RING_SIZE); 952 stream->bufsize = size; 953 stream->start = malloc(size, M_DEVBUF, M_NOWAIT); 954 if (stream->start == NULL) 955 return ENOMEM; 956 frame_size = (param->precision + 7) / 8 * param->channels; 957 size = (size / frame_size) * frame_size; 958 stream->end = stream->start + size; 959 stream->inp = stream->start; 960 stream->outp = stream->start; 961 stream->used = 0; 962 stream->param = *param; 963 stream->loop = false; 964 return 0; 965 } 966 967 static void 968 stream_filter_list_append(stream_filter_list_t *list, 969 stream_filter_factory_t factory, 970 const audio_params_t *param) 971 { 972 973 if (list->req_size >= AUDIO_MAX_FILTERS) { 974 printf("%s: increase AUDIO_MAX_FILTERS in sys/dev/audio_if.h\n", 975 __func__); 976 return; 977 } 978 list->filters[list->req_size].factory = factory; 979 list->filters[list->req_size].param = *param; 980 list->req_size++; 981 } 982 983 static void 984 stream_filter_list_set(stream_filter_list_t *list, int i, 985 stream_filter_factory_t factory, 986 const audio_params_t *param) 987 { 988 989 if (i < 0 || i >= AUDIO_MAX_FILTERS) { 990 printf("%s: invalid index: %d\n", __func__, i); 991 return; 992 } 993 994 list->filters[i].factory = factory; 995 list->filters[i].param = *param; 996 if (list->req_size <= i) 997 list->req_size = i + 1; 998 } 999 1000 static void 1001 stream_filter_list_prepend(stream_filter_list_t *list, 1002 stream_filter_factory_t factory, 1003 const audio_params_t *param) 1004 { 1005 1006 if (list->req_size >= AUDIO_MAX_FILTERS) { 1007 printf("%s: increase AUDIO_MAX_FILTERS in sys/dev/audio_if.h\n", 1008 __func__); 1009 return; 1010 } 1011 memmove(&list->filters[1], &list->filters[0], 1012 sizeof(struct stream_filter_req) * list->req_size); 1013 list->filters[0].factory = factory; 1014 list->filters[0].param = *param; 1015 list->req_size++; 1016 } 1017 1018 int 1019 audioopen(dev_t dev, int flags, int ifmt, struct lwp *l) 1020 { 1021 struct audio_softc *sc; 1022 int error; 1023 1024 sc = device_lookup_private(&audio_cd, AUDIOUNIT(dev)); 1025 if (sc == NULL) 1026 return ENXIO; 1027 1028 if (sc->sc_dying) 1029 return EIO; 1030 1031 device_active(sc->dev, DVA_SYSTEM); 1032 1033 sc->sc_opencnt++; 1034 1035 sc->sc_refcnt++; 1036 switch (AUDIODEV(dev)) { 1037 case SOUND_DEVICE: 1038 case AUDIO_DEVICE: 1039 error = audio_open(dev, sc, flags, ifmt, l); 1040 break; 1041 case AUDIOCTL_DEVICE: 1042 error = 0; 1043 break; 1044 case MIXER_DEVICE: 1045 error = mixer_open(dev, sc, flags, ifmt, l); 1046 break; 1047 default: 1048 error = ENXIO; 1049 break; 1050 } 1051 if (--sc->sc_refcnt < 0) 1052 wakeup(&sc->sc_refcnt); 1053 return error; 1054 } 1055 1056 int 1057 audioclose(dev_t dev, int flags, int ifmt, struct lwp *l) 1058 { 1059 struct audio_softc *sc; 1060 int error; 1061 1062 sc = device_lookup_private(&audio_cd, AUDIOUNIT(dev)); 1063 1064 device_active(sc->dev, DVA_SYSTEM); 1065 1066 switch (AUDIODEV(dev)) { 1067 case SOUND_DEVICE: 1068 case AUDIO_DEVICE: 1069 error = audio_close(sc, flags, ifmt, l); 1070 break; 1071 case MIXER_DEVICE: 1072 error = mixer_close(sc, flags, ifmt, l); 1073 break; 1074 case AUDIOCTL_DEVICE: 1075 error = 0; 1076 break; 1077 default: 1078 error = ENXIO; 1079 break; 1080 } 1081 1082 sc->sc_opencnt--; 1083 1084 return error; 1085 } 1086 1087 int 1088 audioread(dev_t dev, struct uio *uio, int ioflag) 1089 { 1090 struct audio_softc *sc; 1091 int error; 1092 1093 sc = device_lookup_private(&audio_cd, AUDIOUNIT(dev)); 1094 if (sc == NULL) 1095 return ENXIO; 1096 1097 if (sc->sc_dying) 1098 return EIO; 1099 1100 sc->sc_refcnt++; 1101 switch (AUDIODEV(dev)) { 1102 case SOUND_DEVICE: 1103 case AUDIO_DEVICE: 1104 error = audio_read(sc, uio, ioflag); 1105 break; 1106 case AUDIOCTL_DEVICE: 1107 case MIXER_DEVICE: 1108 error = ENODEV; 1109 break; 1110 default: 1111 error = ENXIO; 1112 break; 1113 } 1114 if (--sc->sc_refcnt < 0) 1115 wakeup(&sc->sc_refcnt); 1116 return error; 1117 } 1118 1119 int 1120 audiowrite(dev_t dev, struct uio *uio, int ioflag) 1121 { 1122 struct audio_softc *sc; 1123 int error; 1124 1125 sc = device_lookup_private(&audio_cd, AUDIOUNIT(dev)); 1126 if (sc == NULL) 1127 return ENXIO; 1128 1129 if (sc->sc_dying) 1130 return EIO; 1131 1132 sc->sc_refcnt++; 1133 switch (AUDIODEV(dev)) { 1134 case SOUND_DEVICE: 1135 case AUDIO_DEVICE: 1136 error = audio_write(sc, uio, ioflag); 1137 break; 1138 case AUDIOCTL_DEVICE: 1139 case MIXER_DEVICE: 1140 error = ENODEV; 1141 break; 1142 default: 1143 error = ENXIO; 1144 break; 1145 } 1146 if (--sc->sc_refcnt < 0) 1147 wakeup(&sc->sc_refcnt); 1148 return error; 1149 } 1150 1151 int 1152 audioioctl(dev_t dev, u_long cmd, void *addr, int flag, struct lwp *l) 1153 { 1154 struct audio_softc *sc; 1155 int error; 1156 1157 sc = device_lookup_private(&audio_cd, AUDIOUNIT(dev)); 1158 if (sc->sc_dying) 1159 return EIO; 1160 1161 sc->sc_refcnt++; 1162 switch (AUDIODEV(dev)) { 1163 case SOUND_DEVICE: 1164 case AUDIO_DEVICE: 1165 case AUDIOCTL_DEVICE: 1166 device_active(sc->dev, DVA_SYSTEM); 1167 if (IOCGROUP(cmd) == IOCGROUP(AUDIO_MIXER_READ)) 1168 error = mixer_ioctl(sc, cmd, addr, flag, l); 1169 else 1170 error = audio_ioctl(sc, cmd, addr, flag, l); 1171 break; 1172 case MIXER_DEVICE: 1173 error = mixer_ioctl(sc, cmd, addr, flag, l); 1174 break; 1175 default: 1176 error = ENXIO; 1177 break; 1178 } 1179 if (--sc->sc_refcnt < 0) 1180 wakeup(&sc->sc_refcnt); 1181 return error; 1182 } 1183 1184 int 1185 audiopoll(dev_t dev, int events, struct lwp *l) 1186 { 1187 struct audio_softc *sc; 1188 int revents; 1189 1190 sc = device_lookup_private(&audio_cd, AUDIOUNIT(dev)); 1191 if (sc->sc_dying) 1192 return POLLHUP; 1193 1194 sc->sc_refcnt++; 1195 switch (AUDIODEV(dev)) { 1196 case SOUND_DEVICE: 1197 case AUDIO_DEVICE: 1198 revents = audio_poll(sc, events, l); 1199 break; 1200 case AUDIOCTL_DEVICE: 1201 case MIXER_DEVICE: 1202 revents = 0; 1203 break; 1204 default: 1205 revents = POLLERR; 1206 break; 1207 } 1208 if (--sc->sc_refcnt < 0) 1209 wakeup(&sc->sc_refcnt); 1210 return revents; 1211 } 1212 1213 int 1214 audiokqfilter(dev_t dev, struct knote *kn) 1215 { 1216 struct audio_softc *sc; 1217 int rv; 1218 1219 sc = device_lookup_private(&audio_cd, AUDIOUNIT(dev)); 1220 if (sc->sc_dying) 1221 return 1; 1222 1223 sc->sc_refcnt++; 1224 switch (AUDIODEV(dev)) { 1225 case SOUND_DEVICE: 1226 case AUDIO_DEVICE: 1227 rv = audio_kqfilter(sc, kn); 1228 break; 1229 case AUDIOCTL_DEVICE: 1230 case MIXER_DEVICE: 1231 rv = 1; 1232 break; 1233 default: 1234 rv = 1; 1235 } 1236 if (--sc->sc_refcnt < 0) 1237 wakeup(&sc->sc_refcnt); 1238 return rv; 1239 } 1240 1241 paddr_t 1242 audiommap(dev_t dev, off_t off, int prot) 1243 { 1244 struct audio_softc *sc; 1245 paddr_t error; 1246 1247 sc = device_lookup_private(&audio_cd, AUDIOUNIT(dev)); 1248 if (sc->sc_dying) 1249 return -1; 1250 1251 device_active(sc->dev, DVA_SYSTEM); /* XXXJDM */ 1252 1253 sc->sc_refcnt++; 1254 switch (AUDIODEV(dev)) { 1255 case SOUND_DEVICE: 1256 case AUDIO_DEVICE: 1257 error = audio_mmap(sc, off, prot); 1258 break; 1259 case AUDIOCTL_DEVICE: 1260 case MIXER_DEVICE: 1261 error = -1; 1262 break; 1263 default: 1264 error = -1; 1265 break; 1266 } 1267 if (--sc->sc_refcnt < 0) 1268 wakeup(&sc->sc_refcnt); 1269 return error; 1270 } 1271 1272 /* 1273 * Audio driver 1274 */ 1275 void 1276 audio_init_ringbuffer(struct audio_softc *sc, struct audio_ringbuffer *rp, 1277 int mode) 1278 { 1279 int nblks; 1280 int blksize; 1281 1282 blksize = rp->blksize; 1283 if (blksize < AUMINBLK) 1284 blksize = AUMINBLK; 1285 if (blksize > rp->s.bufsize / AUMINNOBLK) 1286 blksize = rp->s.bufsize / AUMINNOBLK; 1287 ROUNDSIZE(blksize); 1288 DPRINTF(("audio_init_ringbuffer: MI blksize=%d\n", blksize)); 1289 if (sc->hw_if->round_blocksize) 1290 blksize = sc->hw_if->round_blocksize(sc->hw_hdl, blksize, 1291 mode, &rp->s.param); 1292 if (blksize <= 0) 1293 panic("audio_init_ringbuffer: blksize"); 1294 nblks = rp->s.bufsize / blksize; 1295 1296 DPRINTF(("audio_init_ringbuffer: final blksize=%d\n", blksize)); 1297 rp->blksize = blksize; 1298 rp->maxblks = nblks; 1299 rp->s.end = rp->s.start + nblks * blksize; 1300 rp->s.outp = rp->s.inp = rp->s.start; 1301 rp->s.used = 0; 1302 rp->stamp = 0; 1303 rp->stamp_last = 0; 1304 rp->fstamp = 0; 1305 rp->drops = 0; 1306 rp->copying = false; 1307 rp->needfill = false; 1308 rp->mmapped = false; 1309 } 1310 1311 int 1312 audio_initbufs(struct audio_softc *sc) 1313 { 1314 const struct audio_hw_if *hw; 1315 int error; 1316 1317 DPRINTF(("audio_initbufs: mode=0x%x\n", sc->sc_mode)); 1318 hw = sc->hw_if; 1319 if (audio_can_capture(sc)) { 1320 audio_init_ringbuffer(sc, &sc->sc_rr, AUMODE_RECORD); 1321 if (hw->init_input && (sc->sc_mode & AUMODE_RECORD)) { 1322 error = hw->init_input(sc->hw_hdl, sc->sc_rr.s.start, 1323 sc->sc_rr.s.end - sc->sc_rr.s.start); 1324 if (error) 1325 return error; 1326 } 1327 } 1328 1329 if (audio_can_playback(sc)) { 1330 audio_init_ringbuffer(sc, &sc->sc_pr, AUMODE_PLAY); 1331 sc->sc_sil_count = 0; 1332 if (hw->init_output && (sc->sc_mode & AUMODE_PLAY)) { 1333 error = hw->init_output(sc->hw_hdl, sc->sc_pr.s.start, 1334 sc->sc_pr.s.end - sc->sc_pr.s.start); 1335 if (error) 1336 return error; 1337 } 1338 } 1339 1340 #ifdef AUDIO_INTR_TIME 1341 #define double u_long 1342 if (audio_can_playback(sc)) { 1343 sc->sc_pnintr = 0; 1344 sc->sc_pblktime = (u_long)( 1345 (double)sc->sc_pr.blksize * 100000 / 1346 (double)(sc->sc_pparams.precision / NBBY * 1347 sc->sc_pparams.channels * 1348 sc->sc_pparams.sample_rate)) * 10; 1349 DPRINTF(("audio: play blktime = %lu for %d\n", 1350 sc->sc_pblktime, sc->sc_pr.blksize)); 1351 } 1352 if (audio_can_capture(sc)) { 1353 sc->sc_rnintr = 0; 1354 sc->sc_rblktime = (u_long)( 1355 (double)sc->sc_rr.blksize * 100000 / 1356 (double)(sc->sc_rparams.precision / NBBY * 1357 sc->sc_rparams.channels * 1358 sc->sc_rparams.sample_rate)) * 10; 1359 DPRINTF(("audio: record blktime = %lu for %d\n", 1360 sc->sc_rblktime, sc->sc_rr.blksize)); 1361 } 1362 #undef double 1363 #endif 1364 1365 return 0; 1366 } 1367 1368 void 1369 audio_calcwater(struct audio_softc *sc) 1370 { 1371 1372 /* set high at 100% */ 1373 if (audio_can_playback(sc)) { 1374 sc->sc_pr.usedhigh = 1375 sc->sc_pustream->end - sc->sc_pustream->start; 1376 /* set low at 75% of usedhigh */ 1377 sc->sc_pr.usedlow = sc->sc_pr.usedhigh * 3 / 4; 1378 if (sc->sc_pr.usedlow == sc->sc_pr.usedhigh) 1379 sc->sc_pr.usedlow -= sc->sc_pr.blksize; 1380 } 1381 1382 if (audio_can_capture(sc)) { 1383 sc->sc_rr.usedhigh = 1384 sc->sc_rustream->end - sc->sc_rustream->start - 1385 sc->sc_rr.blksize; 1386 sc->sc_rr.usedlow = 0; 1387 DPRINTF(("%s: plow=%d phigh=%d rlow=%d rhigh=%d\n", __func__, 1388 sc->sc_pr.usedlow, sc->sc_pr.usedhigh, 1389 sc->sc_rr.usedlow, sc->sc_rr.usedhigh)); 1390 } 1391 } 1392 1393 static inline int 1394 audio_sleep_timo(int *chan, const char *label, int timo) 1395 { 1396 int st; 1397 1398 if (label == NULL) 1399 label = "audio"; 1400 1401 DPRINTFN(3, ("audio_sleep_timo: chan=%p, label=%s, timo=%d\n", 1402 chan, label, timo)); 1403 *chan = 1; 1404 st = tsleep(chan, PWAIT | PCATCH, label, timo); 1405 *chan = 0; 1406 #ifdef AUDIO_DEBUG 1407 if (st != 0 && st != EINTR) 1408 DPRINTF(("audio_sleep: woke up st=%d\n", st)); 1409 #endif 1410 return st; 1411 } 1412 1413 static inline int 1414 audio_sleep(int *chan, const char *label) 1415 { 1416 1417 return audio_sleep_timo(chan, label, 0); 1418 } 1419 1420 /* call at splaudio() */ 1421 static inline void 1422 audio_wakeup(int *chan) 1423 { 1424 1425 DPRINTFN(3, ("audio_wakeup: chan=%p, *chan=%d\n", chan, *chan)); 1426 if (*chan) { 1427 wakeup(chan); 1428 *chan = 0; 1429 } 1430 } 1431 1432 int 1433 audio_open(dev_t dev, struct audio_softc *sc, int flags, int ifmt, 1434 struct lwp *l) 1435 { 1436 int error; 1437 u_int mode; 1438 const struct audio_hw_if *hw; 1439 1440 hw = sc->hw_if; 1441 if (hw == NULL) 1442 return ENXIO; 1443 1444 DPRINTF(("audio_open: flags=0x%x sc=%p hdl=%p\n", 1445 flags, sc, sc->hw_hdl)); 1446 1447 if (((flags & FREAD) && (sc->sc_open & AUOPEN_READ)) || 1448 ((flags & FWRITE) && (sc->sc_open & AUOPEN_WRITE))) 1449 return EBUSY; 1450 1451 if (hw->open != NULL) { 1452 error = hw->open(sc->hw_hdl, flags); 1453 if (error) 1454 return error; 1455 } 1456 1457 sc->sc_async_audio = 0; 1458 sc->sc_rchan = 0; 1459 sc->sc_wchan = 0; 1460 sc->sc_sil_count = 0; 1461 sc->sc_rbus = false; 1462 sc->sc_pbus = false; 1463 sc->sc_eof = 0; 1464 sc->sc_playdrop = 0; 1465 1466 sc->sc_full_duplex = 1467 (flags & (FWRITE|FREAD)) == (FWRITE|FREAD) && 1468 (audio_get_props(sc) & AUDIO_PROP_FULLDUPLEX); 1469 1470 mode = 0; 1471 if (flags & FREAD) { 1472 sc->sc_open |= AUOPEN_READ; 1473 mode |= AUMODE_RECORD; 1474 } 1475 if (flags & FWRITE) { 1476 sc->sc_open |= AUOPEN_WRITE; 1477 mode |= AUMODE_PLAY | AUMODE_PLAY_ALL; 1478 } 1479 1480 /* 1481 * Multiplex device: /dev/audio (MU-Law) and /dev/sound (linear) 1482 * The /dev/audio is always (re)set to 8-bit MU-Law mono 1483 * For the other devices, you get what they were last set to. 1484 */ 1485 if (ISDEVAUDIO(dev)) { 1486 error = audio_set_defaults(sc, mode); 1487 } else { 1488 struct audio_info ai; 1489 1490 AUDIO_INITINFO(&ai); 1491 ai.mode = mode; 1492 error = audiosetinfo(sc, &ai); 1493 } 1494 if (error) 1495 goto bad; 1496 1497 #ifdef DIAGNOSTIC 1498 /* 1499 * Sample rate and precision are supposed to be set to proper 1500 * default values by the hardware driver, so that it may give 1501 * us these values. 1502 */ 1503 if (sc->sc_rparams.precision == 0 || sc->sc_pparams.precision == 0) { 1504 printf("audio_open: 0 precision\n"); 1505 return EINVAL; 1506 } 1507 #endif 1508 1509 /* audio_close() decreases sc_pr.usedlow, recalculate here */ 1510 audio_calcwater(sc); 1511 1512 DPRINTF(("audio_open: done sc_mode = 0x%x\n", sc->sc_mode)); 1513 1514 return 0; 1515 1516 bad: 1517 if (hw->close != NULL) 1518 hw->close(sc->hw_hdl); 1519 sc->sc_open = 0; 1520 sc->sc_mode = 0; 1521 sc->sc_full_duplex = 0; 1522 return error; 1523 } 1524 1525 /* 1526 * Must be called from task context. 1527 */ 1528 void 1529 audio_init_record(struct audio_softc *sc) 1530 { 1531 int s; 1532 1533 s = splaudio(); 1534 if (sc->hw_if->speaker_ctl && 1535 (!sc->sc_full_duplex || (sc->sc_mode & AUMODE_PLAY) == 0)) 1536 sc->hw_if->speaker_ctl(sc->hw_hdl, SPKR_OFF); 1537 splx(s); 1538 } 1539 1540 /* 1541 * Must be called from task context. 1542 */ 1543 void 1544 audio_init_play(struct audio_softc *sc) 1545 { 1546 int s; 1547 1548 s = splaudio(); 1549 sc->sc_wstamp = sc->sc_pr.stamp; 1550 if (sc->hw_if->speaker_ctl) 1551 sc->hw_if->speaker_ctl(sc->hw_hdl, SPKR_ON); 1552 splx(s); 1553 } 1554 1555 int 1556 audio_drain(struct audio_softc *sc) 1557 { 1558 struct audio_ringbuffer *cb; 1559 int error, drops; 1560 int s; 1561 int i, used; 1562 1563 DPRINTF(("audio_drain: enter busy=%d\n", sc->sc_pbus)); 1564 cb = &sc->sc_pr; 1565 if (cb->mmapped) 1566 return 0; 1567 1568 used = audio_stream_get_used(&sc->sc_pr.s); 1569 s = splaudio(); 1570 for (i = 0; i < sc->sc_npfilters; i++) 1571 used += audio_stream_get_used(&sc->sc_pstreams[i]); 1572 splx(s); 1573 if (used <= 0) 1574 return 0; 1575 1576 if (!sc->sc_pbus) { 1577 /* We've never started playing, probably because the 1578 * block was too short. Pad it and start now. 1579 */ 1580 int cc; 1581 uint8_t *inp = cb->s.inp; 1582 1583 cc = cb->blksize - (inp - cb->s.start) % cb->blksize; 1584 audio_fill_silence(&cb->s.param, inp, cc); 1585 s = splaudio(); 1586 cb->s.inp = audio_stream_add_inp(&cb->s, inp, cc); 1587 error = audiostartp(sc); 1588 splx(s); 1589 if (error) 1590 return error; 1591 } 1592 /* 1593 * Play until a silence block has been played, then we 1594 * know all has been drained. 1595 * XXX This should be done some other way to avoid 1596 * playing silence. 1597 */ 1598 #ifdef DIAGNOSTIC 1599 if (cb->copying) { 1600 printf("audio_drain: copying in progress!?!\n"); 1601 cb->copying = false; 1602 } 1603 #endif 1604 drops = cb->drops; 1605 error = 0; 1606 s = splaudio(); 1607 while (cb->drops == drops && !error) { 1608 DPRINTF(("audio_drain: used=%d, drops=%ld\n", 1609 audio_stream_get_used(&sc->sc_pr.s), cb->drops)); 1610 /* 1611 * When the process is exiting, it ignores all signals and 1612 * we can't interrupt this sleep, so we set a timeout 1613 * just in case. 1614 */ 1615 error = audio_sleep_timo(&sc->sc_wchan, "aud_dr", 30*hz); 1616 if (sc->sc_dying) 1617 error = EIO; 1618 } 1619 splx(s); 1620 return error; 1621 } 1622 1623 /* 1624 * Close an audio chip. 1625 */ 1626 /* ARGSUSED */ 1627 int 1628 audio_close(struct audio_softc *sc, int flags, int ifmt, 1629 struct lwp *l) 1630 { 1631 const struct audio_hw_if *hw; 1632 int s; 1633 1634 DPRINTF(("audio_close: sc=%p\n", sc)); 1635 hw = sc->hw_if; 1636 s = splaudio(); 1637 /* Stop recording. */ 1638 if ((flags & FREAD) && sc->sc_rbus) { 1639 /* 1640 * XXX Some drivers (e.g. SB) use the same routine 1641 * to halt input and output so don't halt input if 1642 * in full duplex mode. These drivers should be fixed. 1643 */ 1644 if (!sc->sc_full_duplex || hw->halt_input != hw->halt_output) 1645 hw->halt_input(sc->hw_hdl); 1646 sc->sc_rbus = false; 1647 } 1648 /* 1649 * Block until output drains, but allow ^C interrupt. 1650 */ 1651 sc->sc_pr.usedlow = sc->sc_pr.blksize; /* avoid excessive wakeups */ 1652 /* 1653 * If there is pending output, let it drain (unless 1654 * the output is paused). 1655 */ 1656 if ((flags & FWRITE) && sc->sc_pbus) { 1657 if (!sc->sc_pr.pause && !audio_drain(sc) && hw->drain) 1658 (void)hw->drain(sc->hw_hdl); 1659 hw->halt_output(sc->hw_hdl); 1660 sc->sc_pbus = false; 1661 } 1662 1663 if (hw->close != NULL) 1664 hw->close(sc->hw_hdl); 1665 1666 sc->sc_open = 0; 1667 sc->sc_mode = 0; 1668 sc->sc_async_audio = 0; 1669 sc->sc_full_duplex = 0; 1670 splx(s); 1671 DPRINTF(("audio_close: done\n")); 1672 1673 return 0; 1674 } 1675 1676 int 1677 audio_read(struct audio_softc *sc, struct uio *uio, int ioflag) 1678 { 1679 struct audio_ringbuffer *cb; 1680 const uint8_t *outp; 1681 uint8_t *inp; 1682 int error, s, used, cc, n; 1683 1684 cb = &sc->sc_rr; 1685 if (cb->mmapped) 1686 return EINVAL; 1687 1688 DPRINTFN(1,("audio_read: cc=%zu mode=%d\n", 1689 uio->uio_resid, sc->sc_mode)); 1690 1691 #ifdef AUDIO_PM_IDLE 1692 if (device_is_active(&sc->dev) || sc->sc_idle) 1693 device_active(&sc->dev, DVA_SYSTEM); 1694 #endif 1695 1696 error = 0; 1697 /* 1698 * If hardware is half-duplex and currently playing, return 1699 * silence blocks based on the number of blocks we have output. 1700 */ 1701 if (!sc->sc_full_duplex && (sc->sc_mode & AUMODE_PLAY)) { 1702 while (uio->uio_resid > 0 && !error) { 1703 s = splaudio(); 1704 for(;;) { 1705 cc = sc->sc_pr.stamp - sc->sc_wstamp; 1706 if (cc > 0) 1707 break; 1708 DPRINTF(("audio_read: stamp=%lu, wstamp=%lu\n", 1709 sc->sc_pr.stamp, sc->sc_wstamp)); 1710 if (ioflag & IO_NDELAY) { 1711 splx(s); 1712 return EWOULDBLOCK; 1713 } 1714 error = audio_sleep(&sc->sc_rchan, "aud_hr"); 1715 if (sc->sc_dying) 1716 error = EIO; 1717 if (error) { 1718 splx(s); 1719 return error; 1720 } 1721 } 1722 splx(s); 1723 1724 if (uio->uio_resid < cc) 1725 cc = uio->uio_resid; 1726 DPRINTFN(1,("audio_read: reading in write mode, " 1727 "cc=%d\n", cc)); 1728 error = audio_silence_copyout(sc, cc, uio); 1729 sc->sc_wstamp += cc; 1730 } 1731 return error; 1732 } 1733 1734 while (uio->uio_resid > 0 && !error) { 1735 s = splaudio(); 1736 while ((used = audio_stream_get_used(sc->sc_rustream)) <= 0) { 1737 if (!sc->sc_rbus && !sc->sc_rr.pause) { 1738 error = audiostartr(sc); 1739 if (error) { 1740 splx(s); 1741 return error; 1742 } 1743 } 1744 if (ioflag & IO_NDELAY) { 1745 splx(s); 1746 return EWOULDBLOCK; 1747 } 1748 DPRINTFN(2, ("audio_read: sleep used=%d\n", used)); 1749 error = audio_sleep(&sc->sc_rchan, "aud_rd"); 1750 if (sc->sc_dying) 1751 error = EIO; 1752 if (error) { 1753 splx(s); 1754 return error; 1755 } 1756 } 1757 1758 outp = sc->sc_rustream->outp; 1759 inp = sc->sc_rustream->inp; 1760 cb->copying = true; 1761 splx(s); 1762 1763 /* 1764 * cc is the amount of data in the sc_rustream excluding 1765 * wrapped data. Note the tricky case of inp == outp, which 1766 * must mean the buffer is full, not empty, because used > 0. 1767 */ 1768 cc = outp < inp ? inp - outp :sc->sc_rustream->end - outp; 1769 DPRINTFN(1,("audio_read: outp=%p, cc=%d\n", outp, cc)); 1770 1771 n = uio->uio_resid; 1772 error = uiomove(__UNCONST(outp), cc, uio); 1773 n -= uio->uio_resid; /* number of bytes actually moved */ 1774 1775 s = splaudio(); 1776 sc->sc_rustream->outp = audio_stream_add_outp 1777 (sc->sc_rustream, outp, n); 1778 cb->copying = false; 1779 splx(s); 1780 } 1781 return error; 1782 } 1783 1784 void 1785 audio_clear(struct audio_softc *sc) 1786 { 1787 int s; 1788 1789 s = splaudio(); 1790 if (sc->sc_rbus) { 1791 audio_wakeup(&sc->sc_rchan); 1792 sc->hw_if->halt_input(sc->hw_hdl); 1793 sc->sc_rbus = false; 1794 sc->sc_rr.pause = false; 1795 } 1796 if (sc->sc_pbus) { 1797 audio_wakeup(&sc->sc_wchan); 1798 sc->hw_if->halt_output(sc->hw_hdl); 1799 sc->sc_pbus = false; 1800 sc->sc_pr.pause = false; 1801 } 1802 splx(s); 1803 } 1804 1805 void 1806 audio_calc_blksize(struct audio_softc *sc, int mode) 1807 { 1808 const audio_params_t *parm; 1809 struct audio_ringbuffer *rb; 1810 1811 if (sc->sc_blkset) 1812 return; 1813 1814 if (mode == AUMODE_PLAY) { 1815 rb = &sc->sc_pr; 1816 parm = &rb->s.param; 1817 } else { 1818 rb = &sc->sc_rr; 1819 parm = &rb->s.param; 1820 } 1821 1822 rb->blksize = parm->sample_rate * audio_blk_ms / 1000 * 1823 parm->channels * parm->precision / NBBY; 1824 1825 DPRINTF(("audio_calc_blksize: %s blksize=%d\n", 1826 mode == AUMODE_PLAY ? "play" : "record", rb->blksize)); 1827 } 1828 1829 void 1830 audio_fill_silence(struct audio_params *params, uint8_t *p, int n) 1831 { 1832 uint8_t auzero0, auzero1; 1833 int nfill; 1834 1835 auzero1 = 0; /* initialize to please gcc */ 1836 nfill = 1; 1837 switch (params->encoding) { 1838 case AUDIO_ENCODING_ULAW: 1839 auzero0 = 0x7f; 1840 break; 1841 case AUDIO_ENCODING_ALAW: 1842 auzero0 = 0x55; 1843 break; 1844 case AUDIO_ENCODING_MPEG_L1_STREAM: 1845 case AUDIO_ENCODING_MPEG_L1_PACKETS: 1846 case AUDIO_ENCODING_MPEG_L1_SYSTEM: 1847 case AUDIO_ENCODING_MPEG_L2_STREAM: 1848 case AUDIO_ENCODING_MPEG_L2_PACKETS: 1849 case AUDIO_ENCODING_MPEG_L2_SYSTEM: 1850 case AUDIO_ENCODING_ADPCM: /* is this right XXX */ 1851 case AUDIO_ENCODING_SLINEAR_LE: 1852 case AUDIO_ENCODING_SLINEAR_BE: 1853 auzero0 = 0;/* fortunately this works for any number of bits */ 1854 break; 1855 case AUDIO_ENCODING_ULINEAR_LE: 1856 case AUDIO_ENCODING_ULINEAR_BE: 1857 if (params->precision > 8) { 1858 nfill = (params->precision + NBBY - 1)/ NBBY; 1859 auzero0 = 0x80; 1860 auzero1 = 0; 1861 } else 1862 auzero0 = 0x80; 1863 break; 1864 default: 1865 DPRINTF(("audio: bad encoding %d\n", params->encoding)); 1866 auzero0 = 0; 1867 break; 1868 } 1869 if (nfill == 1) { 1870 while (--n >= 0) 1871 *p++ = auzero0; /* XXX memset */ 1872 } else /* nfill must no longer be 2 */ { 1873 if (params->encoding == AUDIO_ENCODING_ULINEAR_LE) { 1874 int k = nfill; 1875 while (--k > 0) 1876 *p++ = auzero1; 1877 n -= nfill - 1; 1878 } 1879 while (n >= nfill) { 1880 int k = nfill; 1881 *p++ = auzero0; 1882 while (--k > 0) 1883 *p++ = auzero1; 1884 1885 n -= nfill; 1886 } 1887 if (n-- > 0) /* XXX must be 1 - DIAGNOSTIC check? */ 1888 *p++ = auzero0; 1889 } 1890 } 1891 1892 int 1893 audio_silence_copyout(struct audio_softc *sc, int n, struct uio *uio) 1894 { 1895 uint8_t zerobuf[128]; 1896 int error; 1897 int k; 1898 1899 audio_fill_silence(&sc->sc_rparams, zerobuf, sizeof zerobuf); 1900 1901 error = 0; 1902 while (n > 0 && uio->uio_resid > 0 && !error) { 1903 k = min(n, min(uio->uio_resid, sizeof zerobuf)); 1904 error = uiomove(zerobuf, k, uio); 1905 n -= k; 1906 } 1907 return error; 1908 } 1909 1910 static int 1911 uio_fetcher_fetch_to(stream_fetcher_t *self, audio_stream_t *p, 1912 int max_used) 1913 { 1914 uio_fetcher_t *this; 1915 int size; 1916 int stream_space; 1917 int error; 1918 1919 this = (uio_fetcher_t *)self; 1920 this->last_used = audio_stream_get_used(p); 1921 if (this->last_used >= this->usedhigh) 1922 return 0; 1923 /* 1924 * uio_fetcher ignores max_used and move the data as 1925 * much as possible in order to return the correct value 1926 * for audio_prinfo::seek and kfilters. 1927 */ 1928 stream_space = audio_stream_get_space(p); 1929 size = min(this->uio->uio_resid, stream_space); 1930 1931 /* the first fragment of the space */ 1932 stream_space = p->end - p->inp; 1933 if (stream_space >= size) { 1934 error = uiomove(p->inp, size, this->uio); 1935 if (error) 1936 return error; 1937 p->inp = audio_stream_add_inp(p, p->inp, size); 1938 } else { 1939 error = uiomove(p->inp, stream_space, this->uio); 1940 if (error) 1941 return error; 1942 p->inp = audio_stream_add_inp(p, p->inp, stream_space); 1943 error = uiomove(p->start, size - stream_space, this->uio); 1944 if (error) 1945 return error; 1946 p->inp = audio_stream_add_inp(p, p->inp, size - stream_space); 1947 } 1948 this->last_used = audio_stream_get_used(p); 1949 return 0; 1950 } 1951 1952 static int 1953 null_fetcher_fetch_to(stream_fetcher_t *self, 1954 audio_stream_t *p, int max_used) 1955 { 1956 1957 return 0; 1958 } 1959 1960 static void 1961 uio_fetcher_ctor(uio_fetcher_t *this, struct uio *u, int h) 1962 { 1963 1964 this->base.fetch_to = uio_fetcher_fetch_to; 1965 this->uio = u; 1966 this->usedhigh = h; 1967 } 1968 1969 int 1970 audio_write(struct audio_softc *sc, struct uio *uio, int ioflag) 1971 { 1972 uio_fetcher_t ufetcher; 1973 audio_stream_t stream; 1974 struct audio_ringbuffer *cb; 1975 stream_fetcher_t *fetcher; 1976 stream_filter_t *filter; 1977 uint8_t *inp, *einp; 1978 int saveerror, error, s, n, cc, used; 1979 1980 DPRINTFN(2,("audio_write: sc=%p count=%zu used=%d(hi=%d)\n", 1981 sc, uio->uio_resid, audio_stream_get_used(sc->sc_pustream), 1982 sc->sc_pr.usedhigh)); 1983 cb = &sc->sc_pr; 1984 if (cb->mmapped) 1985 return EINVAL; 1986 1987 if (uio->uio_resid == 0) { 1988 sc->sc_eof++; 1989 return 0; 1990 } 1991 1992 #ifdef AUDIO_PM_IDLE 1993 if (device_is_active(&sc->dev) || sc->sc_idle) 1994 device_active(&sc->dev, DVA_SYSTEM); 1995 #endif 1996 1997 /* 1998 * If half-duplex and currently recording, throw away data. 1999 */ 2000 if (!sc->sc_full_duplex && 2001 (sc->sc_mode & AUMODE_RECORD)) { 2002 uio->uio_offset += uio->uio_resid; 2003 uio->uio_resid = 0; 2004 DPRINTF(("audio_write: half-dpx read busy\n")); 2005 return 0; 2006 } 2007 2008 if (!(sc->sc_mode & AUMODE_PLAY_ALL) && sc->sc_playdrop > 0) { 2009 n = min(sc->sc_playdrop, uio->uio_resid); 2010 DPRINTF(("audio_write: playdrop %d\n", n)); 2011 uio->uio_offset += n; 2012 uio->uio_resid -= n; 2013 sc->sc_playdrop -= n; 2014 if (uio->uio_resid == 0) 2015 return 0; 2016 } 2017 2018 /** 2019 * setup filter pipeline 2020 */ 2021 uio_fetcher_ctor(&ufetcher, uio, cb->usedhigh); 2022 if (sc->sc_npfilters > 0) { 2023 fetcher = &sc->sc_pfilters[sc->sc_npfilters - 1]->base; 2024 } else { 2025 fetcher = &ufetcher.base; 2026 } 2027 2028 error = 0; 2029 while (uio->uio_resid > 0 && !error) { 2030 s = splaudio(); 2031 /* wait if the first buffer is occupied */ 2032 while ((used = audio_stream_get_used(sc->sc_pustream)) 2033 >= cb->usedhigh) { 2034 DPRINTFN(2, ("audio_write: sleep used=%d lowat=%d " 2035 "hiwat=%d\n", used, 2036 cb->usedlow, cb->usedhigh)); 2037 if (ioflag & IO_NDELAY) { 2038 splx(s); 2039 return EWOULDBLOCK; 2040 } 2041 error = audio_sleep(&sc->sc_wchan, "aud_wr"); 2042 if (sc->sc_dying) 2043 error = EIO; 2044 if (error) { 2045 splx(s); 2046 return error; 2047 } 2048 } 2049 inp = cb->s.inp; 2050 cb->copying = true; 2051 stream = cb->s; 2052 used = stream.used; 2053 splx(s); 2054 2055 /* 2056 * write to the sc_pustream as much as possible 2057 * 2058 * work with a temporary audio_stream_t to narrow 2059 * splaudio() enclosure 2060 */ 2061 2062 sc->sc_writing = 1; 2063 2064 if (sc->sc_npfilters > 0) { 2065 filter = sc->sc_pfilters[0]; 2066 filter->set_fetcher(filter, &ufetcher.base); 2067 fetcher = &sc->sc_pfilters[sc->sc_npfilters - 1]->base; 2068 cc = cb->blksize * 2; 2069 error = fetcher->fetch_to(fetcher, &stream, cc); 2070 if (error != 0) { 2071 fetcher = &ufetcher.base; 2072 cc = sc->sc_pustream->end - sc->sc_pustream->start; 2073 error = fetcher->fetch_to(fetcher, sc->sc_pustream, cc); 2074 } 2075 } else { 2076 fetcher = &ufetcher.base; 2077 cc = stream.end - stream.start; 2078 error = fetcher->fetch_to(fetcher, &stream, cc); 2079 } 2080 sc->sc_writing = 0; 2081 if (sc->sc_waitcomp) 2082 wakeup(sc); 2083 2084 s = splaudio(); 2085 if (sc->sc_npfilters > 0) { 2086 cb->fstamp += ufetcher.last_used 2087 - audio_stream_get_used(sc->sc_pustream); 2088 } 2089 cb->s.used += stream.used - used; 2090 cb->s.inp = stream.inp; 2091 einp = cb->s.inp; 2092 2093 /* 2094 * This is a very suboptimal way of keeping track of 2095 * silence in the buffer, but it is simple. 2096 */ 2097 sc->sc_sil_count = 0; 2098 2099 /* 2100 * If the interrupt routine wants the last block filled AND 2101 * the copy did not fill the last block completely it needs to 2102 * be padded. 2103 */ 2104 if (cb->needfill && inp < einp && 2105 (inp - cb->s.start) / cb->blksize == 2106 (einp - cb->s.start) / cb->blksize) { 2107 /* Figure out how many bytes to a block boundary. */ 2108 cc = cb->blksize - (einp - cb->s.start) % cb->blksize; 2109 DPRINTF(("audio_write: partial fill %d\n", cc)); 2110 } else 2111 cc = 0; 2112 cb->needfill = false; 2113 cb->copying = false; 2114 if (!sc->sc_pbus && !cb->pause) { 2115 saveerror = error; 2116 error = audiostartp(sc); 2117 if (saveerror != 0) { 2118 /* Report the first error that occurred. */ 2119 error = saveerror; 2120 } 2121 } 2122 splx(s); 2123 if (cc != 0) { 2124 DPRINTFN(1, ("audio_write: fill %d\n", cc)); 2125 audio_fill_silence(&cb->s.param, einp, cc); 2126 } 2127 } 2128 2129 return error; 2130 } 2131 2132 int 2133 audio_ioctl(struct audio_softc *sc, u_long cmd, void *addr, int flag, 2134 struct lwp *l) 2135 { 2136 const struct audio_hw_if *hw; 2137 struct audio_offset *ao; 2138 u_long stamp; 2139 int error, s, offs, fd; 2140 bool rbus, pbus; 2141 2142 DPRINTF(("audio_ioctl(%lu,'%c',%lu)\n", 2143 IOCPARM_LEN(cmd), (char)IOCGROUP(cmd), cmd&0xff)); 2144 hw = sc->hw_if; 2145 error = 0; 2146 switch (cmd) { 2147 case FIONBIO: 2148 /* All handled in the upper FS layer. */ 2149 break; 2150 2151 case FIONREAD: 2152 *(int *)addr = audio_stream_get_used(sc->sc_rustream); 2153 break; 2154 2155 case FIOASYNC: 2156 if (*(int *)addr) { 2157 if (sc->sc_async_audio) 2158 return EBUSY; 2159 sc->sc_async_audio = l->l_proc; 2160 DPRINTF(("audio_ioctl: FIOASYNC %p\n", l->l_proc)); 2161 } else 2162 sc->sc_async_audio = 0; 2163 break; 2164 2165 case AUDIO_FLUSH: 2166 DPRINTF(("AUDIO_FLUSH\n")); 2167 rbus = sc->sc_rbus; 2168 pbus = sc->sc_pbus; 2169 audio_clear(sc); 2170 s = splaudio(); 2171 error = audio_initbufs(sc); 2172 if (error) { 2173 splx(s); 2174 return error; 2175 } 2176 if ((sc->sc_mode & AUMODE_PLAY) && !sc->sc_pbus && pbus) 2177 error = audiostartp(sc); 2178 if (!error && 2179 (sc->sc_mode & AUMODE_RECORD) && !sc->sc_rbus && rbus) 2180 error = audiostartr(sc); 2181 splx(s); 2182 break; 2183 2184 /* 2185 * Number of read (write) samples dropped. We don't know where or 2186 * when they were dropped. 2187 */ 2188 case AUDIO_RERROR: 2189 *(int *)addr = sc->sc_rr.drops; 2190 break; 2191 2192 case AUDIO_PERROR: 2193 *(int *)addr = sc->sc_pr.drops; 2194 break; 2195 2196 /* 2197 * Offsets into buffer. 2198 */ 2199 case AUDIO_GETIOFFS: 2200 ao = (struct audio_offset *)addr; 2201 s = splaudio(); 2202 /* figure out where next DMA will start */ 2203 stamp = sc->sc_rustream == &sc->sc_rr.s 2204 ? sc->sc_rr.stamp : sc->sc_rr.fstamp; 2205 offs = sc->sc_rustream->inp - sc->sc_rustream->start; 2206 splx(s); 2207 ao->samples = stamp; 2208 ao->deltablks = 2209 (stamp / sc->sc_rr.blksize) - 2210 (sc->sc_rr.stamp_last / sc->sc_rr.blksize); 2211 sc->sc_rr.stamp_last = stamp; 2212 ao->offset = offs; 2213 break; 2214 2215 case AUDIO_GETOOFFS: 2216 ao = (struct audio_offset *)addr; 2217 s = splaudio(); 2218 /* figure out where next DMA will start */ 2219 stamp = sc->sc_pustream == &sc->sc_pr.s 2220 ? sc->sc_pr.stamp : sc->sc_pr.fstamp; 2221 offs = sc->sc_pustream->outp - sc->sc_pustream->start 2222 + sc->sc_pr.blksize; 2223 splx(s); 2224 ao->samples = stamp; 2225 ao->deltablks = 2226 (stamp / sc->sc_pr.blksize) - 2227 (sc->sc_pr.stamp_last / sc->sc_pr.blksize); 2228 sc->sc_pr.stamp_last = stamp; 2229 if (sc->sc_pustream->start + offs >= sc->sc_pustream->end) 2230 offs = 0; 2231 ao->offset = offs; 2232 break; 2233 2234 /* 2235 * How many bytes will elapse until mike hears the first 2236 * sample of what we write next? 2237 */ 2238 case AUDIO_WSEEK: 2239 *(u_long *)addr = audio_stream_get_used(sc->sc_pustream); 2240 break; 2241 2242 case AUDIO_SETINFO: 2243 DPRINTF(("AUDIO_SETINFO mode=0x%x\n", sc->sc_mode)); 2244 error = audiosetinfo(sc, (struct audio_info *)addr); 2245 break; 2246 2247 case AUDIO_GETINFO: 2248 DPRINTF(("AUDIO_GETINFO\n")); 2249 error = audiogetinfo(sc, (struct audio_info *)addr, 0); 2250 break; 2251 2252 case AUDIO_GETBUFINFO: 2253 DPRINTF(("AUDIO_GETBUFINFO\n")); 2254 error = audiogetinfo(sc, (struct audio_info *)addr, 1); 2255 break; 2256 2257 case AUDIO_DRAIN: 2258 DPRINTF(("AUDIO_DRAIN\n")); 2259 error = audio_drain(sc); 2260 if (!error && hw->drain) 2261 error = hw->drain(sc->hw_hdl); 2262 break; 2263 2264 case AUDIO_GETDEV: 2265 DPRINTF(("AUDIO_GETDEV\n")); 2266 error = hw->getdev(sc->hw_hdl, (audio_device_t *)addr); 2267 break; 2268 2269 case AUDIO_GETENC: 2270 DPRINTF(("AUDIO_GETENC\n")); 2271 error = hw->query_encoding(sc->hw_hdl, 2272 (struct audio_encoding *)addr); 2273 break; 2274 2275 case AUDIO_GETFD: 2276 DPRINTF(("AUDIO_GETFD\n")); 2277 *(int *)addr = sc->sc_full_duplex; 2278 break; 2279 2280 case AUDIO_SETFD: 2281 DPRINTF(("AUDIO_SETFD\n")); 2282 fd = *(int *)addr; 2283 if (audio_get_props(sc) & AUDIO_PROP_FULLDUPLEX) { 2284 if (hw->setfd) 2285 error = hw->setfd(sc->hw_hdl, fd); 2286 else 2287 error = 0; 2288 if (!error) 2289 sc->sc_full_duplex = fd; 2290 } else { 2291 if (fd) 2292 error = ENOTTY; 2293 else 2294 error = 0; 2295 } 2296 break; 2297 2298 case AUDIO_GETPROPS: 2299 DPRINTF(("AUDIO_GETPROPS\n")); 2300 *(int *)addr = audio_get_props(sc); 2301 break; 2302 2303 default: 2304 if (hw->dev_ioctl) { 2305 error = hw->dev_ioctl(sc->hw_hdl, cmd, addr, flag, l); 2306 } else { 2307 DPRINTF(("audio_ioctl: unknown ioctl\n")); 2308 error = EINVAL; 2309 } 2310 break; 2311 } 2312 DPRINTF(("audio_ioctl(%lu,'%c',%lu) result %d\n", 2313 IOCPARM_LEN(cmd), (char)IOCGROUP(cmd), cmd&0xff, error)); 2314 return error; 2315 } 2316 2317 int 2318 audio_poll(struct audio_softc *sc, int events, struct lwp *l) 2319 { 2320 int revents; 2321 int s; 2322 int used; 2323 2324 DPRINTF(("audio_poll: events=0x%x mode=%d\n", events, sc->sc_mode)); 2325 2326 revents = 0; 2327 s = splaudio(); 2328 if (events & (POLLIN | POLLRDNORM)) { 2329 used = audio_stream_get_used(sc->sc_rustream); 2330 /* 2331 * If half duplex and playing, audio_read() will generate 2332 * silence at the play rate; poll for silence being 2333 * available. Otherwise, poll for recorded sound. 2334 */ 2335 if ((!sc->sc_full_duplex && (sc->sc_mode & AUMODE_PLAY)) ? 2336 sc->sc_pr.stamp > sc->sc_wstamp : 2337 used > sc->sc_rr.usedlow) 2338 revents |= events & (POLLIN | POLLRDNORM); 2339 } 2340 2341 if (events & (POLLOUT | POLLWRNORM)) { 2342 used = audio_stream_get_used(sc->sc_pustream); 2343 /* 2344 * If half duplex and recording, audio_write() will throw 2345 * away play data, which means we are always ready to write. 2346 * Otherwise, poll for play buffer being below its low water 2347 * mark. 2348 */ 2349 if ((!sc->sc_full_duplex && (sc->sc_mode & AUMODE_RECORD)) || 2350 (!(sc->sc_mode & AUMODE_PLAY_ALL) && sc->sc_playdrop > 0) || 2351 (used <= sc->sc_pr.usedlow)) 2352 revents |= events & (POLLOUT | POLLWRNORM); 2353 } 2354 2355 if (revents == 0) { 2356 if (events & (POLLIN | POLLRDNORM)) 2357 selrecord(l, &sc->sc_rsel); 2358 2359 if (events & (POLLOUT | POLLWRNORM)) 2360 selrecord(l, &sc->sc_wsel); 2361 } 2362 2363 splx(s); 2364 return revents; 2365 } 2366 2367 static void 2368 filt_audiordetach(struct knote *kn) 2369 { 2370 struct audio_softc *sc; 2371 int s; 2372 2373 sc = kn->kn_hook; 2374 s = splaudio(); 2375 SLIST_REMOVE(&sc->sc_rsel.sel_klist, kn, knote, kn_selnext); 2376 splx(s); 2377 } 2378 2379 static int 2380 filt_audioread(struct knote *kn, long hint) 2381 { 2382 struct audio_softc *sc; 2383 int s; 2384 2385 sc = kn->kn_hook; 2386 s = splaudio(); 2387 if (!sc->sc_full_duplex && (sc->sc_mode & AUMODE_PLAY)) 2388 kn->kn_data = sc->sc_pr.stamp - sc->sc_wstamp; 2389 else 2390 kn->kn_data = audio_stream_get_used(sc->sc_rustream) 2391 - sc->sc_rr.usedlow; 2392 splx(s); 2393 2394 return kn->kn_data > 0; 2395 } 2396 2397 static const struct filterops audioread_filtops = 2398 { 1, NULL, filt_audiordetach, filt_audioread }; 2399 2400 static void 2401 filt_audiowdetach(struct knote *kn) 2402 { 2403 struct audio_softc *sc; 2404 int s; 2405 2406 sc = kn->kn_hook; 2407 s = splaudio(); 2408 SLIST_REMOVE(&sc->sc_wsel.sel_klist, kn, knote, kn_selnext); 2409 splx(s); 2410 } 2411 2412 static int 2413 filt_audiowrite(struct knote *kn, long hint) 2414 { 2415 struct audio_softc *sc; 2416 audio_stream_t *stream; 2417 int s; 2418 2419 sc = kn->kn_hook; 2420 s = splaudio(); 2421 stream = sc->sc_pustream; 2422 kn->kn_data = (stream->end - stream->start) 2423 - audio_stream_get_used(stream); 2424 splx(s); 2425 2426 return kn->kn_data > 0; 2427 } 2428 2429 static const struct filterops audiowrite_filtops = 2430 { 1, NULL, filt_audiowdetach, filt_audiowrite }; 2431 2432 int 2433 audio_kqfilter(struct audio_softc *sc, struct knote *kn) 2434 { 2435 struct klist *klist; 2436 int s; 2437 2438 switch (kn->kn_filter) { 2439 case EVFILT_READ: 2440 klist = &sc->sc_rsel.sel_klist; 2441 kn->kn_fop = &audioread_filtops; 2442 break; 2443 2444 case EVFILT_WRITE: 2445 klist = &sc->sc_wsel.sel_klist; 2446 kn->kn_fop = &audiowrite_filtops; 2447 break; 2448 2449 default: 2450 return EINVAL; 2451 } 2452 2453 kn->kn_hook = sc; 2454 2455 s = splaudio(); 2456 SLIST_INSERT_HEAD(klist, kn, kn_selnext); 2457 splx(s); 2458 2459 return 0; 2460 } 2461 2462 paddr_t 2463 audio_mmap(struct audio_softc *sc, off_t off, int prot) 2464 { 2465 const struct audio_hw_if *hw; 2466 struct audio_ringbuffer *cb; 2467 int s; 2468 2469 DPRINTF(("audio_mmap: off=%lld, prot=%d\n", (long long)off, prot)); 2470 hw = sc->hw_if; 2471 if (!(audio_get_props(sc) & AUDIO_PROP_MMAP) || !hw->mappage) 2472 return -1; 2473 #if 0 2474 /* XXX 2475 * The idea here was to use the protection to determine if 2476 * we are mapping the read or write buffer, but it fails. 2477 * The VM system is broken in (at least) two ways. 2478 * 1) If you map memory VM_PROT_WRITE you SIGSEGV 2479 * when writing to it, so VM_PROT_READ|VM_PROT_WRITE 2480 * has to be used for mmapping the play buffer. 2481 * 2) Even if calling mmap() with VM_PROT_READ|VM_PROT_WRITE 2482 * audio_mmap will get called at some point with VM_PROT_READ 2483 * only. 2484 * So, alas, we always map the play buffer for now. 2485 */ 2486 if (prot == (VM_PROT_READ|VM_PROT_WRITE) || 2487 prot == VM_PROT_WRITE) 2488 cb = &sc->sc_pr; 2489 else if (prot == VM_PROT_READ) 2490 cb = &sc->sc_rr; 2491 else 2492 return -1; 2493 #else 2494 cb = &sc->sc_pr; 2495 #endif 2496 2497 if ((u_int)off >= cb->s.bufsize) 2498 return -1; 2499 if (!cb->mmapped) { 2500 cb->mmapped = true; 2501 if (cb == &sc->sc_pr) { 2502 audio_fill_silence(&cb->s.param, cb->s.start, 2503 cb->s.bufsize); 2504 s = splaudio(); 2505 sc->sc_pustream = &cb->s; 2506 if (!sc->sc_pbus && !sc->sc_pr.pause) 2507 (void)audiostartp(sc); 2508 splx(s); 2509 } else { 2510 s = splaudio(); 2511 sc->sc_rustream = &cb->s; 2512 if (!sc->sc_rbus && !sc->sc_rr.pause) 2513 (void)audiostartr(sc); 2514 splx(s); 2515 } 2516 } 2517 2518 return hw->mappage(sc->hw_hdl, cb->s.start, off, prot); 2519 } 2520 2521 int 2522 audiostartr(struct audio_softc *sc) 2523 { 2524 int error; 2525 2526 DPRINTF(("audiostartr: start=%p used=%d(hi=%d) mmapped=%d\n", 2527 sc->sc_rr.s.start, audio_stream_get_used(&sc->sc_rr.s), 2528 sc->sc_rr.usedhigh, sc->sc_rr.mmapped)); 2529 2530 if (!audio_can_capture(sc)) 2531 return EINVAL; 2532 2533 if (sc->hw_if->trigger_input) 2534 error = sc->hw_if->trigger_input(sc->hw_hdl, sc->sc_rr.s.start, 2535 sc->sc_rr.s.end, sc->sc_rr.blksize, 2536 audio_rint, (void *)sc, &sc->sc_rr.s.param); 2537 else 2538 error = sc->hw_if->start_input(sc->hw_hdl, sc->sc_rr.s.start, 2539 sc->sc_rr.blksize, audio_rint, (void *)sc); 2540 if (error) { 2541 DPRINTF(("audiostartr failed: %d\n", error)); 2542 return error; 2543 } 2544 sc->sc_rbus = true; 2545 return 0; 2546 } 2547 2548 int 2549 audiostartp(struct audio_softc *sc) 2550 { 2551 int error; 2552 int used; 2553 2554 used = audio_stream_get_used(&sc->sc_pr.s); 2555 DPRINTF(("audiostartp: start=%p used=%d(hi=%d blk=%d) mmapped=%d\n", 2556 sc->sc_pr.s.start, used, sc->sc_pr.usedhigh, 2557 sc->sc_pr.blksize, sc->sc_pr.mmapped)); 2558 2559 if (!audio_can_playback(sc)) 2560 return EINVAL; 2561 2562 if (!sc->sc_pr.mmapped && used < sc->sc_pr.blksize) { 2563 wakeup(&sc->sc_wchan); 2564 DPRINTF(("%s: wakeup and return\n", __func__)); 2565 return 0; 2566 } 2567 2568 if (sc->hw_if->trigger_output) { 2569 DPRINTF(("%s: call trigger_output\n", __func__)); 2570 error = sc->hw_if->trigger_output(sc->hw_hdl, sc->sc_pr.s.start, 2571 sc->sc_pr.s.end, sc->sc_pr.blksize, 2572 audio_pint, (void *)sc, &sc->sc_pr.s.param); 2573 } else { 2574 DPRINTF(("%s: call start_output\n", __func__)); 2575 error = sc->hw_if->start_output(sc->hw_hdl, 2576 __UNCONST(sc->sc_pr.s.outp), sc->sc_pr.blksize, 2577 audio_pint, (void *)sc); 2578 } 2579 if (error) { 2580 DPRINTF(("audiostartp failed: %d\n", error)); 2581 return error; 2582 } 2583 sc->sc_pbus = true; 2584 return 0; 2585 } 2586 2587 /* 2588 * When the play interrupt routine finds that the write isn't keeping 2589 * the buffer filled it will insert silence in the buffer to make up 2590 * for this. The part of the buffer that is filled with silence 2591 * is kept track of in a very approximate way: it starts at sc_sil_start 2592 * and extends sc_sil_count bytes. If there is already silence in 2593 * the requested area nothing is done; so when the whole buffer is 2594 * silent nothing happens. When the writer starts again sc_sil_count 2595 * is set to 0. 2596 */ 2597 /* XXX 2598 * Putting silence into the output buffer should not really be done 2599 * at splaudio, but there is no softaudio level to do it at yet. 2600 */ 2601 static inline void 2602 audio_pint_silence(struct audio_softc *sc, struct audio_ringbuffer *cb, 2603 uint8_t *inp, int cc) 2604 { 2605 uint8_t *s, *e, *p, *q; 2606 2607 if (sc->sc_sil_count > 0) { 2608 s = sc->sc_sil_start; /* start of silence */ 2609 e = s + sc->sc_sil_count; /* end of sil., may be beyond end */ 2610 p = inp; /* adjusted pointer to area to fill */ 2611 if (p < s) 2612 p += cb->s.end - cb->s.start; 2613 q = p + cc; 2614 /* Check if there is already silence. */ 2615 if (!(s <= p && p < e && 2616 s <= q && q <= e)) { 2617 if (s <= p) 2618 sc->sc_sil_count = max(sc->sc_sil_count, q-s); 2619 DPRINTFN(5,("audio_pint_silence: fill cc=%d inp=%p, " 2620 "count=%d size=%d\n", 2621 cc, inp, sc->sc_sil_count, 2622 (int)(cb->s.end - cb->s.start))); 2623 audio_fill_silence(&cb->s.param, inp, cc); 2624 } else { 2625 DPRINTFN(5,("audio_pint_silence: already silent " 2626 "cc=%d inp=%p\n", cc, inp)); 2627 2628 } 2629 } else { 2630 sc->sc_sil_start = inp; 2631 sc->sc_sil_count = cc; 2632 DPRINTFN(5, ("audio_pint_silence: start fill %p %d\n", 2633 inp, cc)); 2634 audio_fill_silence(&cb->s.param, inp, cc); 2635 } 2636 } 2637 2638 static void 2639 audio_softintr_rd(void *cookie) 2640 { 2641 struct audio_softc *sc = cookie; 2642 struct proc *p; 2643 2644 audio_wakeup(&sc->sc_rchan); 2645 selnotify(&sc->sc_rsel, 0, 0); 2646 if (sc->sc_async_audio != NULL) { 2647 DPRINTFN(3, ("audio_softintr_rd: sending SIGIO %p\n", 2648 sc->sc_async_audio)); 2649 mutex_enter(proc_lock); 2650 if ((p = sc->sc_async_audio) != NULL) 2651 psignal(p, SIGIO); 2652 mutex_exit(proc_lock); 2653 } 2654 } 2655 2656 static void 2657 audio_softintr_wr(void *cookie) 2658 { 2659 struct audio_softc *sc = cookie; 2660 struct proc *p; 2661 2662 audio_wakeup(&sc->sc_wchan); 2663 selnotify(&sc->sc_wsel, 0, 0); 2664 if (sc->sc_async_audio != NULL) { 2665 DPRINTFN(3, ("audio_softintr_wr: sending SIGIO %p\n", 2666 sc->sc_async_audio)); 2667 mutex_enter(proc_lock); 2668 if ((p = sc->sc_async_audio) != NULL) 2669 psignal(p, SIGIO); 2670 mutex_exit(proc_lock); 2671 } 2672 } 2673 2674 /* 2675 * Called from HW driver module on completion of DMA output. 2676 * Start output of new block, wrap in ring buffer if needed. 2677 * If no more buffers to play, output zero instead. 2678 * Do a wakeup if necessary. 2679 */ 2680 void 2681 audio_pint(void *v) 2682 { 2683 stream_fetcher_t null_fetcher; 2684 struct audio_softc *sc; 2685 const struct audio_hw_if *hw; 2686 struct audio_ringbuffer *cb; 2687 stream_fetcher_t *fetcher; 2688 uint8_t *inp; 2689 int cc, used; 2690 int blksize; 2691 int error; 2692 2693 sc = v; 2694 if (!sc->sc_open) 2695 return; /* ignore interrupt if not open */ 2696 2697 hw = sc->hw_if; 2698 cb = &sc->sc_pr; 2699 blksize = cb->blksize; 2700 cb->s.outp = audio_stream_add_outp(&cb->s, cb->s.outp, blksize); 2701 cb->stamp += blksize; 2702 if (cb->mmapped) { 2703 DPRINTFN(5, ("audio_pint: mmapped outp=%p cc=%d inp=%p\n", 2704 cb->s.outp, blksize, cb->s.inp)); 2705 if (hw->trigger_output == NULL) 2706 (void)hw->start_output(sc->hw_hdl, __UNCONST(cb->s.outp), 2707 blksize, audio_pint, (void *)sc); 2708 return; 2709 } 2710 2711 #ifdef AUDIO_INTR_TIME 2712 { 2713 struct timeval tv; 2714 u_long t; 2715 microtime(&tv); 2716 t = tv.tv_usec + 1000000 * tv.tv_sec; 2717 if (sc->sc_pnintr) { 2718 long lastdelta, totdelta; 2719 lastdelta = t - sc->sc_plastintr - sc->sc_pblktime; 2720 if (lastdelta > sc->sc_pblktime / 3) { 2721 printf("audio: play interrupt(%d) off " 2722 "relative by %ld us (%lu)\n", 2723 sc->sc_pnintr, lastdelta, 2724 sc->sc_pblktime); 2725 } 2726 totdelta = t - sc->sc_pfirstintr - 2727 sc->sc_pblktime * sc->sc_pnintr; 2728 if (totdelta > sc->sc_pblktime) { 2729 printf("audio: play interrupt(%d) off " 2730 "absolute by %ld us (%lu) (LOST)\n", 2731 sc->sc_pnintr, totdelta, 2732 sc->sc_pblktime); 2733 sc->sc_pnintr++; /* avoid repeated messages */ 2734 } 2735 } else 2736 sc->sc_pfirstintr = t; 2737 sc->sc_plastintr = t; 2738 sc->sc_pnintr++; 2739 } 2740 #endif 2741 2742 used = audio_stream_get_used(&cb->s); 2743 /* 2744 * "used <= cb->usedlow" should be "used < blksize" ideally. 2745 * Some HW drivers such as uaudio(4) does not call audio_pint() 2746 * at accurate timing. If used < blksize, uaudio(4) already 2747 * request transfer of garbage data. 2748 */ 2749 if (used <= cb->usedlow && !cb->copying && sc->sc_npfilters > 0) { 2750 /* we might have data in filter pipeline */ 2751 null_fetcher.fetch_to = null_fetcher_fetch_to; 2752 fetcher = &sc->sc_pfilters[sc->sc_npfilters - 1]->base; 2753 sc->sc_pfilters[0]->set_fetcher(sc->sc_pfilters[0], 2754 &null_fetcher); 2755 used = audio_stream_get_used(sc->sc_pustream); 2756 cc = cb->s.end - cb->s.start; 2757 if (blksize * 2 < cc) 2758 cc = blksize * 2; 2759 fetcher->fetch_to(fetcher, &cb->s, cc); 2760 cb->fstamp += used - audio_stream_get_used(sc->sc_pustream); 2761 used = audio_stream_get_used(&cb->s); 2762 } 2763 if (used < blksize) { 2764 /* we don't have a full block to use */ 2765 if (cb->copying) { 2766 /* writer is in progress, don't disturb */ 2767 cb->needfill = true; 2768 DPRINTFN(1, ("audio_pint: copying in progress\n")); 2769 } else { 2770 inp = cb->s.inp; 2771 cc = blksize - (inp - cb->s.start) % blksize; 2772 if (cb->pause) 2773 cb->pdrops += cc; 2774 else { 2775 cb->drops += cc; 2776 sc->sc_playdrop += cc; 2777 } 2778 audio_pint_silence(sc, cb, inp, cc); 2779 cb->s.inp = audio_stream_add_inp(&cb->s, inp, cc); 2780 2781 /* Clear next block so we keep ahead of the DMA. */ 2782 used = audio_stream_get_used(&cb->s); 2783 if (used + blksize < cb->s.end - cb->s.start) 2784 audio_pint_silence(sc, cb, cb->s.inp, blksize); 2785 } 2786 } 2787 2788 DPRINTFN(5, ("audio_pint: outp=%p cc=%d\n", cb->s.outp, blksize)); 2789 if (hw->trigger_output == NULL) { 2790 error = hw->start_output(sc->hw_hdl, __UNCONST(cb->s.outp), 2791 blksize, audio_pint, (void *)sc); 2792 if (error) { 2793 /* XXX does this really help? */ 2794 DPRINTF(("audio_pint restart failed: %d\n", error)); 2795 audio_clear(sc); 2796 } 2797 } 2798 2799 DPRINTFN(2, ("audio_pint: mode=%d pause=%d used=%d lowat=%d\n", 2800 sc->sc_mode, cb->pause, 2801 audio_stream_get_used(sc->sc_pustream), cb->usedlow)); 2802 if ((sc->sc_mode & AUMODE_PLAY) && !cb->pause) { 2803 if (audio_stream_get_used(sc->sc_pustream) <= cb->usedlow) 2804 softint_schedule(sc->sc_sih_wr); 2805 } 2806 2807 /* Possible to return one or more "phantom blocks" now. */ 2808 if (!sc->sc_full_duplex && sc->sc_rchan) 2809 softint_schedule(sc->sc_sih_rd); 2810 } 2811 2812 /* 2813 * Called from HW driver module on completion of DMA input. 2814 * Mark it as input in the ring buffer (fiddle pointers). 2815 * Do a wakeup if necessary. 2816 */ 2817 void 2818 audio_rint(void *v) 2819 { 2820 stream_fetcher_t null_fetcher; 2821 struct audio_softc *sc; 2822 const struct audio_hw_if *hw; 2823 struct audio_ringbuffer *cb; 2824 stream_fetcher_t *last_fetcher; 2825 int cc; 2826 int used; 2827 int blksize; 2828 int error; 2829 2830 sc = v; 2831 cb = &sc->sc_rr; 2832 if (!sc->sc_open) 2833 return; /* ignore interrupt if not open */ 2834 2835 hw = sc->hw_if; 2836 blksize = cb->blksize; 2837 cb->s.inp = audio_stream_add_inp(&cb->s, cb->s.inp, blksize); 2838 cb->stamp += blksize; 2839 if (cb->mmapped) { 2840 DPRINTFN(2, ("audio_rint: mmapped inp=%p cc=%d\n", 2841 cb->s.inp, blksize)); 2842 if (hw->trigger_input == NULL) 2843 (void)hw->start_input(sc->hw_hdl, cb->s.inp, blksize, 2844 audio_rint, (void *)sc); 2845 return; 2846 } 2847 2848 #ifdef AUDIO_INTR_TIME 2849 { 2850 struct timeval tv; 2851 u_long t; 2852 microtime(&tv); 2853 t = tv.tv_usec + 1000000 * tv.tv_sec; 2854 if (sc->sc_rnintr) { 2855 long lastdelta, totdelta; 2856 lastdelta = t - sc->sc_rlastintr - sc->sc_rblktime; 2857 if (lastdelta > sc->sc_rblktime / 5) { 2858 printf("audio: record interrupt(%d) off " 2859 "relative by %ld us (%lu)\n", 2860 sc->sc_rnintr, lastdelta, 2861 sc->sc_rblktime); 2862 } 2863 totdelta = t - sc->sc_rfirstintr - 2864 sc->sc_rblktime * sc->sc_rnintr; 2865 if (totdelta > sc->sc_rblktime / 2) { 2866 sc->sc_rnintr++; 2867 printf("audio: record interrupt(%d) off " 2868 "absolute by %ld us (%lu)\n", 2869 sc->sc_rnintr, totdelta, 2870 sc->sc_rblktime); 2871 sc->sc_rnintr++; /* avoid repeated messages */ 2872 } 2873 } else 2874 sc->sc_rfirstintr = t; 2875 sc->sc_rlastintr = t; 2876 sc->sc_rnintr++; 2877 } 2878 #endif 2879 2880 if (!cb->pause && sc->sc_nrfilters > 0) { 2881 null_fetcher.fetch_to = null_fetcher_fetch_to; 2882 last_fetcher = &sc->sc_rfilters[sc->sc_nrfilters - 1]->base; 2883 sc->sc_rfilters[0]->set_fetcher(sc->sc_rfilters[0], 2884 &null_fetcher); 2885 used = audio_stream_get_used(sc->sc_rustream); 2886 cc = sc->sc_rustream->end - sc->sc_rustream->start; 2887 error = last_fetcher->fetch_to 2888 (last_fetcher, sc->sc_rustream, cc); 2889 cb->fstamp += audio_stream_get_used(sc->sc_rustream) - used; 2890 /* XXX what should do for error? */ 2891 } 2892 used = audio_stream_get_used(&sc->sc_rr.s); 2893 if (cb->pause) { 2894 DPRINTFN(1, ("audio_rint: pdrops %lu\n", cb->pdrops)); 2895 cb->pdrops += blksize; 2896 cb->s.outp = audio_stream_add_outp(&cb->s, cb->s.outp, blksize); 2897 } else if (used + blksize > cb->s.end - cb->s.start && !cb->copying) { 2898 DPRINTFN(1, ("audio_rint: drops %lu\n", cb->drops)); 2899 cb->drops += blksize; 2900 cb->s.outp = audio_stream_add_outp(&cb->s, cb->s.outp, blksize); 2901 } 2902 2903 DPRINTFN(2, ("audio_rint: inp=%p cc=%d\n", cb->s.inp, blksize)); 2904 if (hw->trigger_input == NULL) { 2905 error = hw->start_input(sc->hw_hdl, cb->s.inp, blksize, 2906 audio_rint, (void *)sc); 2907 if (error) { 2908 /* XXX does this really help? */ 2909 DPRINTF(("audio_rint: restart failed: %d\n", error)); 2910 audio_clear(sc); 2911 } 2912 } 2913 2914 softint_schedule(sc->sc_sih_rd); 2915 } 2916 2917 int 2918 audio_check_params(struct audio_params *p) 2919 { 2920 2921 if (p->encoding == AUDIO_ENCODING_PCM16) { 2922 if (p->precision == 8) 2923 p->encoding = AUDIO_ENCODING_ULINEAR; 2924 else 2925 p->encoding = AUDIO_ENCODING_SLINEAR; 2926 } else if (p->encoding == AUDIO_ENCODING_PCM8) { 2927 if (p->precision == 8) 2928 p->encoding = AUDIO_ENCODING_ULINEAR; 2929 else 2930 return EINVAL; 2931 } 2932 2933 if (p->encoding == AUDIO_ENCODING_SLINEAR) 2934 #if BYTE_ORDER == LITTLE_ENDIAN 2935 p->encoding = AUDIO_ENCODING_SLINEAR_LE; 2936 #else 2937 p->encoding = AUDIO_ENCODING_SLINEAR_BE; 2938 #endif 2939 if (p->encoding == AUDIO_ENCODING_ULINEAR) 2940 #if BYTE_ORDER == LITTLE_ENDIAN 2941 p->encoding = AUDIO_ENCODING_ULINEAR_LE; 2942 #else 2943 p->encoding = AUDIO_ENCODING_ULINEAR_BE; 2944 #endif 2945 2946 switch (p->encoding) { 2947 case AUDIO_ENCODING_ULAW: 2948 case AUDIO_ENCODING_ALAW: 2949 if (p->precision != 8) 2950 return EINVAL; 2951 break; 2952 case AUDIO_ENCODING_ADPCM: 2953 if (p->precision != 4 && p->precision != 8) 2954 return EINVAL; 2955 break; 2956 case AUDIO_ENCODING_SLINEAR_LE: 2957 case AUDIO_ENCODING_SLINEAR_BE: 2958 case AUDIO_ENCODING_ULINEAR_LE: 2959 case AUDIO_ENCODING_ULINEAR_BE: 2960 /* XXX is: our zero-fill can handle any multiple of 8 */ 2961 if (p->precision != 8 && p->precision != 16 && 2962 p->precision != 24 && p->precision != 32) 2963 return EINVAL; 2964 if (p->precision == 8 && p->encoding == AUDIO_ENCODING_SLINEAR_BE) 2965 p->encoding = AUDIO_ENCODING_SLINEAR_LE; 2966 if (p->precision == 8 && p->encoding == AUDIO_ENCODING_ULINEAR_BE) 2967 p->encoding = AUDIO_ENCODING_ULINEAR_LE; 2968 if (p->validbits > p->precision) 2969 return EINVAL; 2970 break; 2971 case AUDIO_ENCODING_MPEG_L1_STREAM: 2972 case AUDIO_ENCODING_MPEG_L1_PACKETS: 2973 case AUDIO_ENCODING_MPEG_L1_SYSTEM: 2974 case AUDIO_ENCODING_MPEG_L2_STREAM: 2975 case AUDIO_ENCODING_MPEG_L2_PACKETS: 2976 case AUDIO_ENCODING_MPEG_L2_SYSTEM: 2977 break; 2978 default: 2979 return EINVAL; 2980 } 2981 2982 /* sanity check # of channels*/ 2983 if (p->channels < 1 || p->channels > AUDIO_MAX_CHANNELS) 2984 return EINVAL; 2985 2986 return 0; 2987 } 2988 2989 int 2990 audio_set_defaults(struct audio_softc *sc, u_int mode) 2991 { 2992 struct audio_info ai; 2993 2994 /* default parameters */ 2995 sc->sc_rparams = audio_default; 2996 sc->sc_pparams = audio_default; 2997 sc->sc_blkset = false; 2998 2999 AUDIO_INITINFO(&ai); 3000 ai.record.sample_rate = sc->sc_rparams.sample_rate; 3001 ai.record.encoding = sc->sc_rparams.encoding; 3002 ai.record.channels = sc->sc_rparams.channels; 3003 ai.record.precision = sc->sc_rparams.precision; 3004 ai.record.pause = false; 3005 ai.play.sample_rate = sc->sc_pparams.sample_rate; 3006 ai.play.encoding = sc->sc_pparams.encoding; 3007 ai.play.channels = sc->sc_pparams.channels; 3008 ai.play.precision = sc->sc_pparams.precision; 3009 ai.play.pause = false; 3010 ai.mode = mode; 3011 3012 return audiosetinfo(sc, &ai); 3013 } 3014 3015 int 3016 au_set_lr_value(struct audio_softc *sc, mixer_ctrl_t *ct, int l, int r) 3017 { 3018 3019 ct->type = AUDIO_MIXER_VALUE; 3020 ct->un.value.num_channels = 2; 3021 ct->un.value.level[AUDIO_MIXER_LEVEL_LEFT] = l; 3022 ct->un.value.level[AUDIO_MIXER_LEVEL_RIGHT] = r; 3023 if (sc->hw_if->set_port(sc->hw_hdl, ct) == 0) 3024 return 0; 3025 ct->un.value.num_channels = 1; 3026 ct->un.value.level[AUDIO_MIXER_LEVEL_MONO] = (l+r)/2; 3027 return sc->hw_if->set_port(sc->hw_hdl, ct); 3028 } 3029 3030 int 3031 au_set_gain(struct audio_softc *sc, struct au_mixer_ports *ports, 3032 int gain, int balance) 3033 { 3034 mixer_ctrl_t ct; 3035 int i, error; 3036 int l, r; 3037 u_int mask; 3038 int nset; 3039 3040 if (balance == AUDIO_MID_BALANCE) { 3041 l = r = gain; 3042 } else if (balance < AUDIO_MID_BALANCE) { 3043 l = gain; 3044 r = (balance * gain) / AUDIO_MID_BALANCE; 3045 } else { 3046 r = gain; 3047 l = ((AUDIO_RIGHT_BALANCE - balance) * gain) 3048 / AUDIO_MID_BALANCE; 3049 } 3050 DPRINTF(("au_set_gain: gain=%d balance=%d, l=%d r=%d\n", 3051 gain, balance, l, r)); 3052 3053 if (ports->index == -1) { 3054 usemaster: 3055 if (ports->master == -1) 3056 return 0; /* just ignore it silently */ 3057 ct.dev = ports->master; 3058 error = au_set_lr_value(sc, &ct, l, r); 3059 } else { 3060 ct.dev = ports->index; 3061 if (ports->isenum) { 3062 ct.type = AUDIO_MIXER_ENUM; 3063 error = sc->hw_if->get_port(sc->hw_hdl, &ct); 3064 if (error) 3065 return error; 3066 if (ports->isdual) { 3067 if (ports->cur_port == -1) 3068 ct.dev = ports->master; 3069 else 3070 ct.dev = ports->miport[ports->cur_port]; 3071 error = au_set_lr_value(sc, &ct, l, r); 3072 } else { 3073 for(i = 0; i < ports->nports; i++) 3074 if (ports->misel[i] == ct.un.ord) { 3075 ct.dev = ports->miport[i]; 3076 if (ct.dev == -1 || 3077 au_set_lr_value(sc, &ct, l, r)) 3078 goto usemaster; 3079 else 3080 break; 3081 } 3082 } 3083 } else { 3084 ct.type = AUDIO_MIXER_SET; 3085 error = sc->hw_if->get_port(sc->hw_hdl, &ct); 3086 if (error) 3087 return error; 3088 mask = ct.un.mask; 3089 nset = 0; 3090 for(i = 0; i < ports->nports; i++) { 3091 if (ports->misel[i] & mask) { 3092 ct.dev = ports->miport[i]; 3093 if (ct.dev != -1 && 3094 au_set_lr_value(sc, &ct, l, r) == 0) 3095 nset++; 3096 } 3097 } 3098 if (nset == 0) 3099 goto usemaster; 3100 } 3101 } 3102 if (!error) 3103 mixer_signal(sc); 3104 return error; 3105 } 3106 3107 int 3108 au_get_lr_value(struct audio_softc *sc, mixer_ctrl_t *ct, int *l, int *r) 3109 { 3110 int error; 3111 3112 ct->un.value.num_channels = 2; 3113 if (sc->hw_if->get_port(sc->hw_hdl, ct) == 0) { 3114 *l = ct->un.value.level[AUDIO_MIXER_LEVEL_LEFT]; 3115 *r = ct->un.value.level[AUDIO_MIXER_LEVEL_RIGHT]; 3116 } else { 3117 ct->un.value.num_channels = 1; 3118 error = sc->hw_if->get_port(sc->hw_hdl, ct); 3119 if (error) 3120 return error; 3121 *r = *l = ct->un.value.level[AUDIO_MIXER_LEVEL_MONO]; 3122 } 3123 return 0; 3124 } 3125 3126 void 3127 au_get_gain(struct audio_softc *sc, struct au_mixer_ports *ports, 3128 u_int *pgain, u_char *pbalance) 3129 { 3130 mixer_ctrl_t ct; 3131 int i, l, r, n; 3132 int lgain, rgain; 3133 3134 lgain = AUDIO_MAX_GAIN / 2; 3135 rgain = AUDIO_MAX_GAIN / 2; 3136 if (ports->index == -1) { 3137 usemaster: 3138 if (ports->master == -1) 3139 goto bad; 3140 ct.dev = ports->master; 3141 ct.type = AUDIO_MIXER_VALUE; 3142 if (au_get_lr_value(sc, &ct, &lgain, &rgain)) 3143 goto bad; 3144 } else { 3145 ct.dev = ports->index; 3146 if (ports->isenum) { 3147 ct.type = AUDIO_MIXER_ENUM; 3148 if (sc->hw_if->get_port(sc->hw_hdl, &ct)) 3149 goto bad; 3150 ct.type = AUDIO_MIXER_VALUE; 3151 if (ports->isdual) { 3152 if (ports->cur_port == -1) 3153 ct.dev = ports->master; 3154 else 3155 ct.dev = ports->miport[ports->cur_port]; 3156 au_get_lr_value(sc, &ct, &lgain, &rgain); 3157 } else { 3158 for(i = 0; i < ports->nports; i++) 3159 if (ports->misel[i] == ct.un.ord) { 3160 ct.dev = ports->miport[i]; 3161 if (ct.dev == -1 || 3162 au_get_lr_value(sc, &ct, 3163 &lgain, &rgain)) 3164 goto usemaster; 3165 else 3166 break; 3167 } 3168 } 3169 } else { 3170 ct.type = AUDIO_MIXER_SET; 3171 if (sc->hw_if->get_port(sc->hw_hdl, &ct)) 3172 goto bad; 3173 ct.type = AUDIO_MIXER_VALUE; 3174 lgain = rgain = n = 0; 3175 for(i = 0; i < ports->nports; i++) { 3176 if (ports->misel[i] & ct.un.mask) { 3177 ct.dev = ports->miport[i]; 3178 if (ct.dev == -1 || 3179 au_get_lr_value(sc, &ct, &l, &r)) 3180 goto usemaster; 3181 else { 3182 lgain += l; 3183 rgain += r; 3184 n++; 3185 } 3186 } 3187 } 3188 if (n != 0) { 3189 lgain /= n; 3190 rgain /= n; 3191 } 3192 } 3193 } 3194 bad: 3195 if (lgain == rgain) { /* handles lgain==rgain==0 */ 3196 *pgain = lgain; 3197 *pbalance = AUDIO_MID_BALANCE; 3198 } else if (lgain < rgain) { 3199 *pgain = rgain; 3200 /* balance should be > AUDIO_MID_BALANCE */ 3201 *pbalance = AUDIO_RIGHT_BALANCE - 3202 (AUDIO_MID_BALANCE * lgain) / rgain; 3203 } else /* lgain > rgain */ { 3204 *pgain = lgain; 3205 /* balance should be < AUDIO_MID_BALANCE */ 3206 *pbalance = (AUDIO_MID_BALANCE * rgain) / lgain; 3207 } 3208 } 3209 3210 int 3211 au_set_port(struct audio_softc *sc, struct au_mixer_ports *ports, u_int port) 3212 { 3213 mixer_ctrl_t ct; 3214 int i, error, use_mixerout; 3215 3216 use_mixerout = 1; 3217 if (port == 0) { 3218 if (ports->allports == 0) 3219 return 0; /* Allow this special case. */ 3220 else if (ports->isdual) { 3221 if (ports->cur_port == -1) { 3222 return 0; 3223 } else { 3224 port = ports->aumask[ports->cur_port]; 3225 ports->cur_port = -1; 3226 use_mixerout = 0; 3227 } 3228 } 3229 } 3230 if (ports->index == -1) 3231 return EINVAL; 3232 ct.dev = ports->index; 3233 if (ports->isenum) { 3234 if (port & (port-1)) 3235 return EINVAL; /* Only one port allowed */ 3236 ct.type = AUDIO_MIXER_ENUM; 3237 error = EINVAL; 3238 for(i = 0; i < ports->nports; i++) 3239 if (ports->aumask[i] == port) { 3240 if (ports->isdual && use_mixerout) { 3241 ct.un.ord = ports->mixerout; 3242 ports->cur_port = i; 3243 } else { 3244 ct.un.ord = ports->misel[i]; 3245 } 3246 error = sc->hw_if->set_port(sc->hw_hdl, &ct); 3247 break; 3248 } 3249 } else { 3250 ct.type = AUDIO_MIXER_SET; 3251 ct.un.mask = 0; 3252 for(i = 0; i < ports->nports; i++) 3253 if (ports->aumask[i] & port) 3254 ct.un.mask |= ports->misel[i]; 3255 if (port != 0 && ct.un.mask == 0) 3256 error = EINVAL; 3257 else 3258 error = sc->hw_if->set_port(sc->hw_hdl, &ct); 3259 } 3260 if (!error) 3261 mixer_signal(sc); 3262 return error; 3263 } 3264 3265 int 3266 au_get_port(struct audio_softc *sc, struct au_mixer_ports *ports) 3267 { 3268 mixer_ctrl_t ct; 3269 int i, aumask; 3270 3271 if (ports->index == -1) 3272 return 0; 3273 ct.dev = ports->index; 3274 ct.type = ports->isenum ? AUDIO_MIXER_ENUM : AUDIO_MIXER_SET; 3275 if (sc->hw_if->get_port(sc->hw_hdl, &ct)) 3276 return 0; 3277 aumask = 0; 3278 if (ports->isenum) { 3279 if (ports->isdual && ports->cur_port != -1) { 3280 if (ports->mixerout == ct.un.ord) 3281 aumask = ports->aumask[ports->cur_port]; 3282 else 3283 ports->cur_port = -1; 3284 } 3285 if (aumask == 0) 3286 for(i = 0; i < ports->nports; i++) 3287 if (ports->misel[i] == ct.un.ord) 3288 aumask = ports->aumask[i]; 3289 } else { 3290 for(i = 0; i < ports->nports; i++) 3291 if (ct.un.mask & ports->misel[i]) 3292 aumask |= ports->aumask[i]; 3293 } 3294 return aumask; 3295 } 3296 3297 int 3298 audiosetinfo(struct audio_softc *sc, struct audio_info *ai) 3299 { 3300 stream_filter_list_t pfilters, rfilters; 3301 audio_params_t pp, rp; 3302 struct audio_prinfo *r, *p; 3303 const struct audio_hw_if *hw; 3304 audio_stream_t *oldpus, *oldrus; 3305 int s, setmode; 3306 int error; 3307 int np, nr; 3308 unsigned int blks; 3309 int oldpblksize, oldrblksize; 3310 u_int gain; 3311 bool rbus, pbus; 3312 bool cleared, modechange, pausechange; 3313 u_char balance; 3314 3315 hw = sc->hw_if; 3316 if (hw == NULL) /* HW has not attached */ 3317 return ENXIO; 3318 3319 DPRINTF(("%s sc=%p ai=%p\n", __func__, sc, ai)); 3320 r = &ai->record; 3321 p = &ai->play; 3322 rbus = sc->sc_rbus; 3323 pbus = sc->sc_pbus; 3324 error = 0; 3325 cleared = false; 3326 modechange = false; 3327 pausechange = false; 3328 3329 pp = sc->sc_pparams; /* Temporary encoding storage in */ 3330 rp = sc->sc_rparams; /* case setting the modes fails. */ 3331 nr = np = 0; 3332 3333 if (SPECIFIED(p->sample_rate)) { 3334 pp.sample_rate = p->sample_rate; 3335 np++; 3336 } 3337 if (SPECIFIED(r->sample_rate)) { 3338 rp.sample_rate = r->sample_rate; 3339 nr++; 3340 } 3341 if (SPECIFIED(p->encoding)) { 3342 pp.encoding = p->encoding; 3343 np++; 3344 } 3345 if (SPECIFIED(r->encoding)) { 3346 rp.encoding = r->encoding; 3347 nr++; 3348 } 3349 if (SPECIFIED(p->precision)) { 3350 pp.precision = p->precision; 3351 /* we don't have API to specify validbits */ 3352 pp.validbits = p->precision; 3353 np++; 3354 } 3355 if (SPECIFIED(r->precision)) { 3356 rp.precision = r->precision; 3357 /* we don't have API to specify validbits */ 3358 rp.validbits = r->precision; 3359 nr++; 3360 } 3361 if (SPECIFIED(p->channels)) { 3362 pp.channels = p->channels; 3363 np++; 3364 } 3365 if (SPECIFIED(r->channels)) { 3366 rp.channels = r->channels; 3367 nr++; 3368 } 3369 3370 if (!audio_can_capture(sc)) 3371 nr = 0; 3372 if (!audio_can_playback(sc)) 3373 np = 0; 3374 3375 #ifdef AUDIO_DEBUG 3376 if (audiodebug && nr > 0) 3377 audio_print_params("audiosetinfo() Setting record params:", &rp); 3378 if (audiodebug && np > 0) 3379 audio_print_params("audiosetinfo() Setting play params:", &pp); 3380 #endif 3381 if (nr > 0 && (error = audio_check_params(&rp))) 3382 return error; 3383 if (np > 0 && (error = audio_check_params(&pp))) 3384 return error; 3385 3386 oldpblksize = sc->sc_pr.blksize; 3387 oldrblksize = sc->sc_rr.blksize; 3388 3389 setmode = 0; 3390 if (nr > 0) { 3391 if (!cleared) { 3392 audio_clear(sc); 3393 cleared = true; 3394 } 3395 modechange = true; 3396 setmode |= AUMODE_RECORD; 3397 } 3398 if (np > 0) { 3399 if (!cleared) { 3400 audio_clear(sc); 3401 cleared = true; 3402 } 3403 modechange = true; 3404 setmode |= AUMODE_PLAY; 3405 } 3406 3407 if (SPECIFIED(ai->mode)) { 3408 if (!cleared) { 3409 audio_clear(sc); 3410 cleared = true; 3411 } 3412 modechange = true; 3413 sc->sc_mode = ai->mode; 3414 if (sc->sc_mode & AUMODE_PLAY_ALL) 3415 sc->sc_mode |= AUMODE_PLAY; 3416 if ((sc->sc_mode & AUMODE_PLAY) && !sc->sc_full_duplex) 3417 /* Play takes precedence */ 3418 sc->sc_mode &= ~AUMODE_RECORD; 3419 } 3420 3421 oldpus = sc->sc_pustream; 3422 oldrus = sc->sc_rustream; 3423 if (modechange) { 3424 int indep; 3425 3426 indep = audio_get_props(sc) & AUDIO_PROP_INDEPENDENT; 3427 if (!indep) { 3428 if (setmode == AUMODE_RECORD) 3429 pp = rp; 3430 else if (setmode == AUMODE_PLAY) 3431 rp = pp; 3432 } 3433 memset(&pfilters, 0, sizeof(pfilters)); 3434 memset(&rfilters, 0, sizeof(rfilters)); 3435 pfilters.append = stream_filter_list_append; 3436 pfilters.prepend = stream_filter_list_prepend; 3437 pfilters.set = stream_filter_list_set; 3438 rfilters.append = stream_filter_list_append; 3439 rfilters.prepend = stream_filter_list_prepend; 3440 rfilters.set = stream_filter_list_set; 3441 /* Some device drivers change channels/sample_rate and change 3442 * no channels/sample_rate. */ 3443 error = hw->set_params(sc->hw_hdl, setmode, 3444 sc->sc_mode & (AUMODE_PLAY | AUMODE_RECORD), &pp, &rp, 3445 &pfilters, &rfilters); 3446 if (error) { 3447 DPRINTF(("%s: hw->set_params() failed with %d\n", 3448 __func__, error)); 3449 goto cleanup; 3450 } 3451 3452 audio_check_params(&pp); 3453 audio_check_params(&rp); 3454 if (!indep) { 3455 /* XXX for !indep device, we have to use the same 3456 * parameters for the hardware, not userland */ 3457 if (setmode == AUMODE_RECORD) { 3458 pp = rp; 3459 } else if (setmode == AUMODE_PLAY) { 3460 rp = pp; 3461 } 3462 } 3463 3464 if (sc->sc_pr.mmapped && pfilters.req_size > 0) { 3465 DPRINTF(("%s: mmapped, and filters are requested.\n", 3466 __func__)); 3467 error = EINVAL; 3468 goto cleanup; 3469 } 3470 3471 /* construct new filter chain */ 3472 if (setmode & AUMODE_PLAY) { 3473 error = audio_setup_pfilters(sc, &pp, &pfilters); 3474 if (error) 3475 goto cleanup; 3476 } 3477 if (setmode & AUMODE_RECORD) { 3478 error = audio_setup_rfilters(sc, &rp, &rfilters); 3479 if (error) 3480 goto cleanup; 3481 } 3482 DPRINTF(("%s: filter setup is completed.\n", __func__)); 3483 3484 /* userland formats */ 3485 sc->sc_pparams = pp; 3486 sc->sc_rparams = rp; 3487 } 3488 3489 /* Play params can affect the record params, so recalculate blksize. */ 3490 if (nr > 0 || np > 0) { 3491 audio_calc_blksize(sc, AUMODE_RECORD); 3492 audio_calc_blksize(sc, AUMODE_PLAY); 3493 } 3494 #ifdef AUDIO_DEBUG 3495 if (audiodebug > 1 && nr > 0) 3496 audio_print_params("audiosetinfo() After setting record params:", &sc->sc_rparams); 3497 if (audiodebug > 1 && np > 0) 3498 audio_print_params("audiosetinfo() After setting play params:", &sc->sc_pparams); 3499 #endif 3500 3501 if (SPECIFIED(p->port)) { 3502 if (!cleared) { 3503 audio_clear(sc); 3504 cleared = true; 3505 } 3506 error = au_set_port(sc, &sc->sc_outports, p->port); 3507 if (error) 3508 goto cleanup; 3509 } 3510 if (SPECIFIED(r->port)) { 3511 if (!cleared) { 3512 audio_clear(sc); 3513 cleared = true; 3514 } 3515 error = au_set_port(sc, &sc->sc_inports, r->port); 3516 if (error) 3517 goto cleanup; 3518 } 3519 if (SPECIFIED(p->gain)) { 3520 au_get_gain(sc, &sc->sc_outports, &gain, &balance); 3521 error = au_set_gain(sc, &sc->sc_outports, p->gain, balance); 3522 if (error) 3523 goto cleanup; 3524 } 3525 if (SPECIFIED(r->gain)) { 3526 au_get_gain(sc, &sc->sc_inports, &gain, &balance); 3527 error = au_set_gain(sc, &sc->sc_inports, r->gain, balance); 3528 if (error) 3529 goto cleanup; 3530 } 3531 3532 if (SPECIFIED_CH(p->balance)) { 3533 au_get_gain(sc, &sc->sc_outports, &gain, &balance); 3534 error = au_set_gain(sc, &sc->sc_outports, gain, p->balance); 3535 if (error) 3536 goto cleanup; 3537 } 3538 if (SPECIFIED_CH(r->balance)) { 3539 au_get_gain(sc, &sc->sc_inports, &gain, &balance); 3540 error = au_set_gain(sc, &sc->sc_inports, gain, r->balance); 3541 if (error) 3542 goto cleanup; 3543 } 3544 3545 if (SPECIFIED(ai->monitor_gain) && sc->sc_monitor_port != -1) { 3546 mixer_ctrl_t ct; 3547 3548 ct.dev = sc->sc_monitor_port; 3549 ct.type = AUDIO_MIXER_VALUE; 3550 ct.un.value.num_channels = 1; 3551 ct.un.value.level[AUDIO_MIXER_LEVEL_MONO] = ai->monitor_gain; 3552 error = sc->hw_if->set_port(sc->hw_hdl, &ct); 3553 if (error) 3554 goto cleanup; 3555 } 3556 3557 if (SPECIFIED_CH(p->pause)) { 3558 sc->sc_pr.pause = p->pause; 3559 pbus = !p->pause; 3560 pausechange = true; 3561 } 3562 if (SPECIFIED_CH(r->pause)) { 3563 sc->sc_rr.pause = r->pause; 3564 rbus = !r->pause; 3565 pausechange = true; 3566 } 3567 3568 if (SPECIFIED(ai->blocksize)) { 3569 int pblksize, rblksize; 3570 3571 /* Block size specified explicitly. */ 3572 if (ai->blocksize == 0) { 3573 if (!cleared) { 3574 audio_clear(sc); 3575 cleared = true; 3576 } 3577 sc->sc_blkset = false; 3578 audio_calc_blksize(sc, AUMODE_RECORD); 3579 audio_calc_blksize(sc, AUMODE_PLAY); 3580 } else { 3581 sc->sc_blkset = true; 3582 /* check whether new blocksize changes actually */ 3583 if (hw->round_blocksize == NULL) { 3584 if (!cleared) { 3585 audio_clear(sc); 3586 cleared = true; 3587 } 3588 sc->sc_pr.blksize = ai->blocksize; 3589 sc->sc_rr.blksize = ai->blocksize; 3590 } else { 3591 pblksize = hw->round_blocksize(sc->hw_hdl, 3592 ai->blocksize, AUMODE_PLAY, &sc->sc_pr.s.param); 3593 rblksize = hw->round_blocksize(sc->hw_hdl, 3594 ai->blocksize, AUMODE_RECORD, &sc->sc_rr.s.param); 3595 if (pblksize != sc->sc_pr.blksize || 3596 rblksize != sc->sc_rr.blksize) { 3597 if (!cleared) { 3598 audio_clear(sc); 3599 cleared = true; 3600 } 3601 sc->sc_pr.blksize = ai->blocksize; 3602 sc->sc_rr.blksize = ai->blocksize; 3603 } 3604 } 3605 } 3606 } 3607 3608 if (SPECIFIED(ai->mode)) { 3609 if (sc->sc_mode & AUMODE_PLAY) 3610 audio_init_play(sc); 3611 if (sc->sc_mode & AUMODE_RECORD) 3612 audio_init_record(sc); 3613 } 3614 3615 if (hw->commit_settings) { 3616 error = hw->commit_settings(sc->hw_hdl); 3617 if (error) 3618 goto cleanup; 3619 } 3620 3621 sc->sc_lastinfo = *ai; 3622 sc->sc_lastinfovalid = true; 3623 3624 cleanup: 3625 if (cleared || pausechange) { 3626 int init_error; 3627 3628 s = splaudio(); 3629 init_error = audio_initbufs(sc); 3630 if (init_error) goto err; 3631 if (sc->sc_pr.blksize != oldpblksize || 3632 sc->sc_rr.blksize != oldrblksize || 3633 sc->sc_pustream != oldpus || 3634 sc->sc_rustream != oldrus) 3635 audio_calcwater(sc); 3636 if ((sc->sc_mode & AUMODE_PLAY) && 3637 pbus && !sc->sc_pbus) 3638 init_error = audiostartp(sc); 3639 if (!init_error && 3640 (sc->sc_mode & AUMODE_RECORD) && 3641 rbus && !sc->sc_rbus) 3642 init_error = audiostartr(sc); 3643 err: 3644 splx(s); 3645 if (init_error) 3646 return init_error; 3647 } 3648 3649 /* Change water marks after initializing the buffers. */ 3650 if (SPECIFIED(ai->hiwat)) { 3651 blks = ai->hiwat; 3652 if (blks > sc->sc_pr.maxblks) 3653 blks = sc->sc_pr.maxblks; 3654 if (blks < 2) 3655 blks = 2; 3656 sc->sc_pr.usedhigh = blks * sc->sc_pr.blksize; 3657 } 3658 if (SPECIFIED(ai->lowat)) { 3659 blks = ai->lowat; 3660 if (blks > sc->sc_pr.maxblks - 1) 3661 blks = sc->sc_pr.maxblks - 1; 3662 sc->sc_pr.usedlow = blks * sc->sc_pr.blksize; 3663 } 3664 if (SPECIFIED(ai->hiwat) || SPECIFIED(ai->lowat)) { 3665 if (sc->sc_pr.usedlow > sc->sc_pr.usedhigh - sc->sc_pr.blksize) 3666 sc->sc_pr.usedlow = 3667 sc->sc_pr.usedhigh - sc->sc_pr.blksize; 3668 } 3669 3670 return error; 3671 } 3672 3673 int 3674 audiogetinfo(struct audio_softc *sc, struct audio_info *ai, int buf_only_mode) 3675 { 3676 struct audio_prinfo *r, *p; 3677 const struct audio_hw_if *hw; 3678 3679 r = &ai->record; 3680 p = &ai->play; 3681 hw = sc->hw_if; 3682 if (hw == NULL) /* HW has not attached */ 3683 return ENXIO; 3684 3685 p->sample_rate = sc->sc_pparams.sample_rate; 3686 r->sample_rate = sc->sc_rparams.sample_rate; 3687 p->channels = sc->sc_pparams.channels; 3688 r->channels = sc->sc_rparams.channels; 3689 p->precision = sc->sc_pparams.precision; 3690 r->precision = sc->sc_rparams.precision; 3691 p->encoding = sc->sc_pparams.encoding; 3692 r->encoding = sc->sc_rparams.encoding; 3693 3694 if (buf_only_mode) { 3695 r->port = 0; 3696 p->port = 0; 3697 3698 r->avail_ports = 0; 3699 p->avail_ports = 0; 3700 3701 r->gain = 0; 3702 r->balance = 0; 3703 3704 p->gain = 0; 3705 p->balance = 0; 3706 } else { 3707 r->port = au_get_port(sc, &sc->sc_inports); 3708 p->port = au_get_port(sc, &sc->sc_outports); 3709 3710 r->avail_ports = sc->sc_inports.allports; 3711 p->avail_ports = sc->sc_outports.allports; 3712 3713 au_get_gain(sc, &sc->sc_inports, &r->gain, &r->balance); 3714 au_get_gain(sc, &sc->sc_outports, &p->gain, &p->balance); 3715 } 3716 3717 if (sc->sc_monitor_port != -1 && buf_only_mode == 0) { 3718 mixer_ctrl_t ct; 3719 3720 ct.dev = sc->sc_monitor_port; 3721 ct.type = AUDIO_MIXER_VALUE; 3722 ct.un.value.num_channels = 1; 3723 if (sc->hw_if->get_port(sc->hw_hdl, &ct)) 3724 ai->monitor_gain = 0; 3725 else 3726 ai->monitor_gain = 3727 ct.un.value.level[AUDIO_MIXER_LEVEL_MONO]; 3728 } else 3729 ai->monitor_gain = 0; 3730 3731 p->seek = audio_stream_get_used(sc->sc_pustream); 3732 r->seek = audio_stream_get_used(sc->sc_rustream); 3733 3734 /* 3735 * XXX samples should be a value for userland data. 3736 * But drops is a value for HW data. 3737 */ 3738 p->samples = (sc->sc_pustream == &sc->sc_pr.s 3739 ? sc->sc_pr.stamp : sc->sc_pr.fstamp) - sc->sc_pr.drops; 3740 r->samples = (sc->sc_rustream == &sc->sc_rr.s 3741 ? sc->sc_rr.stamp : sc->sc_rr.fstamp) - sc->sc_rr.drops; 3742 3743 p->eof = sc->sc_eof; 3744 r->eof = 0; 3745 3746 p->pause = sc->sc_pr.pause; 3747 r->pause = sc->sc_rr.pause; 3748 3749 p->error = sc->sc_pr.drops != 0; 3750 r->error = sc->sc_rr.drops != 0; 3751 3752 p->waiting = r->waiting = 0; /* open never hangs */ 3753 3754 p->open = (sc->sc_open & AUOPEN_WRITE) != 0; 3755 r->open = (sc->sc_open & AUOPEN_READ) != 0; 3756 3757 p->active = sc->sc_pbus; 3758 r->active = sc->sc_rbus; 3759 3760 p->buffer_size = sc->sc_pustream ? sc->sc_pustream->bufsize : 0; 3761 r->buffer_size = sc->sc_rustream ? sc->sc_rustream->bufsize : 0; 3762 3763 ai->blocksize = sc->sc_pr.blksize; 3764 if (sc->sc_pr.blksize > 0) { 3765 ai->hiwat = sc->sc_pr.usedhigh / sc->sc_pr.blksize; 3766 ai->lowat = sc->sc_pr.usedlow / sc->sc_pr.blksize; 3767 } else 3768 ai->hiwat = ai->lowat = 0; 3769 ai->mode = sc->sc_mode; 3770 3771 return 0; 3772 } 3773 3774 /* 3775 * Mixer driver 3776 */ 3777 int 3778 mixer_open(dev_t dev, struct audio_softc *sc, int flags, 3779 int ifmt, struct lwp *l) 3780 { 3781 if (sc->hw_if == NULL) 3782 return ENXIO; 3783 3784 DPRINTF(("mixer_open: flags=0x%x sc=%p\n", flags, sc)); 3785 3786 return 0; 3787 } 3788 3789 /* 3790 * Remove a process from those to be signalled on mixer activity. 3791 */ 3792 static void 3793 mixer_remove(struct audio_softc *sc, struct lwp *l) 3794 { 3795 struct mixer_asyncs **pm, *m; 3796 struct proc *p; 3797 3798 if (l == NULL) 3799 return; 3800 3801 p = l->l_proc; 3802 3803 for (pm = &sc->sc_async_mixer; *pm; pm = &(*pm)->next) { 3804 if ((*pm)->proc == p) { 3805 m = *pm; 3806 *pm = m->next; 3807 free(m, M_DEVBUF); 3808 return; 3809 } 3810 } 3811 } 3812 3813 /* 3814 * Signal all processes waiting for the mixer. 3815 */ 3816 static void 3817 mixer_signal(struct audio_softc *sc) 3818 { 3819 struct mixer_asyncs *m; 3820 3821 for (m = sc->sc_async_mixer; m; m = m->next) { 3822 mutex_enter(proc_lock); 3823 psignal(m->proc, SIGIO); 3824 mutex_exit(proc_lock); 3825 } 3826 } 3827 3828 /* 3829 * Close a mixer device 3830 */ 3831 /* ARGSUSED */ 3832 int 3833 mixer_close(struct audio_softc *sc, int flags, int ifmt, 3834 struct lwp *l) 3835 { 3836 3837 DPRINTF(("mixer_close: sc %p\n", sc)); 3838 mixer_remove(sc, l); 3839 return 0; 3840 } 3841 3842 int 3843 mixer_ioctl(struct audio_softc *sc, u_long cmd, void *addr, int flag, 3844 struct lwp *l) 3845 { 3846 const struct audio_hw_if *hw; 3847 mixer_ctrl_t *mc; 3848 int error; 3849 3850 DPRINTF(("mixer_ioctl(%lu,'%c',%lu)\n", 3851 IOCPARM_LEN(cmd), (char)IOCGROUP(cmd), cmd&0xff)); 3852 hw = sc->hw_if; 3853 error = EINVAL; 3854 3855 /* we can return cached values if we are sleeping */ 3856 if (cmd != AUDIO_MIXER_READ) 3857 device_active(sc->dev, DVA_SYSTEM); 3858 3859 switch (cmd) { 3860 case FIOASYNC: 3861 mixer_remove(sc, l); /* remove old entry */ 3862 if (*(int *)addr) { 3863 struct mixer_asyncs *ma; 3864 ma = malloc(sizeof (struct mixer_asyncs), 3865 M_DEVBUF, M_WAITOK); 3866 ma->next = sc->sc_async_mixer; 3867 ma->proc = l->l_proc; 3868 sc->sc_async_mixer = ma; 3869 } 3870 error = 0; 3871 break; 3872 3873 case AUDIO_GETDEV: 3874 DPRINTF(("AUDIO_GETDEV\n")); 3875 error = hw->getdev(sc->hw_hdl, (audio_device_t *)addr); 3876 break; 3877 3878 case AUDIO_MIXER_DEVINFO: 3879 DPRINTF(("AUDIO_MIXER_DEVINFO\n")); 3880 ((mixer_devinfo_t *)addr)->un.v.delta = 0; /* default */ 3881 error = hw->query_devinfo(sc->hw_hdl, (mixer_devinfo_t *)addr); 3882 break; 3883 3884 case AUDIO_MIXER_READ: 3885 DPRINTF(("AUDIO_MIXER_READ\n")); 3886 mc = (mixer_ctrl_t *)addr; 3887 3888 if (device_is_active(sc->sc_dev) || 3889 sc->sc_mixer_state == NULL) 3890 error = hw->get_port(sc->hw_hdl, mc); 3891 else if (mc->dev >= sc->sc_nmixer_states) 3892 error = ENXIO; 3893 else { 3894 int dev = mc->dev; 3895 memcpy(mc, &sc->sc_mixer_state[dev], 3896 sizeof(mixer_ctrl_t)); 3897 error = 0; 3898 } 3899 break; 3900 3901 case AUDIO_MIXER_WRITE: 3902 DPRINTF(("AUDIO_MIXER_WRITE\n")); 3903 error = hw->set_port(sc->hw_hdl, (mixer_ctrl_t *)addr); 3904 if (!error && hw->commit_settings) 3905 error = hw->commit_settings(sc->hw_hdl); 3906 if (!error) 3907 mixer_signal(sc); 3908 break; 3909 3910 default: 3911 if (hw->dev_ioctl) 3912 error = hw->dev_ioctl(sc->hw_hdl, cmd, addr, flag, l); 3913 else 3914 error = EINVAL; 3915 break; 3916 } 3917 DPRINTF(("mixer_ioctl(%lu,'%c',%lu) result %d\n", 3918 IOCPARM_LEN(cmd), (char)IOCGROUP(cmd), cmd&0xff, error)); 3919 return error; 3920 } 3921 #endif /* NAUDIO > 0 */ 3922 3923 #include "midi.h" 3924 3925 #if NAUDIO == 0 && (NMIDI > 0 || NMIDIBUS > 0) 3926 #include <sys/param.h> 3927 #include <sys/systm.h> 3928 #include <sys/device.h> 3929 #include <sys/audioio.h> 3930 #include <dev/audio_if.h> 3931 #endif 3932 3933 #if NAUDIO > 0 || (NMIDI > 0 || NMIDIBUS > 0) 3934 int 3935 audioprint(void *aux, const char *pnp) 3936 { 3937 struct audio_attach_args *arg; 3938 const char *type; 3939 3940 if (pnp != NULL) { 3941 arg = aux; 3942 switch (arg->type) { 3943 case AUDIODEV_TYPE_AUDIO: 3944 type = "audio"; 3945 break; 3946 case AUDIODEV_TYPE_MIDI: 3947 type = "midi"; 3948 break; 3949 case AUDIODEV_TYPE_OPL: 3950 type = "opl"; 3951 break; 3952 case AUDIODEV_TYPE_MPU: 3953 type = "mpu"; 3954 break; 3955 default: 3956 panic("audioprint: unknown type %d", arg->type); 3957 } 3958 aprint_normal("%s at %s", type, pnp); 3959 } 3960 return UNCONF; 3961 } 3962 3963 #endif /* NAUDIO > 0 || (NMIDI > 0 || NMIDIBUS > 0) */ 3964 3965 #if NAUDIO > 0 3966 static void 3967 audio_mixer_capture(struct audio_softc *sc) 3968 { 3969 mixer_devinfo_t mi; 3970 mixer_ctrl_t *mc; 3971 3972 for (mi.index = 0; ; mi.index++) 3973 if (sc->hw_if->query_devinfo(sc->hw_hdl, &mi) != 0) 3974 break; 3975 3976 #ifdef DIAGNOSTIC 3977 if (sc->sc_mixer_state != NULL && sc->sc_nmixer_states != mi.index) { 3978 free(sc->sc_mixer_state, M_DEVBUF); 3979 sc->sc_mixer_state = NULL; 3980 } 3981 #endif 3982 3983 sc->sc_nmixer_states = mi.index; 3984 if (sc->sc_mixer_state == NULL) 3985 sc->sc_mixer_state = malloc( 3986 sizeof(mixer_ctrl_t) * sc->sc_nmixer_states, 3987 M_DEVBUF, M_NOWAIT); 3988 if (sc->sc_mixer_state == NULL) { 3989 aprint_error("%s: couldn't allocate memory for mixer state\n", 3990 device_xname(sc->dev)); 3991 return; 3992 } 3993 3994 for (mi.index = 0; ; mi.index++) { 3995 if (sc->hw_if->query_devinfo(sc->hw_hdl, &mi) != 0) 3996 break; 3997 if (mi.type == AUDIO_MIXER_CLASS) 3998 continue; 3999 mc = &sc->sc_mixer_state[mi.index]; 4000 mc->dev = mi.index; 4001 mc->type = mi.type; 4002 mc->un.value.num_channels = mi.un.v.num_channels; 4003 (void)sc->hw_if->get_port(sc->hw_hdl, mc); 4004 } 4005 4006 return; 4007 } 4008 4009 static void 4010 audio_mixer_restore(struct audio_softc *sc) 4011 { 4012 mixer_devinfo_t mi; 4013 mixer_ctrl_t *mc; 4014 4015 if (sc->sc_mixer_state == NULL) 4016 return; 4017 4018 for (mi.index = 0; ; mi.index++) { 4019 if (sc->hw_if->query_devinfo(sc->hw_hdl, &mi) != 0) 4020 break; 4021 if (mi.type == AUDIO_MIXER_CLASS) 4022 continue; 4023 mc = &sc->sc_mixer_state[mi.index]; 4024 (void)sc->hw_if->set_port(sc->hw_hdl, mc); 4025 } 4026 if (sc->hw_if->commit_settings) 4027 sc->hw_if->commit_settings(sc->hw_hdl); 4028 4029 return; 4030 } 4031 4032 #ifdef AUDIO_PM_IDLE 4033 static void 4034 audio_idle(void *arg) 4035 { 4036 device_t dv = arg; 4037 struct audio_softc *sc = device_private(dv); 4038 4039 #ifdef PNP_DEBUG 4040 extern int pnp_debug_idle; 4041 if (pnp_debug_idle) 4042 printf("%s: idle handler called\n", device_xname(dv)); 4043 #endif 4044 4045 sc->sc_idle = true; 4046 4047 /* XXX joerg Make pmf_device_suspend handle children? */ 4048 if (!pmf_device_suspend(dv, PMF_F_SELF)) 4049 return; 4050 4051 if (!pmf_device_suspend(sc->sc_dev, PMF_F_SELF)) 4052 pmf_device_resume(dv, PMF_F_SELF); 4053 } 4054 4055 static void 4056 audio_activity(device_t dv, devactive_t type) 4057 { 4058 struct audio_softc *sc = device_private(dv); 4059 4060 if (type != DVA_SYSTEM) 4061 return; 4062 4063 callout_schedule(&sc->sc_idle_counter, audio_idle_timeout * hz); 4064 4065 sc->sc_idle = false; 4066 if (!device_is_active(dv)) { 4067 /* XXX joerg How to deal with a failing resume... */ 4068 pmf_device_resume(sc->sc_dev, PMF_F_SELF); 4069 pmf_device_resume(dv, PMF_F_SELF); 4070 } 4071 } 4072 #endif 4073 4074 static bool 4075 audio_suspend(device_t dv PMF_FN_ARGS) 4076 { 4077 struct audio_softc *sc = device_private(dv); 4078 const struct audio_hw_if *hwp = sc->hw_if; 4079 int (s); 4080 4081 s = splaudio(); 4082 audio_mixer_capture(sc); 4083 if (sc->sc_pbus == true) 4084 hwp->halt_output(sc->hw_hdl); 4085 if (sc->sc_rbus == true) 4086 hwp->halt_input(sc->hw_hdl); 4087 #ifdef AUDIO_PM_IDLE 4088 callout_stop(&sc->sc_idle_counter); 4089 #endif 4090 splx(s); 4091 4092 return true; 4093 } 4094 4095 static bool 4096 audio_resume(device_t dv PMF_FN_ARGS) 4097 { 4098 struct audio_softc *sc = device_private(dv); 4099 int s; 4100 4101 s = splaudio(); 4102 if (sc->sc_lastinfovalid) 4103 audiosetinfo(sc, &sc->sc_lastinfo); 4104 audio_mixer_restore(sc); 4105 if ((sc->sc_pbus == true) && !sc->sc_pr.pause) 4106 audiostartp(sc); 4107 if ((sc->sc_rbus == true) && !sc->sc_rr.pause) 4108 audiostartr(sc); 4109 splx(s); 4110 4111 return true; 4112 } 4113 4114 static void 4115 audio_volume_down(device_t dv) 4116 { 4117 struct audio_softc *sc = device_private(dv); 4118 mixer_devinfo_t mi; 4119 int newgain; 4120 u_int gain; 4121 u_char balance; 4122 int s; 4123 4124 if (sc->sc_outports.index == -1 && sc->sc_outports.master != -1) { 4125 mi.index = sc->sc_outports.master; 4126 mi.un.v.delta = 0; 4127 if (sc->hw_if->query_devinfo(sc->hw_hdl, &mi) != 0) 4128 return; 4129 4130 s = splaudio(); 4131 au_get_gain(sc, &sc->sc_outports, &gain, &balance); 4132 newgain = gain - mi.un.v.delta; 4133 if (newgain < AUDIO_MIN_GAIN) 4134 newgain = AUDIO_MIN_GAIN; 4135 au_set_gain(sc, &sc->sc_outports, newgain, balance); 4136 splx(s); 4137 } 4138 } 4139 4140 static void 4141 audio_volume_up(device_t dv) 4142 { 4143 struct audio_softc *sc = device_private(dv); 4144 mixer_devinfo_t mi; 4145 u_int gain, newgain; 4146 u_char balance; 4147 int s; 4148 4149 if (sc->sc_outports.index == -1 && sc->sc_outports.master != -1) { 4150 mi.index = sc->sc_outports.master; 4151 mi.un.v.delta = 0; 4152 if (sc->hw_if->query_devinfo(sc->hw_hdl, &mi) != 0) 4153 return; 4154 4155 s = splaudio(); 4156 au_get_gain(sc, &sc->sc_outports, &gain, &balance); 4157 newgain = gain + mi.un.v.delta; 4158 if (newgain > AUDIO_MAX_GAIN) 4159 newgain = AUDIO_MAX_GAIN; 4160 au_set_gain(sc, &sc->sc_outports, newgain, balance); 4161 splx(s); 4162 } 4163 } 4164 4165 static void 4166 audio_volume_toggle(device_t dv) 4167 { 4168 struct audio_softc *sc = device_private(dv); 4169 u_int gain, newgain; 4170 u_char balance; 4171 int s; 4172 4173 s = splaudio(); 4174 au_get_gain(sc, &sc->sc_outports, &gain, &balance); 4175 if (gain != 0) { 4176 sc->sc_lastgain = gain; 4177 newgain = 0; 4178 } else 4179 newgain = sc->sc_lastgain; 4180 au_set_gain(sc, &sc->sc_outports, newgain, balance); 4181 splx(s); 4182 } 4183 4184 static int 4185 audio_get_props(struct audio_softc *sc) 4186 { 4187 const struct audio_hw_if *hw; 4188 int props; 4189 4190 hw = sc->hw_if; 4191 props = hw->get_props(sc->hw_hdl); 4192 4193 /* 4194 * if neither playback nor capture properties are reported, 4195 * assume both are supported by the device driver 4196 */ 4197 if ((props & (AUDIO_PROP_PLAYBACK|AUDIO_PROP_CAPTURE)) == 0) 4198 props |= (AUDIO_PROP_PLAYBACK | AUDIO_PROP_CAPTURE); 4199 4200 return props; 4201 } 4202 4203 static bool 4204 audio_can_playback(struct audio_softc *sc) 4205 { 4206 return audio_get_props(sc) & AUDIO_PROP_PLAYBACK ? true : false; 4207 } 4208 4209 static bool 4210 audio_can_capture(struct audio_softc *sc) 4211 { 4212 return audio_get_props(sc) & AUDIO_PROP_CAPTURE ? true : false; 4213 } 4214 4215 #endif /* NAUDIO > 0 */
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