audio.c revision 1.58
1 /* $NetBSD: audio.c,v 1.58 2020/02/29 09:38:10 isaki Exp $ */ 2 3 /*- 4 * Copyright (c) 2008 The NetBSD Foundation, Inc. 5 * All rights reserved. 6 * 7 * This code is derived from software contributed to The NetBSD Foundation 8 * by Andrew Doran. 9 * 10 * Redistribution and use in source and binary forms, with or without 11 * modification, are permitted provided that the following conditions 12 * are met: 13 * 1. Redistributions of source code must retain the above copyright 14 * notice, this list of conditions and the following disclaimer. 15 * 2. Redistributions in binary form must reproduce the above copyright 16 * notice, this list of conditions and the following disclaimer in the 17 * documentation and/or other materials provided with the distribution. 18 * 19 * THIS SOFTWARE IS PROVIDED BY THE NETBSD FOUNDATION, INC. AND CONTRIBUTORS 20 * ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED 21 * TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR 22 * PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE FOUNDATION OR CONTRIBUTORS 23 * BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR 24 * CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF 25 * SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS 26 * INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN 27 * CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) 28 * ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE 29 * POSSIBILITY OF SUCH DAMAGE. 30 */ 31 32 /* 33 * Copyright (c) 1991-1993 Regents of the University of California. 34 * All rights reserved. 35 * 36 * Redistribution and use in source and binary forms, with or without 37 * modification, are permitted provided that the following conditions 38 * are met: 39 * 1. Redistributions of source code must retain the above copyright 40 * notice, this list of conditions and the following disclaimer. 41 * 2. Redistributions in binary form must reproduce the above copyright 42 * notice, this list of conditions and the following disclaimer in the 43 * documentation and/or other materials provided with the distribution. 44 * 3. All advertising materials mentioning features or use of this software 45 * must display the following acknowledgement: 46 * This product includes software developed by the Computer Systems 47 * Engineering Group at Lawrence Berkeley Laboratory. 48 * 4. Neither the name of the University nor of the Laboratory may be used 49 * to endorse or promote products derived from this software without 50 * specific prior written permission. 51 * 52 * THIS SOFTWARE IS PROVIDED BY THE REGENTS AND CONTRIBUTORS ``AS IS'' AND 53 * ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE 54 * IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE 55 * ARE DISCLAIMED. IN NO EVENT SHALL THE REGENTS OR CONTRIBUTORS BE LIABLE 56 * FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL 57 * DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS 58 * OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) 59 * HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT 60 * LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY 61 * OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF 62 * SUCH DAMAGE. 63 */ 64 65 /* 66 * Locking: there are three locks per device. 67 * 68 * - sc_lock, provided by the underlying driver. This is an adaptive lock, 69 * returned in the second parameter to hw_if->get_locks(). It is known 70 * as the "thread lock". 71 * 72 * It serializes access to state in all places except the 73 * driver's interrupt service routine. This lock is taken from process 74 * context (example: access to /dev/audio). It is also taken from soft 75 * interrupt handlers in this module, primarily to serialize delivery of 76 * wakeups. This lock may be used/provided by modules external to the 77 * audio subsystem, so take care not to introduce a lock order problem. 78 * LONG TERM SLEEPS MUST NOT OCCUR WITH THIS LOCK HELD. 79 * 80 * - sc_intr_lock, provided by the underlying driver. This may be either a 81 * spinlock (at IPL_SCHED or IPL_VM) or an adaptive lock (IPL_NONE or 82 * IPL_SOFT*), returned in the first parameter to hw_if->get_locks(). It 83 * is known as the "interrupt lock". 84 * 85 * It provides atomic access to the device's hardware state, and to audio 86 * channel data that may be accessed by the hardware driver's ISR. 87 * In all places outside the ISR, sc_lock must be held before taking 88 * sc_intr_lock. This is to ensure that groups of hardware operations are 89 * made atomically. SLEEPS CANNOT OCCUR WITH THIS LOCK HELD. 90 * 91 * - sc_exlock, private to this module. This is a variable protected by 92 * sc_lock. It is known as the "critical section". 93 * Some operations release sc_lock in order to allocate memory, to wait 94 * for in-flight I/O to complete, to copy to/from user context, etc. 95 * sc_exlock provides a critical section even under the circumstance. 96 * "+" in following list indicates the interfaces which necessary to be 97 * protected by sc_exlock. 98 * 99 * List of hardware interface methods, and which locks are held when each 100 * is called by this module: 101 * 102 * METHOD INTR THREAD NOTES 103 * ----------------------- ------- ------- ------------------------- 104 * open x x + 105 * close x x + 106 * query_format - x 107 * set_format - x 108 * round_blocksize - x 109 * commit_settings - x 110 * init_output x x 111 * init_input x x 112 * start_output x x + 113 * start_input x x + 114 * halt_output x x + 115 * halt_input x x + 116 * speaker_ctl x x 117 * getdev - x 118 * set_port - x + 119 * get_port - x + 120 * query_devinfo - x 121 * allocm - - + (*1) 122 * freem - - + (*1) 123 * round_buffersize - x 124 * get_props - - Called at attach time 125 * trigger_output x x + 126 * trigger_input x x + 127 * dev_ioctl - x 128 * get_locks - - Called at attach time 129 * 130 * *1 Note: Before 8.0, since these have been called only at attach time, 131 * neither lock were necessary. Currently, on the other hand, since 132 * these may be also called after attach, the thread lock is required. 133 * 134 * In addition, there is an additional lock. 135 * 136 * - track->lock. This is an atomic variable and is similar to the 137 * "interrupt lock". This is one for each track. If any thread context 138 * (and software interrupt context) and hardware interrupt context who 139 * want to access some variables on this track, they must acquire this 140 * lock before. It protects track's consistency between hardware 141 * interrupt context and others. 142 */ 143 144 #include <sys/cdefs.h> 145 __KERNEL_RCSID(0, "$NetBSD: audio.c,v 1.58 2020/02/29 09:38:10 isaki Exp $"); 146 147 #ifdef _KERNEL_OPT 148 #include "audio.h" 149 #include "midi.h" 150 #endif 151 152 #if NAUDIO > 0 153 154 #include <sys/types.h> 155 #include <sys/param.h> 156 #include <sys/atomic.h> 157 #include <sys/audioio.h> 158 #include <sys/conf.h> 159 #include <sys/cpu.h> 160 #include <sys/device.h> 161 #include <sys/fcntl.h> 162 #include <sys/file.h> 163 #include <sys/filedesc.h> 164 #include <sys/intr.h> 165 #include <sys/ioctl.h> 166 #include <sys/kauth.h> 167 #include <sys/kernel.h> 168 #include <sys/kmem.h> 169 #include <sys/malloc.h> 170 #include <sys/mman.h> 171 #include <sys/module.h> 172 #include <sys/poll.h> 173 #include <sys/proc.h> 174 #include <sys/queue.h> 175 #include <sys/select.h> 176 #include <sys/signalvar.h> 177 #include <sys/stat.h> 178 #include <sys/sysctl.h> 179 #include <sys/systm.h> 180 #include <sys/syslog.h> 181 #include <sys/vnode.h> 182 183 #include <dev/audio/audio_if.h> 184 #include <dev/audio/audiovar.h> 185 #include <dev/audio/audiodef.h> 186 #include <dev/audio/linear.h> 187 #include <dev/audio/mulaw.h> 188 189 #include <machine/endian.h> 190 191 #include <uvm/uvm_extern.h> 192 193 #include "ioconf.h" 194 195 /* 196 * 0: No debug logs 197 * 1: action changes like open/close/set_format... 198 * 2: + normal operations like read/write/ioctl... 199 * 3: + TRACEs except interrupt 200 * 4: + TRACEs including interrupt 201 */ 202 //#define AUDIO_DEBUG 1 203 204 #if defined(AUDIO_DEBUG) 205 206 int audiodebug = AUDIO_DEBUG; 207 static void audio_vtrace(struct audio_softc *sc, const char *, const char *, 208 const char *, va_list); 209 static void audio_trace(struct audio_softc *sc, const char *, const char *, ...) 210 __printflike(3, 4); 211 static void audio_tracet(const char *, audio_track_t *, const char *, ...) 212 __printflike(3, 4); 213 static void audio_tracef(const char *, audio_file_t *, const char *, ...) 214 __printflike(3, 4); 215 216 /* XXX sloppy memory logger */ 217 static void audio_mlog_init(void); 218 static void audio_mlog_free(void); 219 static void audio_mlog_softintr(void *); 220 extern void audio_mlog_flush(void); 221 extern void audio_mlog_printf(const char *, ...); 222 223 static int mlog_refs; /* reference counter */ 224 static char *mlog_buf[2]; /* double buffer */ 225 static int mlog_buflen; /* buffer length */ 226 static int mlog_used; /* used length */ 227 static int mlog_full; /* number of dropped lines by buffer full */ 228 static int mlog_drop; /* number of dropped lines by busy */ 229 static volatile uint32_t mlog_inuse; /* in-use */ 230 static int mlog_wpage; /* active page */ 231 static void *mlog_sih; /* softint handle */ 232 233 static void 234 audio_mlog_init(void) 235 { 236 mlog_refs++; 237 if (mlog_refs > 1) 238 return; 239 mlog_buflen = 4096; 240 mlog_buf[0] = kmem_zalloc(mlog_buflen, KM_SLEEP); 241 mlog_buf[1] = kmem_zalloc(mlog_buflen, KM_SLEEP); 242 mlog_used = 0; 243 mlog_full = 0; 244 mlog_drop = 0; 245 mlog_inuse = 0; 246 mlog_wpage = 0; 247 mlog_sih = softint_establish(SOFTINT_SERIAL, audio_mlog_softintr, NULL); 248 if (mlog_sih == NULL) 249 printf("%s: softint_establish failed\n", __func__); 250 } 251 252 static void 253 audio_mlog_free(void) 254 { 255 mlog_refs--; 256 if (mlog_refs > 0) 257 return; 258 259 audio_mlog_flush(); 260 if (mlog_sih) 261 softint_disestablish(mlog_sih); 262 kmem_free(mlog_buf[0], mlog_buflen); 263 kmem_free(mlog_buf[1], mlog_buflen); 264 } 265 266 /* 267 * Flush memory buffer. 268 * It must not be called from hardware interrupt context. 269 */ 270 void 271 audio_mlog_flush(void) 272 { 273 if (mlog_refs == 0) 274 return; 275 276 /* Nothing to do if already in use ? */ 277 if (atomic_swap_32(&mlog_inuse, 1) == 1) 278 return; 279 280 int rpage = mlog_wpage; 281 mlog_wpage ^= 1; 282 mlog_buf[mlog_wpage][0] = '\0'; 283 mlog_used = 0; 284 285 atomic_swap_32(&mlog_inuse, 0); 286 287 if (mlog_buf[rpage][0] != '\0') { 288 printf("%s", mlog_buf[rpage]); 289 if (mlog_drop > 0) 290 printf("mlog_drop %d\n", mlog_drop); 291 if (mlog_full > 0) 292 printf("mlog_full %d\n", mlog_full); 293 } 294 mlog_full = 0; 295 mlog_drop = 0; 296 } 297 298 static void 299 audio_mlog_softintr(void *cookie) 300 { 301 audio_mlog_flush(); 302 } 303 304 void 305 audio_mlog_printf(const char *fmt, ...) 306 { 307 int len; 308 va_list ap; 309 310 if (atomic_swap_32(&mlog_inuse, 1) == 1) { 311 /* already inuse */ 312 mlog_drop++; 313 return; 314 } 315 316 va_start(ap, fmt); 317 len = vsnprintf( 318 mlog_buf[mlog_wpage] + mlog_used, 319 mlog_buflen - mlog_used, 320 fmt, ap); 321 va_end(ap); 322 323 mlog_used += len; 324 if (mlog_buflen - mlog_used <= 1) { 325 mlog_full++; 326 } 327 328 atomic_swap_32(&mlog_inuse, 0); 329 330 if (mlog_sih) 331 softint_schedule(mlog_sih); 332 } 333 334 /* trace functions */ 335 static void 336 audio_vtrace(struct audio_softc *sc, const char *funcname, const char *header, 337 const char *fmt, va_list ap) 338 { 339 char buf[256]; 340 int n; 341 342 n = 0; 343 buf[0] = '\0'; 344 n += snprintf(buf + n, sizeof(buf) - n, "%s@%d %s", 345 funcname, device_unit(sc->sc_dev), header); 346 n += vsnprintf(buf + n, sizeof(buf) - n, fmt, ap); 347 348 if (cpu_intr_p()) { 349 audio_mlog_printf("%s\n", buf); 350 } else { 351 audio_mlog_flush(); 352 printf("%s\n", buf); 353 } 354 } 355 356 static void 357 audio_trace(struct audio_softc *sc, const char *funcname, const char *fmt, ...) 358 { 359 va_list ap; 360 361 va_start(ap, fmt); 362 audio_vtrace(sc, funcname, "", fmt, ap); 363 va_end(ap); 364 } 365 366 static void 367 audio_tracet(const char *funcname, audio_track_t *track, const char *fmt, ...) 368 { 369 char hdr[16]; 370 va_list ap; 371 372 snprintf(hdr, sizeof(hdr), "#%d ", track->id); 373 va_start(ap, fmt); 374 audio_vtrace(track->mixer->sc, funcname, hdr, fmt, ap); 375 va_end(ap); 376 } 377 378 static void 379 audio_tracef(const char *funcname, audio_file_t *file, const char *fmt, ...) 380 { 381 char hdr[32]; 382 char phdr[16], rhdr[16]; 383 va_list ap; 384 385 phdr[0] = '\0'; 386 rhdr[0] = '\0'; 387 if (file->ptrack) 388 snprintf(phdr, sizeof(phdr), "#%d", file->ptrack->id); 389 if (file->rtrack) 390 snprintf(rhdr, sizeof(rhdr), "#%d", file->rtrack->id); 391 snprintf(hdr, sizeof(hdr), "{%s,%s} ", phdr, rhdr); 392 393 va_start(ap, fmt); 394 audio_vtrace(file->sc, funcname, hdr, fmt, ap); 395 va_end(ap); 396 } 397 398 #define DPRINTF(n, fmt...) do { \ 399 if (audiodebug >= (n)) { \ 400 audio_mlog_flush(); \ 401 printf(fmt); \ 402 } \ 403 } while (0) 404 #define TRACE(n, fmt...) do { \ 405 if (audiodebug >= (n)) audio_trace(sc, __func__, fmt); \ 406 } while (0) 407 #define TRACET(n, t, fmt...) do { \ 408 if (audiodebug >= (n)) audio_tracet(__func__, t, fmt); \ 409 } while (0) 410 #define TRACEF(n, f, fmt...) do { \ 411 if (audiodebug >= (n)) audio_tracef(__func__, f, fmt); \ 412 } while (0) 413 414 struct audio_track_debugbuf { 415 char usrbuf[32]; 416 char codec[32]; 417 char chvol[32]; 418 char chmix[32]; 419 char freq[32]; 420 char outbuf[32]; 421 }; 422 423 static void 424 audio_track_bufstat(audio_track_t *track, struct audio_track_debugbuf *buf) 425 { 426 427 memset(buf, 0, sizeof(*buf)); 428 429 snprintf(buf->outbuf, sizeof(buf->outbuf), " out=%d/%d/%d", 430 track->outbuf.head, track->outbuf.used, track->outbuf.capacity); 431 if (track->freq.filter) 432 snprintf(buf->freq, sizeof(buf->freq), " f=%d/%d/%d", 433 track->freq.srcbuf.head, 434 track->freq.srcbuf.used, 435 track->freq.srcbuf.capacity); 436 if (track->chmix.filter) 437 snprintf(buf->chmix, sizeof(buf->chmix), " m=%d", 438 track->chmix.srcbuf.used); 439 if (track->chvol.filter) 440 snprintf(buf->chvol, sizeof(buf->chvol), " v=%d", 441 track->chvol.srcbuf.used); 442 if (track->codec.filter) 443 snprintf(buf->codec, sizeof(buf->codec), " e=%d", 444 track->codec.srcbuf.used); 445 snprintf(buf->usrbuf, sizeof(buf->usrbuf), " usr=%d/%d/H%d", 446 track->usrbuf.head, track->usrbuf.used, track->usrbuf_usedhigh); 447 } 448 #else 449 #define DPRINTF(n, fmt...) do { } while (0) 450 #define TRACE(n, fmt, ...) do { } while (0) 451 #define TRACET(n, t, fmt, ...) do { } while (0) 452 #define TRACEF(n, f, fmt, ...) do { } while (0) 453 #endif 454 455 #define SPECIFIED(x) ((x) != ~0) 456 #define SPECIFIED_CH(x) ((x) != (u_char)~0) 457 458 /* Device timeout in msec */ 459 #define AUDIO_TIMEOUT (3000) 460 461 /* #define AUDIO_PM_IDLE */ 462 #ifdef AUDIO_PM_IDLE 463 int audio_idle_timeout = 30; 464 #endif 465 466 /* Number of elements of async mixer's pid */ 467 #define AM_CAPACITY (4) 468 469 struct portname { 470 const char *name; 471 int mask; 472 }; 473 474 static int audiomatch(device_t, cfdata_t, void *); 475 static void audioattach(device_t, device_t, void *); 476 static int audiodetach(device_t, int); 477 static int audioactivate(device_t, enum devact); 478 static void audiochilddet(device_t, device_t); 479 static int audiorescan(device_t, const char *, const int *); 480 481 static int audio_modcmd(modcmd_t, void *); 482 483 #ifdef AUDIO_PM_IDLE 484 static void audio_idle(void *); 485 static void audio_activity(device_t, devactive_t); 486 #endif 487 488 static bool audio_suspend(device_t dv, const pmf_qual_t *); 489 static bool audio_resume(device_t dv, const pmf_qual_t *); 490 static void audio_volume_down(device_t); 491 static void audio_volume_up(device_t); 492 static void audio_volume_toggle(device_t); 493 494 static void audio_mixer_capture(struct audio_softc *); 495 static void audio_mixer_restore(struct audio_softc *); 496 497 static void audio_softintr_rd(void *); 498 static void audio_softintr_wr(void *); 499 500 static int audio_enter_exclusive(struct audio_softc *); 501 static void audio_exit_exclusive(struct audio_softc *); 502 static struct audio_softc *audio_file_enter(audio_file_t *, struct psref *); 503 static void audio_file_exit(struct audio_softc *, struct psref *); 504 static int audio_track_waitio(struct audio_softc *, audio_track_t *); 505 506 static int audioclose(struct file *); 507 static int audioread(struct file *, off_t *, struct uio *, kauth_cred_t, int); 508 static int audiowrite(struct file *, off_t *, struct uio *, kauth_cred_t, int); 509 static int audioioctl(struct file *, u_long, void *); 510 static int audiopoll(struct file *, int); 511 static int audiokqfilter(struct file *, struct knote *); 512 static int audiommap(struct file *, off_t *, size_t, int, int *, int *, 513 struct uvm_object **, int *); 514 static int audiostat(struct file *, struct stat *); 515 516 static void filt_audiowrite_detach(struct knote *); 517 static int filt_audiowrite_event(struct knote *, long); 518 static void filt_audioread_detach(struct knote *); 519 static int filt_audioread_event(struct knote *, long); 520 521 static int audio_open(dev_t, struct audio_softc *, int, int, struct lwp *, 522 audio_file_t **); 523 static int audio_close(struct audio_softc *, audio_file_t *); 524 static int audio_unlink(struct audio_softc *, audio_file_t *); 525 static int audio_read(struct audio_softc *, struct uio *, int, audio_file_t *); 526 static int audio_write(struct audio_softc *, struct uio *, int, audio_file_t *); 527 static void audio_file_clear(struct audio_softc *, audio_file_t *); 528 static int audio_ioctl(dev_t, struct audio_softc *, u_long, void *, int, 529 struct lwp *, audio_file_t *); 530 static int audio_poll(struct audio_softc *, int, struct lwp *, audio_file_t *); 531 static int audio_kqfilter(struct audio_softc *, audio_file_t *, struct knote *); 532 static int audio_mmap(struct audio_softc *, off_t *, size_t, int, int *, int *, 533 struct uvm_object **, int *, audio_file_t *); 534 535 static int audioctl_open(dev_t, struct audio_softc *, int, int, struct lwp *); 536 537 static void audio_pintr(void *); 538 static void audio_rintr(void *); 539 540 static int audio_query_devinfo(struct audio_softc *, mixer_devinfo_t *); 541 542 static __inline int audio_track_readablebytes(const audio_track_t *); 543 static int audio_file_setinfo(struct audio_softc *, audio_file_t *, 544 const struct audio_info *); 545 static int audio_track_setinfo_check(audio_format2_t *, 546 const struct audio_prinfo *, const audio_format2_t *); 547 static void audio_track_setinfo_water(audio_track_t *, 548 const struct audio_info *); 549 static int audio_hw_setinfo(struct audio_softc *, const struct audio_info *, 550 struct audio_info *); 551 static int audio_hw_set_format(struct audio_softc *, int, 552 const audio_format2_t *, const audio_format2_t *, 553 audio_filter_reg_t *, audio_filter_reg_t *); 554 static int audiogetinfo(struct audio_softc *, struct audio_info *, int, 555 audio_file_t *); 556 static bool audio_can_playback(struct audio_softc *); 557 static bool audio_can_capture(struct audio_softc *); 558 static int audio_check_params(audio_format2_t *); 559 static int audio_mixers_init(struct audio_softc *sc, int, 560 const audio_format2_t *, const audio_format2_t *, 561 const audio_filter_reg_t *, const audio_filter_reg_t *); 562 static int audio_select_freq(const struct audio_format *); 563 static int audio_hw_probe(struct audio_softc *, audio_format2_t *, int); 564 static int audio_hw_validate_format(struct audio_softc *, int, 565 const audio_format2_t *); 566 static int audio_mixers_set_format(struct audio_softc *, 567 const struct audio_info *); 568 static void audio_mixers_get_format(struct audio_softc *, struct audio_info *); 569 static int audio_sysctl_blk_ms(SYSCTLFN_PROTO); 570 static int audio_sysctl_multiuser(SYSCTLFN_PROTO); 571 #if defined(AUDIO_DEBUG) 572 static int audio_sysctl_debug(SYSCTLFN_PROTO); 573 static void audio_format2_tostr(char *, size_t, const audio_format2_t *); 574 static void audio_print_format2(const char *, const audio_format2_t *) __unused; 575 #endif 576 577 static void *audio_realloc(void *, size_t); 578 static int audio_realloc_usrbuf(audio_track_t *, int); 579 static void audio_free_usrbuf(audio_track_t *); 580 581 static audio_track_t *audio_track_create(struct audio_softc *, 582 audio_trackmixer_t *); 583 static void audio_track_destroy(audio_track_t *); 584 static audio_filter_t audio_track_get_codec(audio_track_t *, 585 const audio_format2_t *, const audio_format2_t *); 586 static int audio_track_set_format(audio_track_t *, audio_format2_t *); 587 static void audio_track_play(audio_track_t *); 588 static int audio_track_drain(struct audio_softc *, audio_track_t *); 589 static void audio_track_record(audio_track_t *); 590 static void audio_track_clear(struct audio_softc *, audio_track_t *); 591 592 static int audio_mixer_init(struct audio_softc *, int, 593 const audio_format2_t *, const audio_filter_reg_t *); 594 static void audio_mixer_destroy(struct audio_softc *, audio_trackmixer_t *); 595 static void audio_pmixer_start(struct audio_softc *, bool); 596 static void audio_pmixer_process(struct audio_softc *); 597 static void audio_pmixer_agc(audio_trackmixer_t *, int); 598 static int audio_pmixer_mix_track(audio_trackmixer_t *, audio_track_t *, int); 599 static void audio_pmixer_output(struct audio_softc *); 600 static int audio_pmixer_halt(struct audio_softc *); 601 static void audio_rmixer_start(struct audio_softc *); 602 static void audio_rmixer_process(struct audio_softc *); 603 static void audio_rmixer_input(struct audio_softc *); 604 static int audio_rmixer_halt(struct audio_softc *); 605 606 static void mixer_init(struct audio_softc *); 607 static int mixer_open(dev_t, struct audio_softc *, int, int, struct lwp *); 608 static int mixer_close(struct audio_softc *, audio_file_t *); 609 static int mixer_ioctl(struct audio_softc *, u_long, void *, int, struct lwp *); 610 static void mixer_async_add(struct audio_softc *, pid_t); 611 static void mixer_async_remove(struct audio_softc *, pid_t); 612 static void mixer_signal(struct audio_softc *); 613 614 static int au_portof(struct audio_softc *, char *, int); 615 616 static void au_setup_ports(struct audio_softc *, struct au_mixer_ports *, 617 mixer_devinfo_t *, const struct portname *); 618 static int au_set_lr_value(struct audio_softc *, mixer_ctrl_t *, int, int); 619 static int au_get_lr_value(struct audio_softc *, mixer_ctrl_t *, int *, int *); 620 static int au_set_gain(struct audio_softc *, struct au_mixer_ports *, int, int); 621 static void au_get_gain(struct audio_softc *, struct au_mixer_ports *, 622 u_int *, u_char *); 623 static int au_set_port(struct audio_softc *, struct au_mixer_ports *, u_int); 624 static int au_get_port(struct audio_softc *, struct au_mixer_ports *); 625 static int au_set_monitor_gain(struct audio_softc *, int); 626 static int au_get_monitor_gain(struct audio_softc *); 627 static int audio_get_port(struct audio_softc *, mixer_ctrl_t *); 628 static int audio_set_port(struct audio_softc *, mixer_ctrl_t *); 629 630 static __inline struct audio_params 631 format2_to_params(const audio_format2_t *f2) 632 { 633 audio_params_t p; 634 635 /* validbits/precision <-> precision/stride */ 636 p.sample_rate = f2->sample_rate; 637 p.channels = f2->channels; 638 p.encoding = f2->encoding; 639 p.validbits = f2->precision; 640 p.precision = f2->stride; 641 return p; 642 } 643 644 static __inline audio_format2_t 645 params_to_format2(const struct audio_params *p) 646 { 647 audio_format2_t f2; 648 649 /* precision/stride <-> validbits/precision */ 650 f2.sample_rate = p->sample_rate; 651 f2.channels = p->channels; 652 f2.encoding = p->encoding; 653 f2.precision = p->validbits; 654 f2.stride = p->precision; 655 return f2; 656 } 657 658 /* Return true if this track is a playback track. */ 659 static __inline bool 660 audio_track_is_playback(const audio_track_t *track) 661 { 662 663 return ((track->mode & AUMODE_PLAY) != 0); 664 } 665 666 /* Return true if this track is a recording track. */ 667 static __inline bool 668 audio_track_is_record(const audio_track_t *track) 669 { 670 671 return ((track->mode & AUMODE_RECORD) != 0); 672 } 673 674 #if 0 /* XXX Not used yet */ 675 /* 676 * Convert 0..255 volume used in userland to internal presentation 0..256. 677 */ 678 static __inline u_int 679 audio_volume_to_inner(u_int v) 680 { 681 682 return v < 127 ? v : v + 1; 683 } 684 685 /* 686 * Convert 0..256 internal presentation to 0..255 volume used in userland. 687 */ 688 static __inline u_int 689 audio_volume_to_outer(u_int v) 690 { 691 692 return v < 127 ? v : v - 1; 693 } 694 #endif /* 0 */ 695 696 static dev_type_open(audioopen); 697 /* XXXMRG use more dev_type_xxx */ 698 699 const struct cdevsw audio_cdevsw = { 700 .d_open = audioopen, 701 .d_close = noclose, 702 .d_read = noread, 703 .d_write = nowrite, 704 .d_ioctl = noioctl, 705 .d_stop = nostop, 706 .d_tty = notty, 707 .d_poll = nopoll, 708 .d_mmap = nommap, 709 .d_kqfilter = nokqfilter, 710 .d_discard = nodiscard, 711 .d_flag = D_OTHER | D_MPSAFE 712 }; 713 714 const struct fileops audio_fileops = { 715 .fo_name = "audio", 716 .fo_read = audioread, 717 .fo_write = audiowrite, 718 .fo_ioctl = audioioctl, 719 .fo_fcntl = fnullop_fcntl, 720 .fo_stat = audiostat, 721 .fo_poll = audiopoll, 722 .fo_close = audioclose, 723 .fo_mmap = audiommap, 724 .fo_kqfilter = audiokqfilter, 725 .fo_restart = fnullop_restart 726 }; 727 728 /* The default audio mode: 8 kHz mono mu-law */ 729 static const struct audio_params audio_default = { 730 .sample_rate = 8000, 731 .encoding = AUDIO_ENCODING_ULAW, 732 .precision = 8, 733 .validbits = 8, 734 .channels = 1, 735 }; 736 737 static const char *encoding_names[] = { 738 "none", 739 AudioEmulaw, 740 AudioEalaw, 741 "pcm16", 742 "pcm8", 743 AudioEadpcm, 744 AudioEslinear_le, 745 AudioEslinear_be, 746 AudioEulinear_le, 747 AudioEulinear_be, 748 AudioEslinear, 749 AudioEulinear, 750 AudioEmpeg_l1_stream, 751 AudioEmpeg_l1_packets, 752 AudioEmpeg_l1_system, 753 AudioEmpeg_l2_stream, 754 AudioEmpeg_l2_packets, 755 AudioEmpeg_l2_system, 756 AudioEac3, 757 }; 758 759 /* 760 * Returns encoding name corresponding to AUDIO_ENCODING_*. 761 * Note that it may return a local buffer because it is mainly for debugging. 762 */ 763 const char * 764 audio_encoding_name(int encoding) 765 { 766 static char buf[16]; 767 768 if (0 <= encoding && encoding < __arraycount(encoding_names)) { 769 return encoding_names[encoding]; 770 } else { 771 snprintf(buf, sizeof(buf), "enc=%d", encoding); 772 return buf; 773 } 774 } 775 776 /* 777 * Supported encodings used by AUDIO_GETENC. 778 * index and flags are set by code. 779 * XXX is there any needs for SLINEAR_OE:>=16/ULINEAR_OE:>=16 ? 780 */ 781 static const audio_encoding_t audio_encodings[] = { 782 { 0, AudioEmulaw, AUDIO_ENCODING_ULAW, 8, 0 }, 783 { 0, AudioEalaw, AUDIO_ENCODING_ALAW, 8, 0 }, 784 { 0, AudioEslinear, AUDIO_ENCODING_SLINEAR, 8, 0 }, 785 { 0, AudioEulinear, AUDIO_ENCODING_ULINEAR, 8, 0 }, 786 { 0, AudioEslinear_le, AUDIO_ENCODING_SLINEAR_LE, 16, 0 }, 787 { 0, AudioEulinear_le, AUDIO_ENCODING_ULINEAR_LE, 16, 0 }, 788 { 0, AudioEslinear_be, AUDIO_ENCODING_SLINEAR_BE, 16, 0 }, 789 { 0, AudioEulinear_be, AUDIO_ENCODING_ULINEAR_BE, 16, 0 }, 790 #if defined(AUDIO_SUPPORT_LINEAR24) 791 { 0, AudioEslinear_le, AUDIO_ENCODING_SLINEAR_LE, 24, 0 }, 792 { 0, AudioEulinear_le, AUDIO_ENCODING_ULINEAR_LE, 24, 0 }, 793 { 0, AudioEslinear_be, AUDIO_ENCODING_SLINEAR_BE, 24, 0 }, 794 { 0, AudioEulinear_be, AUDIO_ENCODING_ULINEAR_BE, 24, 0 }, 795 #endif 796 { 0, AudioEslinear_le, AUDIO_ENCODING_SLINEAR_LE, 32, 0 }, 797 { 0, AudioEulinear_le, AUDIO_ENCODING_ULINEAR_LE, 32, 0 }, 798 { 0, AudioEslinear_be, AUDIO_ENCODING_SLINEAR_BE, 32, 0 }, 799 { 0, AudioEulinear_be, AUDIO_ENCODING_ULINEAR_BE, 32, 0 }, 800 }; 801 802 static const struct portname itable[] = { 803 { AudioNmicrophone, AUDIO_MICROPHONE }, 804 { AudioNline, AUDIO_LINE_IN }, 805 { AudioNcd, AUDIO_CD }, 806 { 0, 0 } 807 }; 808 static const struct portname otable[] = { 809 { AudioNspeaker, AUDIO_SPEAKER }, 810 { AudioNheadphone, AUDIO_HEADPHONE }, 811 { AudioNline, AUDIO_LINE_OUT }, 812 { 0, 0 } 813 }; 814 815 static struct psref_class *audio_psref_class __read_mostly; 816 817 CFATTACH_DECL3_NEW(audio, sizeof(struct audio_softc), 818 audiomatch, audioattach, audiodetach, audioactivate, audiorescan, 819 audiochilddet, DVF_DETACH_SHUTDOWN); 820 821 static int 822 audiomatch(device_t parent, cfdata_t match, void *aux) 823 { 824 struct audio_attach_args *sa; 825 826 sa = aux; 827 DPRINTF(1, "%s: type=%d sa=%p hw=%p\n", 828 __func__, sa->type, sa, sa->hwif); 829 return (sa->type == AUDIODEV_TYPE_AUDIO) ? 1 : 0; 830 } 831 832 static void 833 audioattach(device_t parent, device_t self, void *aux) 834 { 835 struct audio_softc *sc; 836 struct audio_attach_args *sa; 837 const struct audio_hw_if *hw_if; 838 audio_format2_t phwfmt; 839 audio_format2_t rhwfmt; 840 audio_filter_reg_t pfil; 841 audio_filter_reg_t rfil; 842 const struct sysctlnode *node; 843 void *hdlp; 844 bool has_playback; 845 bool has_capture; 846 bool has_indep; 847 bool has_fulldup; 848 int mode; 849 int error; 850 851 sc = device_private(self); 852 sc->sc_dev = self; 853 sa = (struct audio_attach_args *)aux; 854 hw_if = sa->hwif; 855 hdlp = sa->hdl; 856 857 if (hw_if == NULL) { 858 panic("audioattach: missing hw_if method"); 859 } 860 if (hw_if->get_locks == NULL || hw_if->get_props == NULL) { 861 aprint_error(": missing mandatory method\n"); 862 return; 863 } 864 865 hw_if->get_locks(hdlp, &sc->sc_intr_lock, &sc->sc_lock); 866 sc->sc_props = hw_if->get_props(hdlp); 867 868 has_playback = (sc->sc_props & AUDIO_PROP_PLAYBACK); 869 has_capture = (sc->sc_props & AUDIO_PROP_CAPTURE); 870 has_indep = (sc->sc_props & AUDIO_PROP_INDEPENDENT); 871 has_fulldup = (sc->sc_props & AUDIO_PROP_FULLDUPLEX); 872 873 #ifdef DIAGNOSTIC 874 if (hw_if->query_format == NULL || 875 hw_if->set_format == NULL || 876 hw_if->getdev == NULL || 877 hw_if->set_port == NULL || 878 hw_if->get_port == NULL || 879 hw_if->query_devinfo == NULL) { 880 aprint_error(": missing mandatory method\n"); 881 return; 882 } 883 if (has_playback) { 884 if ((hw_if->start_output == NULL && hw_if->trigger_output == NULL) || 885 hw_if->halt_output == NULL) { 886 aprint_error(": missing playback method\n"); 887 } 888 } 889 if (has_capture) { 890 if ((hw_if->start_input == NULL && hw_if->trigger_input == NULL) || 891 hw_if->halt_input == NULL) { 892 aprint_error(": missing capture method\n"); 893 } 894 } 895 #endif 896 897 sc->hw_if = hw_if; 898 sc->hw_hdl = hdlp; 899 sc->hw_dev = parent; 900 901 sc->sc_blk_ms = AUDIO_BLK_MS; 902 SLIST_INIT(&sc->sc_files); 903 cv_init(&sc->sc_exlockcv, "audiolk"); 904 sc->sc_am_capacity = 0; 905 sc->sc_am_used = 0; 906 sc->sc_am = NULL; 907 908 /* MMAP is now supported by upper layer. */ 909 sc->sc_props |= AUDIO_PROP_MMAP; 910 911 KASSERT(has_playback || has_capture); 912 /* Unidirectional device must have neither FULLDUP nor INDEPENDENT. */ 913 if (!has_playback || !has_capture) { 914 KASSERT(!has_indep); 915 KASSERT(!has_fulldup); 916 } 917 918 mode = 0; 919 if (has_playback) { 920 aprint_normal(": playback"); 921 mode |= AUMODE_PLAY; 922 } 923 if (has_capture) { 924 aprint_normal("%c capture", has_playback ? ',' : ':'); 925 mode |= AUMODE_RECORD; 926 } 927 if (has_playback && has_capture) { 928 if (has_fulldup) 929 aprint_normal(", full duplex"); 930 else 931 aprint_normal(", half duplex"); 932 933 if (has_indep) 934 aprint_normal(", independent"); 935 } 936 937 aprint_naive("\n"); 938 aprint_normal("\n"); 939 940 /* probe hw params */ 941 memset(&phwfmt, 0, sizeof(phwfmt)); 942 memset(&rhwfmt, 0, sizeof(rhwfmt)); 943 memset(&pfil, 0, sizeof(pfil)); 944 memset(&rfil, 0, sizeof(rfil)); 945 if (has_indep) { 946 int perror, rerror; 947 948 /* On independent devices, probe separately. */ 949 perror = audio_hw_probe(sc, &phwfmt, AUMODE_PLAY); 950 rerror = audio_hw_probe(sc, &rhwfmt, AUMODE_RECORD); 951 if (perror && rerror) { 952 aprint_error_dev(self, "audio_hw_probe failed, " 953 "perror = %d, rerror = %d\n", perror, rerror); 954 goto bad; 955 } 956 if (perror) { 957 mode &= ~AUMODE_PLAY; 958 aprint_error_dev(self, "audio_hw_probe failed with " 959 "%d, playback disabled\n", perror); 960 } 961 if (rerror) { 962 mode &= ~AUMODE_RECORD; 963 aprint_error_dev(self, "audio_hw_probe failed with " 964 "%d, capture disabled\n", rerror); 965 } 966 } else { 967 /* 968 * On non independent devices or uni-directional devices, 969 * probe once (simultaneously). 970 */ 971 audio_format2_t *fmt = has_playback ? &phwfmt : &rhwfmt; 972 error = audio_hw_probe(sc, fmt, mode); 973 if (error) { 974 aprint_error_dev(self, "audio_hw_probe failed, " 975 "error = %d\n", error); 976 goto bad; 977 } 978 if (has_playback && has_capture) 979 rhwfmt = phwfmt; 980 } 981 982 /* Init hardware. */ 983 /* hw_probe() also validates [pr]hwfmt. */ 984 mutex_enter(sc->sc_lock); 985 error = audio_hw_set_format(sc, mode, &phwfmt, &rhwfmt, &pfil, &rfil); 986 if (error) { 987 mutex_exit(sc->sc_lock); 988 aprint_error_dev(self, "audio_hw_set_format failed, " 989 "error = %d\n", error); 990 goto bad; 991 } 992 993 /* 994 * Init track mixers. If at least one direction is available on 995 * attach time, we assume a success. 996 */ 997 error = audio_mixers_init(sc, mode, &phwfmt, &rhwfmt, &pfil, &rfil); 998 mutex_exit(sc->sc_lock); 999 if (sc->sc_pmixer == NULL && sc->sc_rmixer == NULL) { 1000 aprint_error_dev(self, "audio_mixers_init failed, " 1001 "error = %d\n", error); 1002 goto bad; 1003 } 1004 1005 sc->sc_psz = pserialize_create(); 1006 psref_target_init(&sc->sc_psref, audio_psref_class); 1007 1008 selinit(&sc->sc_wsel); 1009 selinit(&sc->sc_rsel); 1010 1011 /* Initial parameter of /dev/sound */ 1012 sc->sc_sound_pparams = params_to_format2(&audio_default); 1013 sc->sc_sound_rparams = params_to_format2(&audio_default); 1014 sc->sc_sound_ppause = false; 1015 sc->sc_sound_rpause = false; 1016 1017 /* XXX TODO: consider about sc_ai */ 1018 1019 mixer_init(sc); 1020 TRACE(2, "inputs ports=0x%x, input master=%d, " 1021 "output ports=0x%x, output master=%d", 1022 sc->sc_inports.allports, sc->sc_inports.master, 1023 sc->sc_outports.allports, sc->sc_outports.master); 1024 1025 sysctl_createv(&sc->sc_log, 0, NULL, &node, 1026 0, 1027 CTLTYPE_NODE, device_xname(sc->sc_dev), 1028 SYSCTL_DESCR("audio test"), 1029 NULL, 0, 1030 NULL, 0, 1031 CTL_HW, 1032 CTL_CREATE, CTL_EOL); 1033 1034 if (node != NULL) { 1035 sysctl_createv(&sc->sc_log, 0, NULL, NULL, 1036 CTLFLAG_READWRITE, 1037 CTLTYPE_INT, "blk_ms", 1038 SYSCTL_DESCR("blocksize in msec"), 1039 audio_sysctl_blk_ms, 0, (void *)sc, 0, 1040 CTL_HW, node->sysctl_num, CTL_CREATE, CTL_EOL); 1041 1042 sysctl_createv(&sc->sc_log, 0, NULL, NULL, 1043 CTLFLAG_READWRITE, 1044 CTLTYPE_BOOL, "multiuser", 1045 SYSCTL_DESCR("allow multiple user access"), 1046 audio_sysctl_multiuser, 0, (void *)sc, 0, 1047 CTL_HW, node->sysctl_num, CTL_CREATE, CTL_EOL); 1048 1049 #if defined(AUDIO_DEBUG) 1050 sysctl_createv(&sc->sc_log, 0, NULL, NULL, 1051 CTLFLAG_READWRITE, 1052 CTLTYPE_INT, "debug", 1053 SYSCTL_DESCR("debug level (0..4)"), 1054 audio_sysctl_debug, 0, (void *)sc, 0, 1055 CTL_HW, node->sysctl_num, CTL_CREATE, CTL_EOL); 1056 #endif 1057 } 1058 1059 #ifdef AUDIO_PM_IDLE 1060 callout_init(&sc->sc_idle_counter, 0); 1061 callout_setfunc(&sc->sc_idle_counter, audio_idle, self); 1062 #endif 1063 1064 if (!pmf_device_register(self, audio_suspend, audio_resume)) 1065 aprint_error_dev(self, "couldn't establish power handler\n"); 1066 #ifdef AUDIO_PM_IDLE 1067 if (!device_active_register(self, audio_activity)) 1068 aprint_error_dev(self, "couldn't register activity handler\n"); 1069 #endif 1070 1071 if (!pmf_event_register(self, PMFE_AUDIO_VOLUME_DOWN, 1072 audio_volume_down, true)) 1073 aprint_error_dev(self, "couldn't add volume down handler\n"); 1074 if (!pmf_event_register(self, PMFE_AUDIO_VOLUME_UP, 1075 audio_volume_up, true)) 1076 aprint_error_dev(self, "couldn't add volume up handler\n"); 1077 if (!pmf_event_register(self, PMFE_AUDIO_VOLUME_TOGGLE, 1078 audio_volume_toggle, true)) 1079 aprint_error_dev(self, "couldn't add volume toggle handler\n"); 1080 1081 #ifdef AUDIO_PM_IDLE 1082 callout_schedule(&sc->sc_idle_counter, audio_idle_timeout * hz); 1083 #endif 1084 1085 #if defined(AUDIO_DEBUG) 1086 audio_mlog_init(); 1087 #endif 1088 1089 audiorescan(self, "audio", NULL); 1090 return; 1091 1092 bad: 1093 /* Clearing hw_if means that device is attached but disabled. */ 1094 sc->hw_if = NULL; 1095 aprint_error_dev(sc->sc_dev, "disabled\n"); 1096 return; 1097 } 1098 1099 /* 1100 * Initialize hardware mixer. 1101 * This function is called from audioattach(). 1102 */ 1103 static void 1104 mixer_init(struct audio_softc *sc) 1105 { 1106 mixer_devinfo_t mi; 1107 int iclass, mclass, oclass, rclass; 1108 int record_master_found, record_source_found; 1109 1110 iclass = mclass = oclass = rclass = -1; 1111 sc->sc_inports.index = -1; 1112 sc->sc_inports.master = -1; 1113 sc->sc_inports.nports = 0; 1114 sc->sc_inports.isenum = false; 1115 sc->sc_inports.allports = 0; 1116 sc->sc_inports.isdual = false; 1117 sc->sc_inports.mixerout = -1; 1118 sc->sc_inports.cur_port = -1; 1119 sc->sc_outports.index = -1; 1120 sc->sc_outports.master = -1; 1121 sc->sc_outports.nports = 0; 1122 sc->sc_outports.isenum = false; 1123 sc->sc_outports.allports = 0; 1124 sc->sc_outports.isdual = false; 1125 sc->sc_outports.mixerout = -1; 1126 sc->sc_outports.cur_port = -1; 1127 sc->sc_monitor_port = -1; 1128 /* 1129 * Read through the underlying driver's list, picking out the class 1130 * names from the mixer descriptions. We'll need them to decode the 1131 * mixer descriptions on the next pass through the loop. 1132 */ 1133 mutex_enter(sc->sc_lock); 1134 for(mi.index = 0; ; mi.index++) { 1135 if (audio_query_devinfo(sc, &mi) != 0) 1136 break; 1137 /* 1138 * The type of AUDIO_MIXER_CLASS merely introduces a class. 1139 * All the other types describe an actual mixer. 1140 */ 1141 if (mi.type == AUDIO_MIXER_CLASS) { 1142 if (strcmp(mi.label.name, AudioCinputs) == 0) 1143 iclass = mi.mixer_class; 1144 if (strcmp(mi.label.name, AudioCmonitor) == 0) 1145 mclass = mi.mixer_class; 1146 if (strcmp(mi.label.name, AudioCoutputs) == 0) 1147 oclass = mi.mixer_class; 1148 if (strcmp(mi.label.name, AudioCrecord) == 0) 1149 rclass = mi.mixer_class; 1150 } 1151 } 1152 mutex_exit(sc->sc_lock); 1153 1154 /* Allocate save area. Ensure non-zero allocation. */ 1155 sc->sc_nmixer_states = mi.index; 1156 sc->sc_mixer_state = kmem_zalloc(sizeof(mixer_ctrl_t) * 1157 (sc->sc_nmixer_states + 1), KM_SLEEP); 1158 1159 /* 1160 * This is where we assign each control in the "audio" model, to the 1161 * underlying "mixer" control. We walk through the whole list once, 1162 * assigning likely candidates as we come across them. 1163 */ 1164 record_master_found = 0; 1165 record_source_found = 0; 1166 mutex_enter(sc->sc_lock); 1167 for(mi.index = 0; ; mi.index++) { 1168 if (audio_query_devinfo(sc, &mi) != 0) 1169 break; 1170 KASSERT(mi.index < sc->sc_nmixer_states); 1171 if (mi.type == AUDIO_MIXER_CLASS) 1172 continue; 1173 if (mi.mixer_class == iclass) { 1174 /* 1175 * AudioCinputs is only a fallback, when we don't 1176 * find what we're looking for in AudioCrecord, so 1177 * check the flags before accepting one of these. 1178 */ 1179 if (strcmp(mi.label.name, AudioNmaster) == 0 1180 && record_master_found == 0) 1181 sc->sc_inports.master = mi.index; 1182 if (strcmp(mi.label.name, AudioNsource) == 0 1183 && record_source_found == 0) { 1184 if (mi.type == AUDIO_MIXER_ENUM) { 1185 int i; 1186 for(i = 0; i < mi.un.e.num_mem; i++) 1187 if (strcmp(mi.un.e.member[i].label.name, 1188 AudioNmixerout) == 0) 1189 sc->sc_inports.mixerout = 1190 mi.un.e.member[i].ord; 1191 } 1192 au_setup_ports(sc, &sc->sc_inports, &mi, 1193 itable); 1194 } 1195 if (strcmp(mi.label.name, AudioNdac) == 0 && 1196 sc->sc_outports.master == -1) 1197 sc->sc_outports.master = mi.index; 1198 } else if (mi.mixer_class == mclass) { 1199 if (strcmp(mi.label.name, AudioNmonitor) == 0) 1200 sc->sc_monitor_port = mi.index; 1201 } else if (mi.mixer_class == oclass) { 1202 if (strcmp(mi.label.name, AudioNmaster) == 0) 1203 sc->sc_outports.master = mi.index; 1204 if (strcmp(mi.label.name, AudioNselect) == 0) 1205 au_setup_ports(sc, &sc->sc_outports, &mi, 1206 otable); 1207 } else if (mi.mixer_class == rclass) { 1208 /* 1209 * These are the preferred mixers for the audio record 1210 * controls, so set the flags here, but don't check. 1211 */ 1212 if (strcmp(mi.label.name, AudioNmaster) == 0) { 1213 sc->sc_inports.master = mi.index; 1214 record_master_found = 1; 1215 } 1216 #if 1 /* Deprecated. Use AudioNmaster. */ 1217 if (strcmp(mi.label.name, AudioNrecord) == 0) { 1218 sc->sc_inports.master = mi.index; 1219 record_master_found = 1; 1220 } 1221 if (strcmp(mi.label.name, AudioNvolume) == 0) { 1222 sc->sc_inports.master = mi.index; 1223 record_master_found = 1; 1224 } 1225 #endif 1226 if (strcmp(mi.label.name, AudioNsource) == 0) { 1227 if (mi.type == AUDIO_MIXER_ENUM) { 1228 int i; 1229 for(i = 0; i < mi.un.e.num_mem; i++) 1230 if (strcmp(mi.un.e.member[i].label.name, 1231 AudioNmixerout) == 0) 1232 sc->sc_inports.mixerout = 1233 mi.un.e.member[i].ord; 1234 } 1235 au_setup_ports(sc, &sc->sc_inports, &mi, 1236 itable); 1237 record_source_found = 1; 1238 } 1239 } 1240 } 1241 mutex_exit(sc->sc_lock); 1242 } 1243 1244 static int 1245 audioactivate(device_t self, enum devact act) 1246 { 1247 struct audio_softc *sc = device_private(self); 1248 1249 switch (act) { 1250 case DVACT_DEACTIVATE: 1251 mutex_enter(sc->sc_lock); 1252 sc->sc_dying = true; 1253 cv_broadcast(&sc->sc_exlockcv); 1254 mutex_exit(sc->sc_lock); 1255 return 0; 1256 default: 1257 return EOPNOTSUPP; 1258 } 1259 } 1260 1261 static int 1262 audiodetach(device_t self, int flags) 1263 { 1264 struct audio_softc *sc; 1265 struct audio_file *file; 1266 int error; 1267 1268 sc = device_private(self); 1269 TRACE(2, "flags=%d", flags); 1270 1271 /* device is not initialized */ 1272 if (sc->hw_if == NULL) 1273 return 0; 1274 1275 /* Start draining existing accessors of the device. */ 1276 error = config_detach_children(self, flags); 1277 if (error) 1278 return error; 1279 1280 /* delete sysctl nodes */ 1281 sysctl_teardown(&sc->sc_log); 1282 1283 mutex_enter(sc->sc_lock); 1284 sc->sc_dying = true; 1285 cv_broadcast(&sc->sc_exlockcv); 1286 if (sc->sc_pmixer) 1287 cv_broadcast(&sc->sc_pmixer->outcv); 1288 if (sc->sc_rmixer) 1289 cv_broadcast(&sc->sc_rmixer->outcv); 1290 1291 /* Prevent new users */ 1292 SLIST_FOREACH(file, &sc->sc_files, entry) { 1293 atomic_store_relaxed(&file->dying, true); 1294 } 1295 1296 /* 1297 * Wait for existing users to drain. 1298 * - pserialize_perform waits for all pserialize_read sections on 1299 * all CPUs; after this, no more new psref_acquire can happen. 1300 * - psref_target_destroy waits for all extant acquired psrefs to 1301 * be psref_released. 1302 */ 1303 pserialize_perform(sc->sc_psz); 1304 mutex_exit(sc->sc_lock); 1305 psref_target_destroy(&sc->sc_psref, audio_psref_class); 1306 1307 /* 1308 * We are now guaranteed that there are no calls to audio fileops 1309 * that hold sc, and any new calls with files that were for sc will 1310 * fail. Thus, we now have exclusive access to the softc. 1311 */ 1312 1313 /* 1314 * Nuke all open instances. 1315 * Here, we no longer need any locks to traverse sc_files. 1316 */ 1317 while ((file = SLIST_FIRST(&sc->sc_files)) != NULL) { 1318 audio_unlink(sc, file); 1319 } 1320 1321 pmf_event_deregister(self, PMFE_AUDIO_VOLUME_DOWN, 1322 audio_volume_down, true); 1323 pmf_event_deregister(self, PMFE_AUDIO_VOLUME_UP, 1324 audio_volume_up, true); 1325 pmf_event_deregister(self, PMFE_AUDIO_VOLUME_TOGGLE, 1326 audio_volume_toggle, true); 1327 1328 #ifdef AUDIO_PM_IDLE 1329 callout_halt(&sc->sc_idle_counter, sc->sc_lock); 1330 1331 device_active_deregister(self, audio_activity); 1332 #endif 1333 1334 pmf_device_deregister(self); 1335 1336 /* Free resources */ 1337 mutex_enter(sc->sc_lock); 1338 if (sc->sc_pmixer) { 1339 audio_mixer_destroy(sc, sc->sc_pmixer); 1340 kmem_free(sc->sc_pmixer, sizeof(*sc->sc_pmixer)); 1341 } 1342 if (sc->sc_rmixer) { 1343 audio_mixer_destroy(sc, sc->sc_rmixer); 1344 kmem_free(sc->sc_rmixer, sizeof(*sc->sc_rmixer)); 1345 } 1346 mutex_exit(sc->sc_lock); 1347 if (sc->sc_am) 1348 kern_free(sc->sc_am); 1349 1350 seldestroy(&sc->sc_wsel); 1351 seldestroy(&sc->sc_rsel); 1352 1353 #ifdef AUDIO_PM_IDLE 1354 callout_destroy(&sc->sc_idle_counter); 1355 #endif 1356 1357 cv_destroy(&sc->sc_exlockcv); 1358 1359 #if defined(AUDIO_DEBUG) 1360 audio_mlog_free(); 1361 #endif 1362 1363 return 0; 1364 } 1365 1366 static void 1367 audiochilddet(device_t self, device_t child) 1368 { 1369 1370 /* we hold no child references, so do nothing */ 1371 } 1372 1373 static int 1374 audiosearch(device_t parent, cfdata_t cf, const int *locs, void *aux) 1375 { 1376 1377 if (config_match(parent, cf, aux)) 1378 config_attach_loc(parent, cf, locs, aux, NULL); 1379 1380 return 0; 1381 } 1382 1383 static int 1384 audiorescan(device_t self, const char *ifattr, const int *flags) 1385 { 1386 struct audio_softc *sc = device_private(self); 1387 1388 if (!ifattr_match(ifattr, "audio")) 1389 return 0; 1390 1391 config_search_loc(audiosearch, sc->sc_dev, "audio", NULL, NULL); 1392 1393 return 0; 1394 } 1395 1396 /* 1397 * Called from hardware driver. This is where the MI audio driver gets 1398 * probed/attached to the hardware driver. 1399 */ 1400 device_t 1401 audio_attach_mi(const struct audio_hw_if *ahwp, void *hdlp, device_t dev) 1402 { 1403 struct audio_attach_args arg; 1404 1405 #ifdef DIAGNOSTIC 1406 if (ahwp == NULL) { 1407 aprint_error("audio_attach_mi: NULL\n"); 1408 return 0; 1409 } 1410 #endif 1411 arg.type = AUDIODEV_TYPE_AUDIO; 1412 arg.hwif = ahwp; 1413 arg.hdl = hdlp; 1414 return config_found(dev, &arg, audioprint); 1415 } 1416 1417 /* 1418 * Acquire sc_lock and enter exlock critical section. 1419 * If successful, it returns 0. Otherwise returns errno. 1420 * Must be called without sc_lock held. 1421 */ 1422 static int 1423 audio_enter_exclusive(struct audio_softc *sc) 1424 { 1425 int error; 1426 1427 mutex_enter(sc->sc_lock); 1428 if (sc->sc_dying) { 1429 mutex_exit(sc->sc_lock); 1430 return EIO; 1431 } 1432 1433 while (__predict_false(sc->sc_exlock != 0)) { 1434 error = cv_wait_sig(&sc->sc_exlockcv, sc->sc_lock); 1435 if (sc->sc_dying) 1436 error = EIO; 1437 if (error) { 1438 mutex_exit(sc->sc_lock); 1439 return error; 1440 } 1441 } 1442 1443 /* Acquire */ 1444 sc->sc_exlock = 1; 1445 return 0; 1446 } 1447 1448 /* 1449 * Leave exlock critical section and release sc_lock. 1450 * Must be called with sc_lock held. 1451 */ 1452 static void 1453 audio_exit_exclusive(struct audio_softc *sc) 1454 { 1455 1456 KASSERT(mutex_owned(sc->sc_lock)); 1457 KASSERT(sc->sc_exlock); 1458 1459 /* Leave critical section */ 1460 sc->sc_exlock = 0; 1461 cv_broadcast(&sc->sc_exlockcv); 1462 mutex_exit(sc->sc_lock); 1463 } 1464 1465 /* 1466 * Acquire sc from file, and increment the psref count. 1467 * If successful, returns sc. Otherwise returns NULL. 1468 */ 1469 struct audio_softc * 1470 audio_file_enter(audio_file_t *file, struct psref *refp) 1471 { 1472 int s; 1473 bool dying; 1474 1475 /* psref(9) forbids to migrate CPUs */ 1476 curlwp_bind(); 1477 1478 /* Block audiodetach while we acquire a reference */ 1479 s = pserialize_read_enter(); 1480 1481 /* If close or audiodetach already ran, tough -- no more audio */ 1482 dying = atomic_load_relaxed(&file->dying); 1483 if (dying) { 1484 pserialize_read_exit(s); 1485 return NULL; 1486 } 1487 1488 /* Acquire a reference */ 1489 psref_acquire(refp, &file->sc->sc_psref, audio_psref_class); 1490 1491 /* Now sc won't go away until we drop the reference count */ 1492 pserialize_read_exit(s); 1493 1494 return file->sc; 1495 } 1496 1497 /* 1498 * Decrement the psref count. 1499 */ 1500 void 1501 audio_file_exit(struct audio_softc *sc, struct psref *refp) 1502 { 1503 1504 psref_release(refp, &sc->sc_psref, audio_psref_class); 1505 } 1506 1507 /* 1508 * Wait for I/O to complete, releasing sc_lock. 1509 * Must be called with sc_lock held. 1510 */ 1511 static int 1512 audio_track_waitio(struct audio_softc *sc, audio_track_t *track) 1513 { 1514 int error; 1515 1516 KASSERT(track); 1517 KASSERT(mutex_owned(sc->sc_lock)); 1518 1519 /* Wait for pending I/O to complete. */ 1520 error = cv_timedwait_sig(&track->mixer->outcv, sc->sc_lock, 1521 mstohz(AUDIO_TIMEOUT)); 1522 if (sc->sc_dying) { 1523 error = EIO; 1524 } 1525 if (error) { 1526 TRACET(2, track, "cv_timedwait_sig failed %d", error); 1527 if (error == EWOULDBLOCK) 1528 device_printf(sc->sc_dev, "device timeout\n"); 1529 } else { 1530 TRACET(3, track, "wakeup"); 1531 } 1532 return error; 1533 } 1534 1535 /* 1536 * Try to acquire track lock. 1537 * It doesn't block if the track lock is already aquired. 1538 * Returns true if the track lock was acquired, or false if the track 1539 * lock was already acquired. 1540 */ 1541 static __inline bool 1542 audio_track_lock_tryenter(audio_track_t *track) 1543 { 1544 return (atomic_cas_uint(&track->lock, 0, 1) == 0); 1545 } 1546 1547 /* 1548 * Acquire track lock. 1549 */ 1550 static __inline void 1551 audio_track_lock_enter(audio_track_t *track) 1552 { 1553 /* Don't sleep here. */ 1554 while (audio_track_lock_tryenter(track) == false) 1555 ; 1556 } 1557 1558 /* 1559 * Release track lock. 1560 */ 1561 static __inline void 1562 audio_track_lock_exit(audio_track_t *track) 1563 { 1564 atomic_swap_uint(&track->lock, 0); 1565 } 1566 1567 1568 static int 1569 audioopen(dev_t dev, int flags, int ifmt, struct lwp *l) 1570 { 1571 struct audio_softc *sc; 1572 int error; 1573 1574 /* Find the device */ 1575 sc = device_lookup_private(&audio_cd, AUDIOUNIT(dev)); 1576 if (sc == NULL || sc->hw_if == NULL) 1577 return ENXIO; 1578 1579 error = audio_enter_exclusive(sc); 1580 if (error) 1581 return error; 1582 1583 device_active(sc->sc_dev, DVA_SYSTEM); 1584 switch (AUDIODEV(dev)) { 1585 case SOUND_DEVICE: 1586 case AUDIO_DEVICE: 1587 error = audio_open(dev, sc, flags, ifmt, l, NULL); 1588 break; 1589 case AUDIOCTL_DEVICE: 1590 error = audioctl_open(dev, sc, flags, ifmt, l); 1591 break; 1592 case MIXER_DEVICE: 1593 error = mixer_open(dev, sc, flags, ifmt, l); 1594 break; 1595 default: 1596 error = ENXIO; 1597 break; 1598 } 1599 audio_exit_exclusive(sc); 1600 1601 return error; 1602 } 1603 1604 static int 1605 audioclose(struct file *fp) 1606 { 1607 struct audio_softc *sc; 1608 struct psref sc_ref; 1609 audio_file_t *file; 1610 int error; 1611 dev_t dev; 1612 1613 KASSERT(fp->f_audioctx); 1614 file = fp->f_audioctx; 1615 dev = file->dev; 1616 error = 0; 1617 1618 /* 1619 * audioclose() must 1620 * - unplug track from the trackmixer (and unplug anything from softc), 1621 * if sc exists. 1622 * - free all memory objects, regardless of sc. 1623 */ 1624 1625 sc = audio_file_enter(file, &sc_ref); 1626 if (sc) { 1627 switch (AUDIODEV(dev)) { 1628 case SOUND_DEVICE: 1629 case AUDIO_DEVICE: 1630 error = audio_close(sc, file); 1631 break; 1632 case AUDIOCTL_DEVICE: 1633 error = 0; 1634 break; 1635 case MIXER_DEVICE: 1636 error = mixer_close(sc, file); 1637 break; 1638 default: 1639 error = ENXIO; 1640 break; 1641 } 1642 1643 audio_file_exit(sc, &sc_ref); 1644 } 1645 1646 /* Free memory objects anyway */ 1647 TRACEF(2, file, "free memory"); 1648 if (file->ptrack) 1649 audio_track_destroy(file->ptrack); 1650 if (file->rtrack) 1651 audio_track_destroy(file->rtrack); 1652 kmem_free(file, sizeof(*file)); 1653 fp->f_audioctx = NULL; 1654 1655 return error; 1656 } 1657 1658 static int 1659 audioread(struct file *fp, off_t *offp, struct uio *uio, kauth_cred_t cred, 1660 int ioflag) 1661 { 1662 struct audio_softc *sc; 1663 struct psref sc_ref; 1664 audio_file_t *file; 1665 int error; 1666 dev_t dev; 1667 1668 KASSERT(fp->f_audioctx); 1669 file = fp->f_audioctx; 1670 dev = file->dev; 1671 1672 sc = audio_file_enter(file, &sc_ref); 1673 if (sc == NULL) 1674 return EIO; 1675 1676 if (fp->f_flag & O_NONBLOCK) 1677 ioflag |= IO_NDELAY; 1678 1679 switch (AUDIODEV(dev)) { 1680 case SOUND_DEVICE: 1681 case AUDIO_DEVICE: 1682 error = audio_read(sc, uio, ioflag, file); 1683 break; 1684 case AUDIOCTL_DEVICE: 1685 case MIXER_DEVICE: 1686 error = ENODEV; 1687 break; 1688 default: 1689 error = ENXIO; 1690 break; 1691 } 1692 1693 audio_file_exit(sc, &sc_ref); 1694 return error; 1695 } 1696 1697 static int 1698 audiowrite(struct file *fp, off_t *offp, struct uio *uio, kauth_cred_t cred, 1699 int ioflag) 1700 { 1701 struct audio_softc *sc; 1702 struct psref sc_ref; 1703 audio_file_t *file; 1704 int error; 1705 dev_t dev; 1706 1707 KASSERT(fp->f_audioctx); 1708 file = fp->f_audioctx; 1709 dev = file->dev; 1710 1711 sc = audio_file_enter(file, &sc_ref); 1712 if (sc == NULL) 1713 return EIO; 1714 1715 if (fp->f_flag & O_NONBLOCK) 1716 ioflag |= IO_NDELAY; 1717 1718 switch (AUDIODEV(dev)) { 1719 case SOUND_DEVICE: 1720 case AUDIO_DEVICE: 1721 error = audio_write(sc, uio, ioflag, file); 1722 break; 1723 case AUDIOCTL_DEVICE: 1724 case MIXER_DEVICE: 1725 error = ENODEV; 1726 break; 1727 default: 1728 error = ENXIO; 1729 break; 1730 } 1731 1732 audio_file_exit(sc, &sc_ref); 1733 return error; 1734 } 1735 1736 static int 1737 audioioctl(struct file *fp, u_long cmd, void *addr) 1738 { 1739 struct audio_softc *sc; 1740 struct psref sc_ref; 1741 audio_file_t *file; 1742 struct lwp *l = curlwp; 1743 int error; 1744 dev_t dev; 1745 1746 KASSERT(fp->f_audioctx); 1747 file = fp->f_audioctx; 1748 dev = file->dev; 1749 1750 sc = audio_file_enter(file, &sc_ref); 1751 if (sc == NULL) 1752 return EIO; 1753 1754 switch (AUDIODEV(dev)) { 1755 case SOUND_DEVICE: 1756 case AUDIO_DEVICE: 1757 case AUDIOCTL_DEVICE: 1758 mutex_enter(sc->sc_lock); 1759 device_active(sc->sc_dev, DVA_SYSTEM); 1760 mutex_exit(sc->sc_lock); 1761 if (IOCGROUP(cmd) == IOCGROUP(AUDIO_MIXER_READ)) 1762 error = mixer_ioctl(sc, cmd, addr, fp->f_flag, l); 1763 else 1764 error = audio_ioctl(dev, sc, cmd, addr, fp->f_flag, l, 1765 file); 1766 break; 1767 case MIXER_DEVICE: 1768 error = mixer_ioctl(sc, cmd, addr, fp->f_flag, l); 1769 break; 1770 default: 1771 error = ENXIO; 1772 break; 1773 } 1774 1775 audio_file_exit(sc, &sc_ref); 1776 return error; 1777 } 1778 1779 static int 1780 audiostat(struct file *fp, struct stat *st) 1781 { 1782 struct audio_softc *sc; 1783 struct psref sc_ref; 1784 audio_file_t *file; 1785 1786 KASSERT(fp->f_audioctx); 1787 file = fp->f_audioctx; 1788 1789 sc = audio_file_enter(file, &sc_ref); 1790 if (sc == NULL) 1791 return EIO; 1792 1793 memset(st, 0, sizeof(*st)); 1794 1795 st->st_dev = file->dev; 1796 st->st_uid = kauth_cred_geteuid(fp->f_cred); 1797 st->st_gid = kauth_cred_getegid(fp->f_cred); 1798 st->st_mode = S_IFCHR; 1799 1800 audio_file_exit(sc, &sc_ref); 1801 return 0; 1802 } 1803 1804 static int 1805 audiopoll(struct file *fp, int events) 1806 { 1807 struct audio_softc *sc; 1808 struct psref sc_ref; 1809 audio_file_t *file; 1810 struct lwp *l = curlwp; 1811 int revents; 1812 dev_t dev; 1813 1814 KASSERT(fp->f_audioctx); 1815 file = fp->f_audioctx; 1816 dev = file->dev; 1817 1818 sc = audio_file_enter(file, &sc_ref); 1819 if (sc == NULL) 1820 return EIO; 1821 1822 switch (AUDIODEV(dev)) { 1823 case SOUND_DEVICE: 1824 case AUDIO_DEVICE: 1825 revents = audio_poll(sc, events, l, file); 1826 break; 1827 case AUDIOCTL_DEVICE: 1828 case MIXER_DEVICE: 1829 revents = 0; 1830 break; 1831 default: 1832 revents = POLLERR; 1833 break; 1834 } 1835 1836 audio_file_exit(sc, &sc_ref); 1837 return revents; 1838 } 1839 1840 static int 1841 audiokqfilter(struct file *fp, struct knote *kn) 1842 { 1843 struct audio_softc *sc; 1844 struct psref sc_ref; 1845 audio_file_t *file; 1846 dev_t dev; 1847 int error; 1848 1849 KASSERT(fp->f_audioctx); 1850 file = fp->f_audioctx; 1851 dev = file->dev; 1852 1853 sc = audio_file_enter(file, &sc_ref); 1854 if (sc == NULL) 1855 return EIO; 1856 1857 switch (AUDIODEV(dev)) { 1858 case SOUND_DEVICE: 1859 case AUDIO_DEVICE: 1860 error = audio_kqfilter(sc, file, kn); 1861 break; 1862 case AUDIOCTL_DEVICE: 1863 case MIXER_DEVICE: 1864 error = ENODEV; 1865 break; 1866 default: 1867 error = ENXIO; 1868 break; 1869 } 1870 1871 audio_file_exit(sc, &sc_ref); 1872 return error; 1873 } 1874 1875 static int 1876 audiommap(struct file *fp, off_t *offp, size_t len, int prot, int *flagsp, 1877 int *advicep, struct uvm_object **uobjp, int *maxprotp) 1878 { 1879 struct audio_softc *sc; 1880 struct psref sc_ref; 1881 audio_file_t *file; 1882 dev_t dev; 1883 int error; 1884 1885 KASSERT(fp->f_audioctx); 1886 file = fp->f_audioctx; 1887 dev = file->dev; 1888 1889 sc = audio_file_enter(file, &sc_ref); 1890 if (sc == NULL) 1891 return EIO; 1892 1893 mutex_enter(sc->sc_lock); 1894 device_active(sc->sc_dev, DVA_SYSTEM); /* XXXJDM */ 1895 mutex_exit(sc->sc_lock); 1896 1897 switch (AUDIODEV(dev)) { 1898 case SOUND_DEVICE: 1899 case AUDIO_DEVICE: 1900 error = audio_mmap(sc, offp, len, prot, flagsp, advicep, 1901 uobjp, maxprotp, file); 1902 break; 1903 case AUDIOCTL_DEVICE: 1904 case MIXER_DEVICE: 1905 default: 1906 error = ENOTSUP; 1907 break; 1908 } 1909 1910 audio_file_exit(sc, &sc_ref); 1911 return error; 1912 } 1913 1914 1915 /* Exported interfaces for audiobell. */ 1916 1917 /* 1918 * Open for audiobell. 1919 * It stores allocated file to *filep. 1920 * If successful returns 0, otherwise errno. 1921 */ 1922 int 1923 audiobellopen(dev_t dev, audio_file_t **filep) 1924 { 1925 struct audio_softc *sc; 1926 int error; 1927 1928 /* Find the device */ 1929 sc = device_lookup_private(&audio_cd, AUDIOUNIT(dev)); 1930 if (sc == NULL || sc->hw_if == NULL) 1931 return ENXIO; 1932 1933 error = audio_enter_exclusive(sc); 1934 if (error) 1935 return error; 1936 1937 device_active(sc->sc_dev, DVA_SYSTEM); 1938 error = audio_open(dev, sc, FWRITE, 0, curlwp, filep); 1939 1940 audio_exit_exclusive(sc); 1941 return error; 1942 } 1943 1944 /* Close for audiobell */ 1945 int 1946 audiobellclose(audio_file_t *file) 1947 { 1948 struct audio_softc *sc; 1949 struct psref sc_ref; 1950 int error; 1951 1952 sc = audio_file_enter(file, &sc_ref); 1953 if (sc == NULL) 1954 return EIO; 1955 1956 error = audio_close(sc, file); 1957 1958 audio_file_exit(sc, &sc_ref); 1959 1960 KASSERT(file->ptrack); 1961 audio_track_destroy(file->ptrack); 1962 KASSERT(file->rtrack == NULL); 1963 kmem_free(file, sizeof(*file)); 1964 return error; 1965 } 1966 1967 /* Set sample rate for audiobell */ 1968 int 1969 audiobellsetrate(audio_file_t *file, u_int sample_rate) 1970 { 1971 struct audio_softc *sc; 1972 struct psref sc_ref; 1973 struct audio_info ai; 1974 int error; 1975 1976 sc = audio_file_enter(file, &sc_ref); 1977 if (sc == NULL) 1978 return EIO; 1979 1980 AUDIO_INITINFO(&ai); 1981 ai.play.sample_rate = sample_rate; 1982 1983 error = audio_enter_exclusive(sc); 1984 if (error) 1985 goto done; 1986 error = audio_file_setinfo(sc, file, &ai); 1987 audio_exit_exclusive(sc); 1988 1989 done: 1990 audio_file_exit(sc, &sc_ref); 1991 return error; 1992 } 1993 1994 /* Playback for audiobell */ 1995 int 1996 audiobellwrite(audio_file_t *file, struct uio *uio) 1997 { 1998 struct audio_softc *sc; 1999 struct psref sc_ref; 2000 int error; 2001 2002 sc = audio_file_enter(file, &sc_ref); 2003 if (sc == NULL) 2004 return EIO; 2005 2006 error = audio_write(sc, uio, 0, file); 2007 2008 audio_file_exit(sc, &sc_ref); 2009 return error; 2010 } 2011 2012 2013 /* 2014 * Audio driver 2015 */ 2016 int 2017 audio_open(dev_t dev, struct audio_softc *sc, int flags, int ifmt, 2018 struct lwp *l, audio_file_t **bellfile) 2019 { 2020 struct audio_info ai; 2021 struct file *fp; 2022 audio_file_t *af; 2023 audio_ring_t *hwbuf; 2024 bool fullduplex; 2025 int fd; 2026 int error; 2027 2028 KASSERT(mutex_owned(sc->sc_lock)); 2029 KASSERT(sc->sc_exlock); 2030 2031 TRACE(1, "%sdev=%s flags=0x%x po=%d ro=%d", 2032 (audiodebug >= 3) ? "start " : "", 2033 ISDEVSOUND(dev) ? "sound" : "audio", 2034 flags, sc->sc_popens, sc->sc_ropens); 2035 2036 af = kmem_zalloc(sizeof(audio_file_t), KM_SLEEP); 2037 af->sc = sc; 2038 af->dev = dev; 2039 if ((flags & FWRITE) != 0 && audio_can_playback(sc)) 2040 af->mode |= AUMODE_PLAY | AUMODE_PLAY_ALL; 2041 if ((flags & FREAD) != 0 && audio_can_capture(sc)) 2042 af->mode |= AUMODE_RECORD; 2043 if (af->mode == 0) { 2044 error = ENXIO; 2045 goto bad1; 2046 } 2047 2048 fullduplex = (sc->sc_props & AUDIO_PROP_FULLDUPLEX); 2049 2050 /* 2051 * On half duplex hardware, 2052 * 1. if mode is (PLAY | REC), let mode PLAY. 2053 * 2. if mode is PLAY, let mode PLAY if no rec tracks, otherwise error. 2054 * 3. if mode is REC, let mode REC if no play tracks, otherwise error. 2055 */ 2056 if (fullduplex == false) { 2057 if ((af->mode & AUMODE_PLAY)) { 2058 if (sc->sc_ropens != 0) { 2059 TRACE(1, "record track already exists"); 2060 error = ENODEV; 2061 goto bad1; 2062 } 2063 /* Play takes precedence */ 2064 af->mode &= ~AUMODE_RECORD; 2065 } 2066 if ((af->mode & AUMODE_RECORD)) { 2067 if (sc->sc_popens != 0) { 2068 TRACE(1, "play track already exists"); 2069 error = ENODEV; 2070 goto bad1; 2071 } 2072 } 2073 } 2074 2075 /* Create tracks */ 2076 if ((af->mode & AUMODE_PLAY)) 2077 af->ptrack = audio_track_create(sc, sc->sc_pmixer); 2078 if ((af->mode & AUMODE_RECORD)) 2079 af->rtrack = audio_track_create(sc, sc->sc_rmixer); 2080 2081 /* Set parameters */ 2082 AUDIO_INITINFO(&ai); 2083 if (bellfile) { 2084 /* If audiobell, only sample_rate will be set later. */ 2085 ai.play.sample_rate = audio_default.sample_rate; 2086 ai.play.encoding = AUDIO_ENCODING_SLINEAR_NE; 2087 ai.play.channels = 1; 2088 ai.play.precision = 16; 2089 ai.play.pause = 0; 2090 } else if (ISDEVAUDIO(dev)) { 2091 /* If /dev/audio, initialize everytime. */ 2092 ai.play.sample_rate = audio_default.sample_rate; 2093 ai.play.encoding = audio_default.encoding; 2094 ai.play.channels = audio_default.channels; 2095 ai.play.precision = audio_default.precision; 2096 ai.play.pause = 0; 2097 ai.record.sample_rate = audio_default.sample_rate; 2098 ai.record.encoding = audio_default.encoding; 2099 ai.record.channels = audio_default.channels; 2100 ai.record.precision = audio_default.precision; 2101 ai.record.pause = 0; 2102 } else { 2103 /* If /dev/sound, take over the previous parameters. */ 2104 ai.play.sample_rate = sc->sc_sound_pparams.sample_rate; 2105 ai.play.encoding = sc->sc_sound_pparams.encoding; 2106 ai.play.channels = sc->sc_sound_pparams.channels; 2107 ai.play.precision = sc->sc_sound_pparams.precision; 2108 ai.play.pause = sc->sc_sound_ppause; 2109 ai.record.sample_rate = sc->sc_sound_rparams.sample_rate; 2110 ai.record.encoding = sc->sc_sound_rparams.encoding; 2111 ai.record.channels = sc->sc_sound_rparams.channels; 2112 ai.record.precision = sc->sc_sound_rparams.precision; 2113 ai.record.pause = sc->sc_sound_rpause; 2114 } 2115 error = audio_file_setinfo(sc, af, &ai); 2116 if (error) 2117 goto bad2; 2118 2119 if (sc->sc_popens + sc->sc_ropens == 0) { 2120 /* First open */ 2121 2122 sc->sc_cred = kauth_cred_get(); 2123 kauth_cred_hold(sc->sc_cred); 2124 2125 if (sc->hw_if->open) { 2126 int hwflags; 2127 2128 /* 2129 * Call hw_if->open() only at first open of 2130 * combination of playback and recording. 2131 * On full duplex hardware, the flags passed to 2132 * hw_if->open() is always (FREAD | FWRITE) 2133 * regardless of this open()'s flags. 2134 * see also dev/isa/aria.c 2135 * On half duplex hardware, the flags passed to 2136 * hw_if->open() is either FREAD or FWRITE. 2137 * see also arch/evbarm/mini2440/audio_mini2440.c 2138 */ 2139 if (fullduplex) { 2140 hwflags = FREAD | FWRITE; 2141 } else { 2142 /* Construct hwflags from af->mode. */ 2143 hwflags = 0; 2144 if ((af->mode & AUMODE_PLAY) != 0) 2145 hwflags |= FWRITE; 2146 if ((af->mode & AUMODE_RECORD) != 0) 2147 hwflags |= FREAD; 2148 } 2149 2150 mutex_enter(sc->sc_intr_lock); 2151 error = sc->hw_if->open(sc->hw_hdl, hwflags); 2152 mutex_exit(sc->sc_intr_lock); 2153 if (error) 2154 goto bad2; 2155 } 2156 2157 /* 2158 * Set speaker mode when a half duplex. 2159 * XXX I'm not sure this is correct. 2160 */ 2161 if (1/*XXX*/) { 2162 if (sc->hw_if->speaker_ctl) { 2163 int on; 2164 if (af->ptrack) { 2165 on = 1; 2166 } else { 2167 on = 0; 2168 } 2169 mutex_enter(sc->sc_intr_lock); 2170 error = sc->hw_if->speaker_ctl(sc->hw_hdl, on); 2171 mutex_exit(sc->sc_intr_lock); 2172 if (error) 2173 goto bad3; 2174 } 2175 } 2176 } else if (sc->sc_multiuser == false) { 2177 uid_t euid = kauth_cred_geteuid(kauth_cred_get()); 2178 if (euid != 0 && euid != kauth_cred_geteuid(sc->sc_cred)) { 2179 error = EPERM; 2180 goto bad2; 2181 } 2182 } 2183 2184 /* Call init_output if this is the first playback open. */ 2185 if (af->ptrack && sc->sc_popens == 0) { 2186 if (sc->hw_if->init_output) { 2187 hwbuf = &sc->sc_pmixer->hwbuf; 2188 mutex_enter(sc->sc_intr_lock); 2189 error = sc->hw_if->init_output(sc->hw_hdl, 2190 hwbuf->mem, 2191 hwbuf->capacity * 2192 hwbuf->fmt.channels * hwbuf->fmt.stride / NBBY); 2193 mutex_exit(sc->sc_intr_lock); 2194 if (error) 2195 goto bad3; 2196 } 2197 } 2198 /* Call init_input if this is the first recording open. */ 2199 if (af->rtrack && sc->sc_ropens == 0) { 2200 if (sc->hw_if->init_input) { 2201 hwbuf = &sc->sc_rmixer->hwbuf; 2202 mutex_enter(sc->sc_intr_lock); 2203 error = sc->hw_if->init_input(sc->hw_hdl, 2204 hwbuf->mem, 2205 hwbuf->capacity * 2206 hwbuf->fmt.channels * hwbuf->fmt.stride / NBBY); 2207 mutex_exit(sc->sc_intr_lock); 2208 if (error) 2209 goto bad3; 2210 } 2211 } 2212 2213 if (bellfile == NULL) { 2214 error = fd_allocfile(&fp, &fd); 2215 if (error) 2216 goto bad3; 2217 } 2218 2219 /* 2220 * Count up finally. 2221 * Don't fail from here. 2222 */ 2223 if (af->ptrack) 2224 sc->sc_popens++; 2225 if (af->rtrack) 2226 sc->sc_ropens++; 2227 mutex_enter(sc->sc_intr_lock); 2228 SLIST_INSERT_HEAD(&sc->sc_files, af, entry); 2229 mutex_exit(sc->sc_intr_lock); 2230 2231 if (bellfile) { 2232 *bellfile = af; 2233 } else { 2234 error = fd_clone(fp, fd, flags, &audio_fileops, af); 2235 KASSERTMSG(error == EMOVEFD, "error=%d", error); 2236 } 2237 2238 TRACEF(3, af, "done"); 2239 return error; 2240 2241 /* 2242 * Since track here is not yet linked to sc_files, 2243 * you can call track_destroy() without sc_intr_lock. 2244 */ 2245 bad3: 2246 if (sc->sc_popens + sc->sc_ropens == 0) { 2247 if (sc->hw_if->close) { 2248 mutex_enter(sc->sc_intr_lock); 2249 sc->hw_if->close(sc->hw_hdl); 2250 mutex_exit(sc->sc_intr_lock); 2251 } 2252 } 2253 bad2: 2254 if (af->rtrack) { 2255 audio_track_destroy(af->rtrack); 2256 af->rtrack = NULL; 2257 } 2258 if (af->ptrack) { 2259 audio_track_destroy(af->ptrack); 2260 af->ptrack = NULL; 2261 } 2262 bad1: 2263 kmem_free(af, sizeof(*af)); 2264 return error; 2265 } 2266 2267 /* 2268 * Must be called without sc_lock nor sc_exlock held. 2269 */ 2270 int 2271 audio_close(struct audio_softc *sc, audio_file_t *file) 2272 { 2273 2274 /* Protect entering new fileops to this file */ 2275 atomic_store_relaxed(&file->dying, true); 2276 2277 /* 2278 * Drain first. 2279 * It must be done before unlinking(acquiring exclusive lock). 2280 */ 2281 if (file->ptrack) { 2282 mutex_enter(sc->sc_lock); 2283 audio_track_drain(sc, file->ptrack); 2284 mutex_exit(sc->sc_lock); 2285 } 2286 2287 return audio_unlink(sc, file); 2288 } 2289 2290 /* 2291 * Unlink this file, but not freeing memory here. 2292 * Must be called without sc_lock nor sc_exlock held. 2293 */ 2294 int 2295 audio_unlink(struct audio_softc *sc, audio_file_t *file) 2296 { 2297 int error; 2298 2299 TRACEF(1, file, "%spid=%d.%d po=%d ro=%d", 2300 (audiodebug >= 3) ? "start " : "", 2301 (int)curproc->p_pid, (int)curlwp->l_lid, 2302 sc->sc_popens, sc->sc_ropens); 2303 KASSERTMSG(sc->sc_popens + sc->sc_ropens > 0, 2304 "sc->sc_popens=%d, sc->sc_ropens=%d", 2305 sc->sc_popens, sc->sc_ropens); 2306 2307 mutex_enter(sc->sc_lock); 2308 /* 2309 * Acquire exclusive lock to protect counters. 2310 * Does not use audio_enter_exclusive() due to sc_dying. 2311 */ 2312 while (__predict_false(sc->sc_exlock != 0)) { 2313 error = cv_timedwait_sig(&sc->sc_exlockcv, sc->sc_lock, 2314 mstohz(AUDIO_TIMEOUT)); 2315 /* XXX what should I do on error? */ 2316 if (error == EWOULDBLOCK) { 2317 mutex_exit(sc->sc_lock); 2318 device_printf(sc->sc_dev, 2319 "%s: cv_timedwait_sig failed %d", __func__, error); 2320 return error; 2321 } 2322 } 2323 sc->sc_exlock = 1; 2324 2325 device_active(sc->sc_dev, DVA_SYSTEM); 2326 2327 mutex_enter(sc->sc_intr_lock); 2328 SLIST_REMOVE(&sc->sc_files, file, audio_file, entry); 2329 mutex_exit(sc->sc_intr_lock); 2330 2331 if (file->ptrack) { 2332 TRACET(3, file->ptrack, "dropframes=%" PRIu64, 2333 file->ptrack->dropframes); 2334 2335 KASSERT(sc->sc_popens > 0); 2336 sc->sc_popens--; 2337 2338 /* Call hw halt_output if this is the last playback track. */ 2339 if (sc->sc_popens == 0 && sc->sc_pbusy) { 2340 error = audio_pmixer_halt(sc); 2341 if (error) { 2342 device_printf(sc->sc_dev, 2343 "halt_output failed with %d (ignored)\n", 2344 error); 2345 } 2346 } 2347 2348 /* Restore mixing volume if all tracks are gone. */ 2349 if (sc->sc_popens == 0) { 2350 /* intr_lock is not necessary, but just manners. */ 2351 mutex_enter(sc->sc_intr_lock); 2352 sc->sc_pmixer->volume = 256; 2353 sc->sc_pmixer->voltimer = 0; 2354 mutex_exit(sc->sc_intr_lock); 2355 } 2356 } 2357 if (file->rtrack) { 2358 TRACET(3, file->rtrack, "dropframes=%" PRIu64, 2359 file->rtrack->dropframes); 2360 2361 KASSERT(sc->sc_ropens > 0); 2362 sc->sc_ropens--; 2363 2364 /* Call hw halt_input if this is the last recording track. */ 2365 if (sc->sc_ropens == 0 && sc->sc_rbusy) { 2366 error = audio_rmixer_halt(sc); 2367 if (error) { 2368 device_printf(sc->sc_dev, 2369 "halt_input failed with %d (ignored)\n", 2370 error); 2371 } 2372 } 2373 2374 } 2375 2376 /* Call hw close if this is the last track. */ 2377 if (sc->sc_popens + sc->sc_ropens == 0) { 2378 if (sc->hw_if->close) { 2379 TRACE(2, "hw_if close"); 2380 mutex_enter(sc->sc_intr_lock); 2381 sc->hw_if->close(sc->hw_hdl); 2382 mutex_exit(sc->sc_intr_lock); 2383 } 2384 2385 kauth_cred_free(sc->sc_cred); 2386 } 2387 2388 TRACE(3, "done"); 2389 audio_exit_exclusive(sc); 2390 2391 return 0; 2392 } 2393 2394 /* 2395 * Must be called without sc_lock nor sc_exlock held. 2396 */ 2397 int 2398 audio_read(struct audio_softc *sc, struct uio *uio, int ioflag, 2399 audio_file_t *file) 2400 { 2401 audio_track_t *track; 2402 audio_ring_t *usrbuf; 2403 audio_ring_t *input; 2404 int error; 2405 2406 /* 2407 * On half-duplex hardware, O_RDWR is treated as O_WRONLY. 2408 * However read() system call itself can be called because it's 2409 * opened with O_RDWR. So in this case, deny this read(). 2410 */ 2411 track = file->rtrack; 2412 if (track == NULL) { 2413 return EBADF; 2414 } 2415 2416 /* I think it's better than EINVAL. */ 2417 if (track->mmapped) 2418 return EPERM; 2419 2420 TRACET(2, track, "resid=%zd", uio->uio_resid); 2421 2422 #ifdef AUDIO_PM_IDLE 2423 mutex_enter(sc->sc_lock); 2424 if (device_is_active(&sc->sc_dev) || sc->sc_idle) 2425 device_active(&sc->sc_dev, DVA_SYSTEM); 2426 mutex_exit(sc->sc_lock); 2427 #endif 2428 2429 usrbuf = &track->usrbuf; 2430 input = track->input; 2431 2432 /* 2433 * The first read starts rmixer. 2434 */ 2435 error = audio_enter_exclusive(sc); 2436 if (error) 2437 return error; 2438 if (sc->sc_rbusy == false) 2439 audio_rmixer_start(sc); 2440 audio_exit_exclusive(sc); 2441 2442 error = 0; 2443 while (uio->uio_resid > 0 && error == 0) { 2444 int bytes; 2445 2446 TRACET(3, track, 2447 "while resid=%zd input=%d/%d/%d usrbuf=%d/%d/H%d", 2448 uio->uio_resid, 2449 input->head, input->used, input->capacity, 2450 usrbuf->head, usrbuf->used, track->usrbuf_usedhigh); 2451 2452 /* Wait when buffers are empty. */ 2453 mutex_enter(sc->sc_lock); 2454 for (;;) { 2455 bool empty; 2456 audio_track_lock_enter(track); 2457 empty = (input->used == 0 && usrbuf->used == 0); 2458 audio_track_lock_exit(track); 2459 if (!empty) 2460 break; 2461 2462 if ((ioflag & IO_NDELAY)) { 2463 mutex_exit(sc->sc_lock); 2464 return EWOULDBLOCK; 2465 } 2466 2467 TRACET(3, track, "sleep"); 2468 error = audio_track_waitio(sc, track); 2469 if (error) { 2470 mutex_exit(sc->sc_lock); 2471 return error; 2472 } 2473 } 2474 mutex_exit(sc->sc_lock); 2475 2476 audio_track_lock_enter(track); 2477 audio_track_record(track); 2478 2479 /* uiomove from usrbuf as much as possible. */ 2480 bytes = uimin(usrbuf->used, uio->uio_resid); 2481 while (bytes > 0) { 2482 int head = usrbuf->head; 2483 int len = uimin(bytes, usrbuf->capacity - head); 2484 error = uiomove((uint8_t *)usrbuf->mem + head, len, 2485 uio); 2486 if (error) { 2487 audio_track_lock_exit(track); 2488 device_printf(sc->sc_dev, 2489 "uiomove(len=%d) failed with %d\n", 2490 len, error); 2491 goto abort; 2492 } 2493 auring_take(usrbuf, len); 2494 track->useriobytes += len; 2495 TRACET(3, track, "uiomove(len=%d) usrbuf=%d/%d/C%d", 2496 len, 2497 usrbuf->head, usrbuf->used, usrbuf->capacity); 2498 bytes -= len; 2499 } 2500 2501 audio_track_lock_exit(track); 2502 } 2503 2504 abort: 2505 return error; 2506 } 2507 2508 2509 /* 2510 * Clear file's playback and/or record track buffer immediately. 2511 */ 2512 static void 2513 audio_file_clear(struct audio_softc *sc, audio_file_t *file) 2514 { 2515 2516 if (file->ptrack) 2517 audio_track_clear(sc, file->ptrack); 2518 if (file->rtrack) 2519 audio_track_clear(sc, file->rtrack); 2520 } 2521 2522 /* 2523 * Must be called without sc_lock nor sc_exlock held. 2524 */ 2525 int 2526 audio_write(struct audio_softc *sc, struct uio *uio, int ioflag, 2527 audio_file_t *file) 2528 { 2529 audio_track_t *track; 2530 audio_ring_t *usrbuf; 2531 audio_ring_t *outbuf; 2532 int error; 2533 2534 track = file->ptrack; 2535 KASSERT(track); 2536 2537 /* I think it's better than EINVAL. */ 2538 if (track->mmapped) 2539 return EPERM; 2540 2541 TRACET(2, track, "%sresid=%zd pid=%d.%d ioflag=0x%x", 2542 audiodebug >= 3 ? "begin " : "", 2543 uio->uio_resid, (int)curproc->p_pid, (int)curlwp->l_lid, ioflag); 2544 2545 if (uio->uio_resid == 0) { 2546 track->eofcounter++; 2547 return 0; 2548 } 2549 2550 #ifdef AUDIO_PM_IDLE 2551 mutex_enter(sc->sc_lock); 2552 if (device_is_active(&sc->sc_dev) || sc->sc_idle) 2553 device_active(&sc->sc_dev, DVA_SYSTEM); 2554 mutex_exit(sc->sc_lock); 2555 #endif 2556 2557 usrbuf = &track->usrbuf; 2558 outbuf = &track->outbuf; 2559 2560 /* 2561 * The first write starts pmixer. 2562 */ 2563 error = audio_enter_exclusive(sc); 2564 if (error) 2565 return error; 2566 if (sc->sc_pbusy == false) 2567 audio_pmixer_start(sc, false); 2568 audio_exit_exclusive(sc); 2569 2570 track->pstate = AUDIO_STATE_RUNNING; 2571 error = 0; 2572 while (uio->uio_resid > 0 && error == 0) { 2573 int bytes; 2574 2575 TRACET(3, track, "while resid=%zd usrbuf=%d/%d/H%d", 2576 uio->uio_resid, 2577 usrbuf->head, usrbuf->used, track->usrbuf_usedhigh); 2578 2579 /* Wait when buffers are full. */ 2580 mutex_enter(sc->sc_lock); 2581 for (;;) { 2582 bool full; 2583 audio_track_lock_enter(track); 2584 full = (usrbuf->used >= track->usrbuf_usedhigh && 2585 outbuf->used >= outbuf->capacity); 2586 audio_track_lock_exit(track); 2587 if (!full) 2588 break; 2589 2590 if ((ioflag & IO_NDELAY)) { 2591 error = EWOULDBLOCK; 2592 mutex_exit(sc->sc_lock); 2593 goto abort; 2594 } 2595 2596 TRACET(3, track, "sleep usrbuf=%d/H%d", 2597 usrbuf->used, track->usrbuf_usedhigh); 2598 error = audio_track_waitio(sc, track); 2599 if (error) { 2600 mutex_exit(sc->sc_lock); 2601 goto abort; 2602 } 2603 } 2604 mutex_exit(sc->sc_lock); 2605 2606 audio_track_lock_enter(track); 2607 2608 /* uiomove to usrbuf as much as possible. */ 2609 bytes = uimin(track->usrbuf_usedhigh - usrbuf->used, 2610 uio->uio_resid); 2611 while (bytes > 0) { 2612 int tail = auring_tail(usrbuf); 2613 int len = uimin(bytes, usrbuf->capacity - tail); 2614 error = uiomove((uint8_t *)usrbuf->mem + tail, len, 2615 uio); 2616 if (error) { 2617 audio_track_lock_exit(track); 2618 device_printf(sc->sc_dev, 2619 "uiomove(len=%d) failed with %d\n", 2620 len, error); 2621 goto abort; 2622 } 2623 auring_push(usrbuf, len); 2624 track->useriobytes += len; 2625 TRACET(3, track, "uiomove(len=%d) usrbuf=%d/%d/C%d", 2626 len, 2627 usrbuf->head, usrbuf->used, usrbuf->capacity); 2628 bytes -= len; 2629 } 2630 2631 /* Convert them as much as possible. */ 2632 while (usrbuf->used >= track->usrbuf_blksize && 2633 outbuf->used < outbuf->capacity) { 2634 audio_track_play(track); 2635 } 2636 2637 audio_track_lock_exit(track); 2638 } 2639 2640 abort: 2641 TRACET(3, track, "done error=%d", error); 2642 return error; 2643 } 2644 2645 /* 2646 * Must be called without sc_lock nor sc_exlock held. 2647 */ 2648 int 2649 audio_ioctl(dev_t dev, struct audio_softc *sc, u_long cmd, void *addr, int flag, 2650 struct lwp *l, audio_file_t *file) 2651 { 2652 struct audio_offset *ao; 2653 struct audio_info ai; 2654 audio_track_t *track; 2655 audio_encoding_t *ae; 2656 audio_format_query_t *query; 2657 u_int stamp; 2658 u_int offs; 2659 int fd; 2660 int index; 2661 int error; 2662 2663 #if defined(AUDIO_DEBUG) 2664 const char *ioctlnames[] = { 2665 " AUDIO_GETINFO", /* 21 */ 2666 " AUDIO_SETINFO", /* 22 */ 2667 " AUDIO_DRAIN", /* 23 */ 2668 " AUDIO_FLUSH", /* 24 */ 2669 " AUDIO_WSEEK", /* 25 */ 2670 " AUDIO_RERROR", /* 26 */ 2671 " AUDIO_GETDEV", /* 27 */ 2672 " AUDIO_GETENC", /* 28 */ 2673 " AUDIO_GETFD", /* 29 */ 2674 " AUDIO_SETFD", /* 30 */ 2675 " AUDIO_PERROR", /* 31 */ 2676 " AUDIO_GETIOFFS", /* 32 */ 2677 " AUDIO_GETOOFFS", /* 33 */ 2678 " AUDIO_GETPROPS", /* 34 */ 2679 " AUDIO_GETBUFINFO", /* 35 */ 2680 " AUDIO_SETCHAN", /* 36 */ 2681 " AUDIO_GETCHAN", /* 37 */ 2682 " AUDIO_QUERYFORMAT", /* 38 */ 2683 " AUDIO_GETFORMAT", /* 39 */ 2684 " AUDIO_SETFORMAT", /* 40 */ 2685 }; 2686 int nameidx = (cmd & 0xff); 2687 const char *ioctlname = ""; 2688 if (21 <= nameidx && nameidx <= 21 + __arraycount(ioctlnames)) 2689 ioctlname = ioctlnames[nameidx - 21]; 2690 TRACEF(2, file, "(%lu,'%c',%lu)%s pid=%d.%d", 2691 IOCPARM_LEN(cmd), (char)IOCGROUP(cmd), cmd&0xff, ioctlname, 2692 (int)curproc->p_pid, (int)l->l_lid); 2693 #endif 2694 2695 error = 0; 2696 switch (cmd) { 2697 case FIONBIO: 2698 /* All handled in the upper FS layer. */ 2699 break; 2700 2701 case FIONREAD: 2702 /* Get the number of bytes that can be read. */ 2703 if (file->rtrack) { 2704 *(int *)addr = audio_track_readablebytes(file->rtrack); 2705 } else { 2706 *(int *)addr = 0; 2707 } 2708 break; 2709 2710 case FIOASYNC: 2711 /* Set/Clear ASYNC I/O. */ 2712 if (*(int *)addr) { 2713 file->async_audio = curproc->p_pid; 2714 TRACEF(2, file, "FIOASYNC pid %d", file->async_audio); 2715 } else { 2716 file->async_audio = 0; 2717 TRACEF(2, file, "FIOASYNC off"); 2718 } 2719 break; 2720 2721 case AUDIO_FLUSH: 2722 /* XXX TODO: clear errors and restart? */ 2723 audio_file_clear(sc, file); 2724 break; 2725 2726 case AUDIO_RERROR: 2727 /* 2728 * Number of read bytes dropped. We don't know where 2729 * or when they were dropped (including conversion stage). 2730 * Therefore, the number of accurate bytes or samples is 2731 * also unknown. 2732 */ 2733 track = file->rtrack; 2734 if (track) { 2735 *(int *)addr = frametobyte(&track->usrbuf.fmt, 2736 track->dropframes); 2737 } 2738 break; 2739 2740 case AUDIO_PERROR: 2741 /* 2742 * Number of write bytes dropped. We don't know where 2743 * or when they were dropped (including conversion stage). 2744 * Therefore, the number of accurate bytes or samples is 2745 * also unknown. 2746 */ 2747 track = file->ptrack; 2748 if (track) { 2749 *(int *)addr = frametobyte(&track->usrbuf.fmt, 2750 track->dropframes); 2751 } 2752 break; 2753 2754 case AUDIO_GETIOFFS: 2755 /* XXX TODO */ 2756 ao = (struct audio_offset *)addr; 2757 ao->samples = 0; 2758 ao->deltablks = 0; 2759 ao->offset = 0; 2760 break; 2761 2762 case AUDIO_GETOOFFS: 2763 ao = (struct audio_offset *)addr; 2764 track = file->ptrack; 2765 if (track == NULL) { 2766 ao->samples = 0; 2767 ao->deltablks = 0; 2768 ao->offset = 0; 2769 break; 2770 } 2771 mutex_enter(sc->sc_lock); 2772 mutex_enter(sc->sc_intr_lock); 2773 /* figure out where next DMA will start */ 2774 stamp = track->usrbuf_stamp; 2775 offs = track->usrbuf.head; 2776 mutex_exit(sc->sc_intr_lock); 2777 mutex_exit(sc->sc_lock); 2778 2779 ao->samples = stamp; 2780 ao->deltablks = (stamp / track->usrbuf_blksize) - 2781 (track->usrbuf_stamp_last / track->usrbuf_blksize); 2782 track->usrbuf_stamp_last = stamp; 2783 offs = rounddown(offs, track->usrbuf_blksize) 2784 + track->usrbuf_blksize; 2785 if (offs >= track->usrbuf.capacity) 2786 offs -= track->usrbuf.capacity; 2787 ao->offset = offs; 2788 2789 TRACET(3, track, "GETOOFFS: samples=%u deltablks=%u offset=%u", 2790 ao->samples, ao->deltablks, ao->offset); 2791 break; 2792 2793 case AUDIO_WSEEK: 2794 /* XXX return value does not include outbuf one. */ 2795 if (file->ptrack) 2796 *(u_long *)addr = file->ptrack->usrbuf.used; 2797 break; 2798 2799 case AUDIO_SETINFO: 2800 error = audio_enter_exclusive(sc); 2801 if (error) 2802 break; 2803 error = audio_file_setinfo(sc, file, (struct audio_info *)addr); 2804 if (error) { 2805 audio_exit_exclusive(sc); 2806 break; 2807 } 2808 /* XXX TODO: update last_ai if /dev/sound ? */ 2809 if (ISDEVSOUND(dev)) 2810 error = audiogetinfo(sc, &sc->sc_ai, 0, file); 2811 audio_exit_exclusive(sc); 2812 break; 2813 2814 case AUDIO_GETINFO: 2815 error = audio_enter_exclusive(sc); 2816 if (error) 2817 break; 2818 error = audiogetinfo(sc, (struct audio_info *)addr, 1, file); 2819 audio_exit_exclusive(sc); 2820 break; 2821 2822 case AUDIO_GETBUFINFO: 2823 mutex_enter(sc->sc_lock); 2824 error = audiogetinfo(sc, (struct audio_info *)addr, 0, file); 2825 mutex_exit(sc->sc_lock); 2826 break; 2827 2828 case AUDIO_DRAIN: 2829 if (file->ptrack) { 2830 mutex_enter(sc->sc_lock); 2831 error = audio_track_drain(sc, file->ptrack); 2832 mutex_exit(sc->sc_lock); 2833 } 2834 break; 2835 2836 case AUDIO_GETDEV: 2837 mutex_enter(sc->sc_lock); 2838 error = sc->hw_if->getdev(sc->hw_hdl, (audio_device_t *)addr); 2839 mutex_exit(sc->sc_lock); 2840 break; 2841 2842 case AUDIO_GETENC: 2843 ae = (audio_encoding_t *)addr; 2844 index = ae->index; 2845 if (index < 0 || index >= __arraycount(audio_encodings)) { 2846 error = EINVAL; 2847 break; 2848 } 2849 *ae = audio_encodings[index]; 2850 ae->index = index; 2851 /* 2852 * EMULATED always. 2853 * EMULATED flag at that time used to mean that it could 2854 * not be passed directly to the hardware as-is. But 2855 * currently, all formats including hardware native is not 2856 * passed directly to the hardware. So I set EMULATED 2857 * flag for all formats. 2858 */ 2859 ae->flags = AUDIO_ENCODINGFLAG_EMULATED; 2860 break; 2861 2862 case AUDIO_GETFD: 2863 /* 2864 * Returns the current setting of full duplex mode. 2865 * If HW has full duplex mode and there are two mixers, 2866 * it is full duplex. Otherwise half duplex. 2867 */ 2868 mutex_enter(sc->sc_lock); 2869 fd = (sc->sc_props & AUDIO_PROP_FULLDUPLEX) 2870 && (sc->sc_pmixer && sc->sc_rmixer); 2871 mutex_exit(sc->sc_lock); 2872 *(int *)addr = fd; 2873 break; 2874 2875 case AUDIO_GETPROPS: 2876 *(int *)addr = sc->sc_props; 2877 break; 2878 2879 case AUDIO_QUERYFORMAT: 2880 query = (audio_format_query_t *)addr; 2881 mutex_enter(sc->sc_lock); 2882 error = sc->hw_if->query_format(sc->hw_hdl, query); 2883 mutex_exit(sc->sc_lock); 2884 /* Hide internal infomations */ 2885 query->fmt.driver_data = NULL; 2886 break; 2887 2888 case AUDIO_GETFORMAT: 2889 audio_mixers_get_format(sc, (struct audio_info *)addr); 2890 break; 2891 2892 case AUDIO_SETFORMAT: 2893 mutex_enter(sc->sc_lock); 2894 audio_mixers_get_format(sc, &ai); 2895 error = audio_mixers_set_format(sc, (struct audio_info *)addr); 2896 if (error) { 2897 /* Rollback */ 2898 audio_mixers_set_format(sc, &ai); 2899 } 2900 mutex_exit(sc->sc_lock); 2901 break; 2902 2903 case AUDIO_SETFD: 2904 case AUDIO_SETCHAN: 2905 case AUDIO_GETCHAN: 2906 /* Obsoleted */ 2907 break; 2908 2909 default: 2910 if (sc->hw_if->dev_ioctl) { 2911 error = audio_enter_exclusive(sc); 2912 if (error) 2913 break; 2914 error = sc->hw_if->dev_ioctl(sc->hw_hdl, 2915 cmd, addr, flag, l); 2916 audio_exit_exclusive(sc); 2917 } else { 2918 TRACEF(2, file, "unknown ioctl"); 2919 error = EINVAL; 2920 } 2921 break; 2922 } 2923 TRACEF(2, file, "(%lu,'%c',%lu)%s result %d", 2924 IOCPARM_LEN(cmd), (char)IOCGROUP(cmd), cmd&0xff, ioctlname, 2925 error); 2926 return error; 2927 } 2928 2929 /* 2930 * Returns the number of bytes that can be read on recording buffer. 2931 */ 2932 static __inline int 2933 audio_track_readablebytes(const audio_track_t *track) 2934 { 2935 int bytes; 2936 2937 KASSERT(track); 2938 KASSERT(track->mode == AUMODE_RECORD); 2939 2940 /* 2941 * Although usrbuf is primarily readable data, recorded data 2942 * also stays in track->input until reading. So it is necessary 2943 * to add it. track->input is in frame, usrbuf is in byte. 2944 */ 2945 bytes = track->usrbuf.used + 2946 track->input->used * frametobyte(&track->usrbuf.fmt, 1); 2947 return bytes; 2948 } 2949 2950 /* 2951 * Must be called without sc_lock nor sc_exlock held. 2952 */ 2953 int 2954 audio_poll(struct audio_softc *sc, int events, struct lwp *l, 2955 audio_file_t *file) 2956 { 2957 audio_track_t *track; 2958 int revents; 2959 bool in_is_valid; 2960 bool out_is_valid; 2961 2962 #if defined(AUDIO_DEBUG) 2963 #define POLLEV_BITMAP "\177\020" \ 2964 "b\10WRBAND\0" \ 2965 "b\7RDBAND\0" "b\6RDNORM\0" "b\5NVAL\0" "b\4HUP\0" \ 2966 "b\3ERR\0" "b\2OUT\0" "b\1PRI\0" "b\0IN\0" 2967 char evbuf[64]; 2968 snprintb(evbuf, sizeof(evbuf), POLLEV_BITMAP, events); 2969 TRACEF(2, file, "pid=%d.%d events=%s", 2970 (int)curproc->p_pid, (int)l->l_lid, evbuf); 2971 #endif 2972 2973 revents = 0; 2974 in_is_valid = false; 2975 out_is_valid = false; 2976 if (events & (POLLIN | POLLRDNORM)) { 2977 track = file->rtrack; 2978 if (track) { 2979 int used; 2980 in_is_valid = true; 2981 used = audio_track_readablebytes(track); 2982 if (used > 0) 2983 revents |= events & (POLLIN | POLLRDNORM); 2984 } 2985 } 2986 if (events & (POLLOUT | POLLWRNORM)) { 2987 track = file->ptrack; 2988 if (track) { 2989 out_is_valid = true; 2990 if (track->usrbuf.used <= track->usrbuf_usedlow) 2991 revents |= events & (POLLOUT | POLLWRNORM); 2992 } 2993 } 2994 2995 if (revents == 0) { 2996 mutex_enter(sc->sc_lock); 2997 if (in_is_valid) { 2998 TRACEF(3, file, "selrecord rsel"); 2999 selrecord(l, &sc->sc_rsel); 3000 } 3001 if (out_is_valid) { 3002 TRACEF(3, file, "selrecord wsel"); 3003 selrecord(l, &sc->sc_wsel); 3004 } 3005 mutex_exit(sc->sc_lock); 3006 } 3007 3008 #if defined(AUDIO_DEBUG) 3009 snprintb(evbuf, sizeof(evbuf), POLLEV_BITMAP, revents); 3010 TRACEF(2, file, "revents=%s", evbuf); 3011 #endif 3012 return revents; 3013 } 3014 3015 static const struct filterops audioread_filtops = { 3016 .f_isfd = 1, 3017 .f_attach = NULL, 3018 .f_detach = filt_audioread_detach, 3019 .f_event = filt_audioread_event, 3020 }; 3021 3022 static void 3023 filt_audioread_detach(struct knote *kn) 3024 { 3025 struct audio_softc *sc; 3026 audio_file_t *file; 3027 3028 file = kn->kn_hook; 3029 sc = file->sc; 3030 TRACEF(3, file, ""); 3031 3032 mutex_enter(sc->sc_lock); 3033 SLIST_REMOVE(&sc->sc_rsel.sel_klist, kn, knote, kn_selnext); 3034 mutex_exit(sc->sc_lock); 3035 } 3036 3037 static int 3038 filt_audioread_event(struct knote *kn, long hint) 3039 { 3040 audio_file_t *file; 3041 audio_track_t *track; 3042 3043 file = kn->kn_hook; 3044 track = file->rtrack; 3045 3046 /* 3047 * kn_data must contain the number of bytes can be read. 3048 * The return value indicates whether the event occurs or not. 3049 */ 3050 3051 if (track == NULL) { 3052 /* can not read with this descriptor. */ 3053 kn->kn_data = 0; 3054 return 0; 3055 } 3056 3057 kn->kn_data = audio_track_readablebytes(track); 3058 TRACEF(3, file, "data=%" PRId64, kn->kn_data); 3059 return kn->kn_data > 0; 3060 } 3061 3062 static const struct filterops audiowrite_filtops = { 3063 .f_isfd = 1, 3064 .f_attach = NULL, 3065 .f_detach = filt_audiowrite_detach, 3066 .f_event = filt_audiowrite_event, 3067 }; 3068 3069 static void 3070 filt_audiowrite_detach(struct knote *kn) 3071 { 3072 struct audio_softc *sc; 3073 audio_file_t *file; 3074 3075 file = kn->kn_hook; 3076 sc = file->sc; 3077 TRACEF(3, file, ""); 3078 3079 mutex_enter(sc->sc_lock); 3080 SLIST_REMOVE(&sc->sc_wsel.sel_klist, kn, knote, kn_selnext); 3081 mutex_exit(sc->sc_lock); 3082 } 3083 3084 static int 3085 filt_audiowrite_event(struct knote *kn, long hint) 3086 { 3087 audio_file_t *file; 3088 audio_track_t *track; 3089 3090 file = kn->kn_hook; 3091 track = file->ptrack; 3092 3093 /* 3094 * kn_data must contain the number of bytes can be write. 3095 * The return value indicates whether the event occurs or not. 3096 */ 3097 3098 if (track == NULL) { 3099 /* can not write with this descriptor. */ 3100 kn->kn_data = 0; 3101 return 0; 3102 } 3103 3104 kn->kn_data = track->usrbuf_usedhigh - track->usrbuf.used; 3105 TRACEF(3, file, "data=%" PRId64, kn->kn_data); 3106 return (track->usrbuf.used < track->usrbuf_usedlow); 3107 } 3108 3109 /* 3110 * Must be called without sc_lock nor sc_exlock held. 3111 */ 3112 int 3113 audio_kqfilter(struct audio_softc *sc, audio_file_t *file, struct knote *kn) 3114 { 3115 struct klist *klist; 3116 3117 TRACEF(3, file, "kn=%p kn_filter=%x", kn, (int)kn->kn_filter); 3118 3119 switch (kn->kn_filter) { 3120 case EVFILT_READ: 3121 klist = &sc->sc_rsel.sel_klist; 3122 kn->kn_fop = &audioread_filtops; 3123 break; 3124 3125 case EVFILT_WRITE: 3126 klist = &sc->sc_wsel.sel_klist; 3127 kn->kn_fop = &audiowrite_filtops; 3128 break; 3129 3130 default: 3131 return EINVAL; 3132 } 3133 3134 kn->kn_hook = file; 3135 3136 mutex_enter(sc->sc_lock); 3137 SLIST_INSERT_HEAD(klist, kn, kn_selnext); 3138 mutex_exit(sc->sc_lock); 3139 3140 return 0; 3141 } 3142 3143 /* 3144 * Must be called without sc_lock nor sc_exlock held. 3145 */ 3146 int 3147 audio_mmap(struct audio_softc *sc, off_t *offp, size_t len, int prot, 3148 int *flagsp, int *advicep, struct uvm_object **uobjp, int *maxprotp, 3149 audio_file_t *file) 3150 { 3151 audio_track_t *track; 3152 vsize_t vsize; 3153 int error; 3154 3155 TRACEF(2, file, "off=%lld, prot=%d", (long long)(*offp), prot); 3156 3157 if (*offp < 0) 3158 return EINVAL; 3159 3160 #if 0 3161 /* XXX 3162 * The idea here was to use the protection to determine if 3163 * we are mapping the read or write buffer, but it fails. 3164 * The VM system is broken in (at least) two ways. 3165 * 1) If you map memory VM_PROT_WRITE you SIGSEGV 3166 * when writing to it, so VM_PROT_READ|VM_PROT_WRITE 3167 * has to be used for mmapping the play buffer. 3168 * 2) Even if calling mmap() with VM_PROT_READ|VM_PROT_WRITE 3169 * audio_mmap will get called at some point with VM_PROT_READ 3170 * only. 3171 * So, alas, we always map the play buffer for now. 3172 */ 3173 if (prot == (VM_PROT_READ|VM_PROT_WRITE) || 3174 prot == VM_PROT_WRITE) 3175 track = file->ptrack; 3176 else if (prot == VM_PROT_READ) 3177 track = file->rtrack; 3178 else 3179 return EINVAL; 3180 #else 3181 track = file->ptrack; 3182 #endif 3183 if (track == NULL) 3184 return EACCES; 3185 3186 vsize = roundup2(MAX(track->usrbuf.capacity, PAGE_SIZE), PAGE_SIZE); 3187 if (len > vsize) 3188 return EOVERFLOW; 3189 if (*offp > (uint)(vsize - len)) 3190 return EOVERFLOW; 3191 3192 /* XXX TODO: what happens when mmap twice. */ 3193 if (!track->mmapped) { 3194 track->mmapped = true; 3195 3196 if (!track->is_pause) { 3197 error = audio_enter_exclusive(sc); 3198 if (error) 3199 return error; 3200 if (sc->sc_pbusy == false) 3201 audio_pmixer_start(sc, true); 3202 audio_exit_exclusive(sc); 3203 } 3204 /* XXX mmapping record buffer is not supported */ 3205 } 3206 3207 /* get ringbuffer */ 3208 *uobjp = track->uobj; 3209 3210 /* Acquire a reference for the mmap. munmap will release. */ 3211 uao_reference(*uobjp); 3212 *maxprotp = prot; 3213 *advicep = UVM_ADV_RANDOM; 3214 *flagsp = MAP_SHARED; 3215 return 0; 3216 } 3217 3218 /* 3219 * /dev/audioctl has to be able to open at any time without interference 3220 * with any /dev/audio or /dev/sound. 3221 */ 3222 static int 3223 audioctl_open(dev_t dev, struct audio_softc *sc, int flags, int ifmt, 3224 struct lwp *l) 3225 { 3226 struct file *fp; 3227 audio_file_t *af; 3228 int fd; 3229 int error; 3230 3231 KASSERT(mutex_owned(sc->sc_lock)); 3232 KASSERT(sc->sc_exlock); 3233 3234 TRACE(1, ""); 3235 3236 error = fd_allocfile(&fp, &fd); 3237 if (error) 3238 return error; 3239 3240 af = kmem_zalloc(sizeof(audio_file_t), KM_SLEEP); 3241 af->sc = sc; 3242 af->dev = dev; 3243 3244 /* Not necessary to insert sc_files. */ 3245 3246 error = fd_clone(fp, fd, flags, &audio_fileops, af); 3247 KASSERTMSG(error == EMOVEFD, "error=%d", error); 3248 3249 return error; 3250 } 3251 3252 /* 3253 * Free 'mem' if available, and initialize the pointer. 3254 * For this reason, this is implemented as macro. 3255 */ 3256 #define audio_free(mem) do { \ 3257 if (mem != NULL) { \ 3258 kern_free(mem); \ 3259 mem = NULL; \ 3260 } \ 3261 } while (0) 3262 3263 /* 3264 * (Re)allocate 'memblock' with specified 'bytes'. 3265 * bytes must not be 0. 3266 * This function never returns NULL. 3267 */ 3268 static void * 3269 audio_realloc(void *memblock, size_t bytes) 3270 { 3271 3272 KASSERT(bytes != 0); 3273 audio_free(memblock); 3274 return kern_malloc(bytes, M_WAITOK); 3275 } 3276 3277 /* 3278 * (Re)allocate usrbuf with 'newbufsize' bytes. 3279 * Use this function for usrbuf because only usrbuf can be mmapped. 3280 * If successful, it updates track->usrbuf.mem, track->usrbuf.capacity and 3281 * returns 0. Otherwise, it clears track->usrbuf.mem, track->usrbuf.capacity 3282 * and returns errno. 3283 * It must be called before updating usrbuf.capacity. 3284 */ 3285 static int 3286 audio_realloc_usrbuf(audio_track_t *track, int newbufsize) 3287 { 3288 struct audio_softc *sc; 3289 vaddr_t vstart; 3290 vsize_t oldvsize; 3291 vsize_t newvsize; 3292 int error; 3293 3294 KASSERT(newbufsize > 0); 3295 sc = track->mixer->sc; 3296 3297 /* Get a nonzero multiple of PAGE_SIZE */ 3298 newvsize = roundup2(MAX(newbufsize, PAGE_SIZE), PAGE_SIZE); 3299 3300 if (track->usrbuf.mem != NULL) { 3301 oldvsize = roundup2(MAX(track->usrbuf.capacity, PAGE_SIZE), 3302 PAGE_SIZE); 3303 if (oldvsize == newvsize) { 3304 track->usrbuf.capacity = newbufsize; 3305 return 0; 3306 } 3307 vstart = (vaddr_t)track->usrbuf.mem; 3308 uvm_unmap(kernel_map, vstart, vstart + oldvsize); 3309 /* uvm_unmap also detach uobj */ 3310 track->uobj = NULL; /* paranoia */ 3311 track->usrbuf.mem = NULL; 3312 } 3313 3314 /* Create a uvm anonymous object */ 3315 track->uobj = uao_create(newvsize, 0); 3316 3317 /* Map it into the kernel virtual address space */ 3318 vstart = 0; 3319 error = uvm_map(kernel_map, &vstart, newvsize, track->uobj, 0, 0, 3320 UVM_MAPFLAG(UVM_PROT_RW, UVM_PROT_RW, UVM_INH_NONE, 3321 UVM_ADV_RANDOM, 0)); 3322 if (error) { 3323 device_printf(sc->sc_dev, "uvm_map failed with %d\n", error); 3324 uao_detach(track->uobj); /* release reference */ 3325 goto abort; 3326 } 3327 3328 error = uvm_map_pageable(kernel_map, vstart, vstart + newvsize, 3329 false, 0); 3330 if (error) { 3331 device_printf(sc->sc_dev, "uvm_map_pageable failed with %d\n", 3332 error); 3333 uvm_unmap(kernel_map, vstart, vstart + newvsize); 3334 /* uvm_unmap also detach uobj */ 3335 goto abort; 3336 } 3337 3338 track->usrbuf.mem = (void *)vstart; 3339 track->usrbuf.capacity = newbufsize; 3340 memset(track->usrbuf.mem, 0, newvsize); 3341 return 0; 3342 3343 /* failure */ 3344 abort: 3345 track->uobj = NULL; /* paranoia */ 3346 track->usrbuf.mem = NULL; 3347 track->usrbuf.capacity = 0; 3348 return error; 3349 } 3350 3351 /* 3352 * Free usrbuf (if available). 3353 */ 3354 static void 3355 audio_free_usrbuf(audio_track_t *track) 3356 { 3357 vaddr_t vstart; 3358 vsize_t vsize; 3359 3360 vstart = (vaddr_t)track->usrbuf.mem; 3361 vsize = roundup2(MAX(track->usrbuf.capacity, PAGE_SIZE), PAGE_SIZE); 3362 if (track->usrbuf.mem != NULL) { 3363 /* 3364 * Unmap the kernel mapping. uvm_unmap releases the 3365 * reference to the uvm object, and this should be the 3366 * last virtual mapping of the uvm object, so no need 3367 * to explicitly release (`detach') the object. 3368 */ 3369 uvm_unmap(kernel_map, vstart, vstart + vsize); 3370 3371 track->uobj = NULL; 3372 track->usrbuf.mem = NULL; 3373 track->usrbuf.capacity = 0; 3374 } 3375 } 3376 3377 /* 3378 * This filter changes the volume for each channel. 3379 * arg->context points track->ch_volume[]. 3380 */ 3381 static void 3382 audio_track_chvol(audio_filter_arg_t *arg) 3383 { 3384 int16_t *ch_volume; 3385 const aint_t *s; 3386 aint_t *d; 3387 u_int i; 3388 u_int ch; 3389 u_int channels; 3390 3391 DIAGNOSTIC_filter_arg(arg); 3392 KASSERTMSG(arg->srcfmt->channels == arg->dstfmt->channels, 3393 "arg->srcfmt->channels=%d, arg->dstfmt->channels=%d", 3394 arg->srcfmt->channels, arg->dstfmt->channels); 3395 KASSERT(arg->context != NULL); 3396 KASSERTMSG(arg->srcfmt->channels <= AUDIO_MAX_CHANNELS, 3397 "arg->srcfmt->channels=%d", arg->srcfmt->channels); 3398 3399 s = arg->src; 3400 d = arg->dst; 3401 ch_volume = arg->context; 3402 3403 channels = arg->srcfmt->channels; 3404 for (i = 0; i < arg->count; i++) { 3405 for (ch = 0; ch < channels; ch++) { 3406 aint2_t val; 3407 val = *s++; 3408 val = AUDIO_SCALEDOWN(val * ch_volume[ch], 8); 3409 *d++ = (aint_t)val; 3410 } 3411 } 3412 } 3413 3414 /* 3415 * This filter performs conversion from stereo (or more channels) to mono. 3416 */ 3417 static void 3418 audio_track_chmix_mixLR(audio_filter_arg_t *arg) 3419 { 3420 const aint_t *s; 3421 aint_t *d; 3422 u_int i; 3423 3424 DIAGNOSTIC_filter_arg(arg); 3425 3426 s = arg->src; 3427 d = arg->dst; 3428 3429 for (i = 0; i < arg->count; i++) { 3430 *d++ = AUDIO_SCALEDOWN(s[0], 1) + AUDIO_SCALEDOWN(s[1], 1); 3431 s += arg->srcfmt->channels; 3432 } 3433 } 3434 3435 /* 3436 * This filter performs conversion from mono to stereo (or more channels). 3437 */ 3438 static void 3439 audio_track_chmix_dupLR(audio_filter_arg_t *arg) 3440 { 3441 const aint_t *s; 3442 aint_t *d; 3443 u_int i; 3444 u_int ch; 3445 u_int dstchannels; 3446 3447 DIAGNOSTIC_filter_arg(arg); 3448 3449 s = arg->src; 3450 d = arg->dst; 3451 dstchannels = arg->dstfmt->channels; 3452 3453 for (i = 0; i < arg->count; i++) { 3454 d[0] = s[0]; 3455 d[1] = s[0]; 3456 s++; 3457 d += dstchannels; 3458 } 3459 if (dstchannels > 2) { 3460 d = arg->dst; 3461 for (i = 0; i < arg->count; i++) { 3462 for (ch = 2; ch < dstchannels; ch++) { 3463 d[ch] = 0; 3464 } 3465 d += dstchannels; 3466 } 3467 } 3468 } 3469 3470 /* 3471 * This filter shrinks M channels into N channels. 3472 * Extra channels are discarded. 3473 */ 3474 static void 3475 audio_track_chmix_shrink(audio_filter_arg_t *arg) 3476 { 3477 const aint_t *s; 3478 aint_t *d; 3479 u_int i; 3480 u_int ch; 3481 3482 DIAGNOSTIC_filter_arg(arg); 3483 3484 s = arg->src; 3485 d = arg->dst; 3486 3487 for (i = 0; i < arg->count; i++) { 3488 for (ch = 0; ch < arg->dstfmt->channels; ch++) { 3489 *d++ = s[ch]; 3490 } 3491 s += arg->srcfmt->channels; 3492 } 3493 } 3494 3495 /* 3496 * This filter expands M channels into N channels. 3497 * Silence is inserted for missing channels. 3498 */ 3499 static void 3500 audio_track_chmix_expand(audio_filter_arg_t *arg) 3501 { 3502 const aint_t *s; 3503 aint_t *d; 3504 u_int i; 3505 u_int ch; 3506 u_int srcchannels; 3507 u_int dstchannels; 3508 3509 DIAGNOSTIC_filter_arg(arg); 3510 3511 s = arg->src; 3512 d = arg->dst; 3513 3514 srcchannels = arg->srcfmt->channels; 3515 dstchannels = arg->dstfmt->channels; 3516 for (i = 0; i < arg->count; i++) { 3517 for (ch = 0; ch < srcchannels; ch++) { 3518 *d++ = *s++; 3519 } 3520 for (; ch < dstchannels; ch++) { 3521 *d++ = 0; 3522 } 3523 } 3524 } 3525 3526 /* 3527 * This filter performs frequency conversion (up sampling). 3528 * It uses linear interpolation. 3529 */ 3530 static void 3531 audio_track_freq_up(audio_filter_arg_t *arg) 3532 { 3533 audio_track_t *track; 3534 audio_ring_t *src; 3535 audio_ring_t *dst; 3536 const aint_t *s; 3537 aint_t *d; 3538 aint_t prev[AUDIO_MAX_CHANNELS]; 3539 aint_t curr[AUDIO_MAX_CHANNELS]; 3540 aint_t grad[AUDIO_MAX_CHANNELS]; 3541 u_int i; 3542 u_int t; 3543 u_int step; 3544 u_int channels; 3545 u_int ch; 3546 int srcused; 3547 3548 track = arg->context; 3549 KASSERT(track); 3550 src = &track->freq.srcbuf; 3551 dst = track->freq.dst; 3552 DIAGNOSTIC_ring(dst); 3553 DIAGNOSTIC_ring(src); 3554 KASSERT(src->used > 0); 3555 KASSERTMSG(src->fmt.channels == dst->fmt.channels, 3556 "src->fmt.channels=%d dst->fmt.channels=%d", 3557 src->fmt.channels, dst->fmt.channels); 3558 KASSERTMSG(src->head % track->mixer->frames_per_block == 0, 3559 "src->head=%d track->mixer->frames_per_block=%d", 3560 src->head, track->mixer->frames_per_block); 3561 3562 s = arg->src; 3563 d = arg->dst; 3564 3565 /* 3566 * In order to faciliate interpolation for each block, slide (delay) 3567 * input by one sample. As a result, strictly speaking, the output 3568 * phase is delayed by 1/dstfreq. However, I believe there is no 3569 * observable impact. 3570 * 3571 * Example) 3572 * srcfreq:dstfreq = 1:3 3573 * 3574 * A - - 3575 * | 3576 * | 3577 * | B - - 3578 * +-----+-----> input timeframe 3579 * 0 1 3580 * 3581 * 0 1 3582 * +-----+-----> input timeframe 3583 * | A 3584 * | x x 3585 * | x x 3586 * x (B) 3587 * +-+-+-+-+-+-> output timeframe 3588 * 0 1 2 3 4 5 3589 */ 3590 3591 /* Last samples in previous block */ 3592 channels = src->fmt.channels; 3593 for (ch = 0; ch < channels; ch++) { 3594 prev[ch] = track->freq_prev[ch]; 3595 curr[ch] = track->freq_curr[ch]; 3596 grad[ch] = curr[ch] - prev[ch]; 3597 } 3598 3599 step = track->freq_step; 3600 t = track->freq_current; 3601 //#define FREQ_DEBUG 3602 #if defined(FREQ_DEBUG) 3603 #define PRINTF(fmt...) printf(fmt) 3604 #else 3605 #define PRINTF(fmt...) do { } while (0) 3606 #endif 3607 srcused = src->used; 3608 PRINTF("upstart step=%d leap=%d", step, track->freq_leap); 3609 PRINTF(" srcused=%d arg->count=%u", src->used, arg->count); 3610 PRINTF(" prev=%d curr=%d grad=%d", prev[0], curr[0], grad[0]); 3611 PRINTF(" t=%d\n", t); 3612 3613 for (i = 0; i < arg->count; i++) { 3614 PRINTF("i=%d t=%5d", i, t); 3615 if (t >= 65536) { 3616 for (ch = 0; ch < channels; ch++) { 3617 prev[ch] = curr[ch]; 3618 curr[ch] = *s++; 3619 grad[ch] = curr[ch] - prev[ch]; 3620 } 3621 PRINTF(" prev=%d s[%d]=%d", 3622 prev[0], src->used - srcused, curr[0]); 3623 3624 /* Update */ 3625 t -= 65536; 3626 srcused--; 3627 if (srcused < 0) { 3628 PRINTF(" break\n"); 3629 break; 3630 } 3631 } 3632 3633 for (ch = 0; ch < channels; ch++) { 3634 *d++ = prev[ch] + (aint2_t)grad[ch] * t / 65536; 3635 #if defined(FREQ_DEBUG) 3636 if (ch == 0) 3637 printf(" t=%5d *d=%d", t, d[-1]); 3638 #endif 3639 } 3640 t += step; 3641 3642 PRINTF("\n"); 3643 } 3644 PRINTF("end prev=%d curr=%d\n", prev[0], curr[0]); 3645 3646 auring_take(src, src->used); 3647 auring_push(dst, i); 3648 3649 /* Adjust */ 3650 t += track->freq_leap; 3651 3652 track->freq_current = t; 3653 for (ch = 0; ch < channels; ch++) { 3654 track->freq_prev[ch] = prev[ch]; 3655 track->freq_curr[ch] = curr[ch]; 3656 } 3657 } 3658 3659 /* 3660 * This filter performs frequency conversion (down sampling). 3661 * It uses simple thinning. 3662 */ 3663 static void 3664 audio_track_freq_down(audio_filter_arg_t *arg) 3665 { 3666 audio_track_t *track; 3667 audio_ring_t *src; 3668 audio_ring_t *dst; 3669 const aint_t *s0; 3670 aint_t *d; 3671 u_int i; 3672 u_int t; 3673 u_int step; 3674 u_int ch; 3675 u_int channels; 3676 3677 track = arg->context; 3678 KASSERT(track); 3679 src = &track->freq.srcbuf; 3680 dst = track->freq.dst; 3681 3682 DIAGNOSTIC_ring(dst); 3683 DIAGNOSTIC_ring(src); 3684 KASSERT(src->used > 0); 3685 KASSERTMSG(src->fmt.channels == dst->fmt.channels, 3686 "src->fmt.channels=%d dst->fmt.channels=%d", 3687 src->fmt.channels, dst->fmt.channels); 3688 KASSERTMSG(src->head % track->mixer->frames_per_block == 0, 3689 "src->head=%d track->mixer->frames_per_block=%d", 3690 src->head, track->mixer->frames_per_block); 3691 3692 s0 = arg->src; 3693 d = arg->dst; 3694 t = track->freq_current; 3695 step = track->freq_step; 3696 channels = dst->fmt.channels; 3697 PRINTF("downstart step=%d leap=%d", step, track->freq_leap); 3698 PRINTF(" srcused=%d arg->count=%u", src->used, arg->count); 3699 PRINTF(" t=%d\n", t); 3700 3701 for (i = 0; i < arg->count && t / 65536 < src->used; i++) { 3702 const aint_t *s; 3703 PRINTF("i=%4d t=%10d", i, t); 3704 s = s0 + (t / 65536) * channels; 3705 PRINTF(" s=%5ld", (s - s0) / channels); 3706 for (ch = 0; ch < channels; ch++) { 3707 if (ch == 0) PRINTF(" *s=%d", s[ch]); 3708 *d++ = s[ch]; 3709 } 3710 PRINTF("\n"); 3711 t += step; 3712 } 3713 t += track->freq_leap; 3714 PRINTF("end t=%d\n", t); 3715 auring_take(src, src->used); 3716 auring_push(dst, i); 3717 track->freq_current = t % 65536; 3718 } 3719 3720 /* 3721 * Creates track and returns it. 3722 */ 3723 audio_track_t * 3724 audio_track_create(struct audio_softc *sc, audio_trackmixer_t *mixer) 3725 { 3726 audio_track_t *track; 3727 static int newid = 0; 3728 3729 track = kmem_zalloc(sizeof(*track), KM_SLEEP); 3730 3731 track->id = newid++; 3732 track->mixer = mixer; 3733 track->mode = mixer->mode; 3734 3735 /* Do TRACE after id is assigned. */ 3736 TRACET(3, track, "for %s", 3737 mixer->mode == AUMODE_PLAY ? "playback" : "recording"); 3738 3739 #if defined(AUDIO_SUPPORT_TRACK_VOLUME) 3740 track->volume = 256; 3741 #endif 3742 for (int i = 0; i < AUDIO_MAX_CHANNELS; i++) { 3743 track->ch_volume[i] = 256; 3744 } 3745 3746 return track; 3747 } 3748 3749 /* 3750 * Release all resources of the track and track itself. 3751 * track must not be NULL. Don't specify the track within the file 3752 * structure linked from sc->sc_files. 3753 */ 3754 static void 3755 audio_track_destroy(audio_track_t *track) 3756 { 3757 3758 KASSERT(track); 3759 3760 audio_free_usrbuf(track); 3761 audio_free(track->codec.srcbuf.mem); 3762 audio_free(track->chvol.srcbuf.mem); 3763 audio_free(track->chmix.srcbuf.mem); 3764 audio_free(track->freq.srcbuf.mem); 3765 audio_free(track->outbuf.mem); 3766 3767 kmem_free(track, sizeof(*track)); 3768 } 3769 3770 /* 3771 * It returns encoding conversion filter according to src and dst format. 3772 * If it is not a convertible pair, it returns NULL. Either src or dst 3773 * must be internal format. 3774 */ 3775 static audio_filter_t 3776 audio_track_get_codec(audio_track_t *track, const audio_format2_t *src, 3777 const audio_format2_t *dst) 3778 { 3779 3780 if (audio_format2_is_internal(src)) { 3781 if (dst->encoding == AUDIO_ENCODING_ULAW) { 3782 return audio_internal_to_mulaw; 3783 } else if (dst->encoding == AUDIO_ENCODING_ALAW) { 3784 return audio_internal_to_alaw; 3785 } else if (audio_format2_is_linear(dst)) { 3786 switch (dst->stride) { 3787 case 8: 3788 return audio_internal_to_linear8; 3789 case 16: 3790 return audio_internal_to_linear16; 3791 #if defined(AUDIO_SUPPORT_LINEAR24) 3792 case 24: 3793 return audio_internal_to_linear24; 3794 #endif 3795 case 32: 3796 return audio_internal_to_linear32; 3797 default: 3798 TRACET(1, track, "unsupported %s stride %d", 3799 "dst", dst->stride); 3800 goto abort; 3801 } 3802 } 3803 } else if (audio_format2_is_internal(dst)) { 3804 if (src->encoding == AUDIO_ENCODING_ULAW) { 3805 return audio_mulaw_to_internal; 3806 } else if (src->encoding == AUDIO_ENCODING_ALAW) { 3807 return audio_alaw_to_internal; 3808 } else if (audio_format2_is_linear(src)) { 3809 switch (src->stride) { 3810 case 8: 3811 return audio_linear8_to_internal; 3812 case 16: 3813 return audio_linear16_to_internal; 3814 #if defined(AUDIO_SUPPORT_LINEAR24) 3815 case 24: 3816 return audio_linear24_to_internal; 3817 #endif 3818 case 32: 3819 return audio_linear32_to_internal; 3820 default: 3821 TRACET(1, track, "unsupported %s stride %d", 3822 "src", src->stride); 3823 goto abort; 3824 } 3825 } 3826 } 3827 3828 TRACET(1, track, "unsupported encoding"); 3829 abort: 3830 #if defined(AUDIO_DEBUG) 3831 if (audiodebug >= 2) { 3832 char buf[100]; 3833 audio_format2_tostr(buf, sizeof(buf), src); 3834 TRACET(2, track, "src %s", buf); 3835 audio_format2_tostr(buf, sizeof(buf), dst); 3836 TRACET(2, track, "dst %s", buf); 3837 } 3838 #endif 3839 return NULL; 3840 } 3841 3842 /* 3843 * Initialize the codec stage of this track as necessary. 3844 * If successful, it initializes the codec stage as necessary, stores updated 3845 * last_dst in *last_dstp in any case, and returns 0. 3846 * Otherwise, it returns errno without modifying *last_dstp. 3847 */ 3848 static int 3849 audio_track_init_codec(audio_track_t *track, audio_ring_t **last_dstp) 3850 { 3851 audio_ring_t *last_dst; 3852 audio_ring_t *srcbuf; 3853 audio_format2_t *srcfmt; 3854 audio_format2_t *dstfmt; 3855 audio_filter_arg_t *arg; 3856 u_int len; 3857 int error; 3858 3859 KASSERT(track); 3860 3861 last_dst = *last_dstp; 3862 dstfmt = &last_dst->fmt; 3863 srcfmt = &track->inputfmt; 3864 srcbuf = &track->codec.srcbuf; 3865 error = 0; 3866 3867 if (srcfmt->encoding != dstfmt->encoding 3868 || srcfmt->precision != dstfmt->precision 3869 || srcfmt->stride != dstfmt->stride) { 3870 track->codec.dst = last_dst; 3871 3872 srcbuf->fmt = *dstfmt; 3873 srcbuf->fmt.encoding = srcfmt->encoding; 3874 srcbuf->fmt.precision = srcfmt->precision; 3875 srcbuf->fmt.stride = srcfmt->stride; 3876 3877 track->codec.filter = audio_track_get_codec(track, 3878 &srcbuf->fmt, dstfmt); 3879 if (track->codec.filter == NULL) { 3880 error = EINVAL; 3881 goto abort; 3882 } 3883 3884 srcbuf->head = 0; 3885 srcbuf->used = 0; 3886 srcbuf->capacity = frame_per_block(track->mixer, &srcbuf->fmt); 3887 len = auring_bytelen(srcbuf); 3888 srcbuf->mem = audio_realloc(srcbuf->mem, len); 3889 3890 arg = &track->codec.arg; 3891 arg->srcfmt = &srcbuf->fmt; 3892 arg->dstfmt = dstfmt; 3893 arg->context = NULL; 3894 3895 *last_dstp = srcbuf; 3896 return 0; 3897 } 3898 3899 abort: 3900 track->codec.filter = NULL; 3901 audio_free(srcbuf->mem); 3902 return error; 3903 } 3904 3905 /* 3906 * Initialize the chvol stage of this track as necessary. 3907 * If successful, it initializes the chvol stage as necessary, stores updated 3908 * last_dst in *last_dstp in any case, and returns 0. 3909 * Otherwise, it returns errno without modifying *last_dstp. 3910 */ 3911 static int 3912 audio_track_init_chvol(audio_track_t *track, audio_ring_t **last_dstp) 3913 { 3914 audio_ring_t *last_dst; 3915 audio_ring_t *srcbuf; 3916 audio_format2_t *srcfmt; 3917 audio_format2_t *dstfmt; 3918 audio_filter_arg_t *arg; 3919 u_int len; 3920 int error; 3921 3922 KASSERT(track); 3923 3924 last_dst = *last_dstp; 3925 dstfmt = &last_dst->fmt; 3926 srcfmt = &track->inputfmt; 3927 srcbuf = &track->chvol.srcbuf; 3928 error = 0; 3929 3930 /* Check whether channel volume conversion is necessary. */ 3931 bool use_chvol = false; 3932 for (int ch = 0; ch < srcfmt->channels; ch++) { 3933 if (track->ch_volume[ch] != 256) { 3934 use_chvol = true; 3935 break; 3936 } 3937 } 3938 3939 if (use_chvol == true) { 3940 track->chvol.dst = last_dst; 3941 track->chvol.filter = audio_track_chvol; 3942 3943 srcbuf->fmt = *dstfmt; 3944 /* no format conversion occurs */ 3945 3946 srcbuf->head = 0; 3947 srcbuf->used = 0; 3948 srcbuf->capacity = frame_per_block(track->mixer, &srcbuf->fmt); 3949 len = auring_bytelen(srcbuf); 3950 srcbuf->mem = audio_realloc(srcbuf->mem, len); 3951 3952 arg = &track->chvol.arg; 3953 arg->srcfmt = &srcbuf->fmt; 3954 arg->dstfmt = dstfmt; 3955 arg->context = track->ch_volume; 3956 3957 *last_dstp = srcbuf; 3958 return 0; 3959 } 3960 3961 track->chvol.filter = NULL; 3962 audio_free(srcbuf->mem); 3963 return error; 3964 } 3965 3966 /* 3967 * Initialize the chmix stage of this track as necessary. 3968 * If successful, it initializes the chmix stage as necessary, stores updated 3969 * last_dst in *last_dstp in any case, and returns 0. 3970 * Otherwise, it returns errno without modifying *last_dstp. 3971 */ 3972 static int 3973 audio_track_init_chmix(audio_track_t *track, audio_ring_t **last_dstp) 3974 { 3975 audio_ring_t *last_dst; 3976 audio_ring_t *srcbuf; 3977 audio_format2_t *srcfmt; 3978 audio_format2_t *dstfmt; 3979 audio_filter_arg_t *arg; 3980 u_int srcch; 3981 u_int dstch; 3982 u_int len; 3983 int error; 3984 3985 KASSERT(track); 3986 3987 last_dst = *last_dstp; 3988 dstfmt = &last_dst->fmt; 3989 srcfmt = &track->inputfmt; 3990 srcbuf = &track->chmix.srcbuf; 3991 error = 0; 3992 3993 srcch = srcfmt->channels; 3994 dstch = dstfmt->channels; 3995 if (srcch != dstch) { 3996 track->chmix.dst = last_dst; 3997 3998 if (srcch >= 2 && dstch == 1) { 3999 track->chmix.filter = audio_track_chmix_mixLR; 4000 } else if (srcch == 1 && dstch >= 2) { 4001 track->chmix.filter = audio_track_chmix_dupLR; 4002 } else if (srcch > dstch) { 4003 track->chmix.filter = audio_track_chmix_shrink; 4004 } else { 4005 track->chmix.filter = audio_track_chmix_expand; 4006 } 4007 4008 srcbuf->fmt = *dstfmt; 4009 srcbuf->fmt.channels = srcch; 4010 4011 srcbuf->head = 0; 4012 srcbuf->used = 0; 4013 /* XXX The buffer size should be able to calculate. */ 4014 srcbuf->capacity = frame_per_block(track->mixer, &srcbuf->fmt); 4015 len = auring_bytelen(srcbuf); 4016 srcbuf->mem = audio_realloc(srcbuf->mem, len); 4017 4018 arg = &track->chmix.arg; 4019 arg->srcfmt = &srcbuf->fmt; 4020 arg->dstfmt = dstfmt; 4021 arg->context = NULL; 4022 4023 *last_dstp = srcbuf; 4024 return 0; 4025 } 4026 4027 track->chmix.filter = NULL; 4028 audio_free(srcbuf->mem); 4029 return error; 4030 } 4031 4032 /* 4033 * Initialize the freq stage of this track as necessary. 4034 * If successful, it initializes the freq stage as necessary, stores updated 4035 * last_dst in *last_dstp in any case, and returns 0. 4036 * Otherwise, it returns errno without modifying *last_dstp. 4037 */ 4038 static int 4039 audio_track_init_freq(audio_track_t *track, audio_ring_t **last_dstp) 4040 { 4041 audio_ring_t *last_dst; 4042 audio_ring_t *srcbuf; 4043 audio_format2_t *srcfmt; 4044 audio_format2_t *dstfmt; 4045 audio_filter_arg_t *arg; 4046 uint32_t srcfreq; 4047 uint32_t dstfreq; 4048 u_int dst_capacity; 4049 u_int mod; 4050 u_int len; 4051 int error; 4052 4053 KASSERT(track); 4054 4055 last_dst = *last_dstp; 4056 dstfmt = &last_dst->fmt; 4057 srcfmt = &track->inputfmt; 4058 srcbuf = &track->freq.srcbuf; 4059 error = 0; 4060 4061 srcfreq = srcfmt->sample_rate; 4062 dstfreq = dstfmt->sample_rate; 4063 if (srcfreq != dstfreq) { 4064 track->freq.dst = last_dst; 4065 4066 memset(track->freq_prev, 0, sizeof(track->freq_prev)); 4067 memset(track->freq_curr, 0, sizeof(track->freq_curr)); 4068 4069 /* freq_step is the ratio of src/dst when let dst 65536. */ 4070 track->freq_step = (uint64_t)srcfreq * 65536 / dstfreq; 4071 4072 dst_capacity = frame_per_block(track->mixer, dstfmt); 4073 mod = (uint64_t)srcfreq * 65536 % dstfreq; 4074 track->freq_leap = (mod * dst_capacity + dstfreq / 2) / dstfreq; 4075 4076 if (track->freq_step < 65536) { 4077 track->freq.filter = audio_track_freq_up; 4078 /* In order to carry at the first time. */ 4079 track->freq_current = 65536; 4080 } else { 4081 track->freq.filter = audio_track_freq_down; 4082 track->freq_current = 0; 4083 } 4084 4085 srcbuf->fmt = *dstfmt; 4086 srcbuf->fmt.sample_rate = srcfreq; 4087 4088 srcbuf->head = 0; 4089 srcbuf->used = 0; 4090 srcbuf->capacity = frame_per_block(track->mixer, &srcbuf->fmt); 4091 len = auring_bytelen(srcbuf); 4092 srcbuf->mem = audio_realloc(srcbuf->mem, len); 4093 4094 arg = &track->freq.arg; 4095 arg->srcfmt = &srcbuf->fmt; 4096 arg->dstfmt = dstfmt;/*&last_dst->fmt;*/ 4097 arg->context = track; 4098 4099 *last_dstp = srcbuf; 4100 return 0; 4101 } 4102 4103 track->freq.filter = NULL; 4104 audio_free(srcbuf->mem); 4105 return error; 4106 } 4107 4108 /* 4109 * When playing back: (e.g. if codec and freq stage are valid) 4110 * 4111 * write 4112 * | uiomove 4113 * v 4114 * usrbuf [...............] byte ring buffer (mmap-able) 4115 * | memcpy 4116 * v 4117 * codec.srcbuf[....] 1 block (ring) buffer <-- stage input 4118 * .dst ----+ 4119 * | convert 4120 * v 4121 * freq.srcbuf [....] 1 block (ring) buffer 4122 * .dst ----+ 4123 * | convert 4124 * v 4125 * outbuf [...............] NBLKOUT blocks ring buffer 4126 * 4127 * 4128 * When recording: 4129 * 4130 * freq.srcbuf [...............] NBLKOUT blocks ring buffer <-- stage input 4131 * .dst ----+ 4132 * | convert 4133 * v 4134 * codec.srcbuf[.....] 1 block (ring) buffer 4135 * .dst ----+ 4136 * | convert 4137 * v 4138 * outbuf [.....] 1 block (ring) buffer 4139 * | memcpy 4140 * v 4141 * usrbuf [...............] byte ring buffer (mmap-able *) 4142 * | uiomove 4143 * v 4144 * read 4145 * 4146 * *: usrbuf for recording is also mmap-able due to symmetry with 4147 * playback buffer, but for now mmap will never happen for recording. 4148 */ 4149 4150 /* 4151 * Set the userland format of this track. 4152 * usrfmt argument should be parameter verified with audio_check_params(). 4153 * It will release and reallocate all internal conversion buffers. 4154 * It returns 0 if successful. Otherwise it returns errno with clearing all 4155 * internal buffers. 4156 * It must be called without sc_intr_lock since uvm_* routines require non 4157 * intr_lock state. 4158 * It must be called with track lock held since it may release and reallocate 4159 * outbuf. 4160 */ 4161 static int 4162 audio_track_set_format(audio_track_t *track, audio_format2_t *usrfmt) 4163 { 4164 struct audio_softc *sc; 4165 u_int newbufsize; 4166 u_int oldblksize; 4167 u_int len; 4168 int error; 4169 4170 KASSERT(track); 4171 sc = track->mixer->sc; 4172 4173 /* usrbuf is the closest buffer to the userland. */ 4174 track->usrbuf.fmt = *usrfmt; 4175 4176 /* 4177 * For references, one block size (in 40msec) is: 4178 * 320 bytes = 204 blocks/64KB for mulaw/8kHz/1ch 4179 * 7680 bytes = 8 blocks/64KB for s16/48kHz/2ch 4180 * 30720 bytes = 90 KB/3blocks for s16/48kHz/8ch 4181 * 61440 bytes = 180 KB/3blocks for s16/96kHz/8ch 4182 * 245760 bytes = 720 KB/3blocks for s32/192kHz/8ch 4183 * 4184 * For example, 4185 * 1) If usrbuf_blksize = 7056 (s16/44.1k/2ch) and PAGE_SIZE = 8192, 4186 * newbufsize = rounddown(65536 / 7056) = 63504 4187 * newvsize = roundup2(63504, PAGE_SIZE) = 65536 4188 * Therefore it maps 8 * 8K pages and usrbuf->capacity = 63504. 4189 * 4190 * 2) If usrbuf_blksize = 7680 (s16/48k/2ch) and PAGE_SIZE = 4096, 4191 * newbufsize = rounddown(65536 / 7680) = 61440 4192 * newvsize = roundup2(61440, PAGE_SIZE) = 61440 (= 15 pages) 4193 * Therefore it maps 15 * 4K pages and usrbuf->capacity = 61440. 4194 */ 4195 oldblksize = track->usrbuf_blksize; 4196 track->usrbuf_blksize = frametobyte(&track->usrbuf.fmt, 4197 frame_per_block(track->mixer, &track->usrbuf.fmt)); 4198 track->usrbuf.head = 0; 4199 track->usrbuf.used = 0; 4200 newbufsize = MAX(track->usrbuf_blksize * AUMINNOBLK, 65536); 4201 newbufsize = rounddown(newbufsize, track->usrbuf_blksize); 4202 error = audio_realloc_usrbuf(track, newbufsize); 4203 if (error) { 4204 device_printf(sc->sc_dev, "malloc usrbuf(%d) failed\n", 4205 newbufsize); 4206 goto error; 4207 } 4208 4209 /* Recalc water mark. */ 4210 if (track->usrbuf_blksize != oldblksize) { 4211 if (audio_track_is_playback(track)) { 4212 /* Set high at 100%, low at 75%. */ 4213 track->usrbuf_usedhigh = track->usrbuf.capacity; 4214 track->usrbuf_usedlow = track->usrbuf.capacity * 3 / 4; 4215 } else { 4216 /* Set high at 100% minus 1block(?), low at 0% */ 4217 track->usrbuf_usedhigh = track->usrbuf.capacity - 4218 track->usrbuf_blksize; 4219 track->usrbuf_usedlow = 0; 4220 } 4221 } 4222 4223 /* Stage buffer */ 4224 audio_ring_t *last_dst = &track->outbuf; 4225 if (audio_track_is_playback(track)) { 4226 /* On playback, initialize from the mixer side in order. */ 4227 track->inputfmt = *usrfmt; 4228 track->outbuf.fmt = track->mixer->track_fmt; 4229 4230 if ((error = audio_track_init_freq(track, &last_dst)) != 0) 4231 goto error; 4232 if ((error = audio_track_init_chmix(track, &last_dst)) != 0) 4233 goto error; 4234 if ((error = audio_track_init_chvol(track, &last_dst)) != 0) 4235 goto error; 4236 if ((error = audio_track_init_codec(track, &last_dst)) != 0) 4237 goto error; 4238 } else { 4239 /* On recording, initialize from userland side in order. */ 4240 track->inputfmt = track->mixer->track_fmt; 4241 track->outbuf.fmt = *usrfmt; 4242 4243 if ((error = audio_track_init_codec(track, &last_dst)) != 0) 4244 goto error; 4245 if ((error = audio_track_init_chvol(track, &last_dst)) != 0) 4246 goto error; 4247 if ((error = audio_track_init_chmix(track, &last_dst)) != 0) 4248 goto error; 4249 if ((error = audio_track_init_freq(track, &last_dst)) != 0) 4250 goto error; 4251 } 4252 #if 0 4253 /* debug */ 4254 if (track->freq.filter) { 4255 audio_print_format2("freq src", &track->freq.srcbuf.fmt); 4256 audio_print_format2("freq dst", &track->freq.dst->fmt); 4257 } 4258 if (track->chmix.filter) { 4259 audio_print_format2("chmix src", &track->chmix.srcbuf.fmt); 4260 audio_print_format2("chmix dst", &track->chmix.dst->fmt); 4261 } 4262 if (track->chvol.filter) { 4263 audio_print_format2("chvol src", &track->chvol.srcbuf.fmt); 4264 audio_print_format2("chvol dst", &track->chvol.dst->fmt); 4265 } 4266 if (track->codec.filter) { 4267 audio_print_format2("codec src", &track->codec.srcbuf.fmt); 4268 audio_print_format2("codec dst", &track->codec.dst->fmt); 4269 } 4270 #endif 4271 4272 /* Stage input buffer */ 4273 track->input = last_dst; 4274 4275 /* 4276 * On the recording track, make the first stage a ring buffer. 4277 * XXX is there a better way? 4278 */ 4279 if (audio_track_is_record(track)) { 4280 track->input->capacity = NBLKOUT * 4281 frame_per_block(track->mixer, &track->input->fmt); 4282 len = auring_bytelen(track->input); 4283 track->input->mem = audio_realloc(track->input->mem, len); 4284 } 4285 4286 /* 4287 * Output buffer. 4288 * On the playback track, its capacity is NBLKOUT blocks. 4289 * On the recording track, its capacity is 1 block. 4290 */ 4291 track->outbuf.head = 0; 4292 track->outbuf.used = 0; 4293 track->outbuf.capacity = frame_per_block(track->mixer, 4294 &track->outbuf.fmt); 4295 if (audio_track_is_playback(track)) 4296 track->outbuf.capacity *= NBLKOUT; 4297 len = auring_bytelen(&track->outbuf); 4298 track->outbuf.mem = audio_realloc(track->outbuf.mem, len); 4299 if (track->outbuf.mem == NULL) { 4300 device_printf(sc->sc_dev, "malloc outbuf(%d) failed\n", len); 4301 error = ENOMEM; 4302 goto error; 4303 } 4304 4305 #if defined(AUDIO_DEBUG) 4306 if (audiodebug >= 3) { 4307 struct audio_track_debugbuf m; 4308 4309 memset(&m, 0, sizeof(m)); 4310 snprintf(m.outbuf, sizeof(m.outbuf), " out=%d", 4311 track->outbuf.capacity * frametobyte(&track->outbuf.fmt,1)); 4312 if (track->freq.filter) 4313 snprintf(m.freq, sizeof(m.freq), " freq=%d", 4314 track->freq.srcbuf.capacity * 4315 frametobyte(&track->freq.srcbuf.fmt, 1)); 4316 if (track->chmix.filter) 4317 snprintf(m.chmix, sizeof(m.chmix), " chmix=%d", 4318 track->chmix.srcbuf.capacity * 4319 frametobyte(&track->chmix.srcbuf.fmt, 1)); 4320 if (track->chvol.filter) 4321 snprintf(m.chvol, sizeof(m.chvol), " chvol=%d", 4322 track->chvol.srcbuf.capacity * 4323 frametobyte(&track->chvol.srcbuf.fmt, 1)); 4324 if (track->codec.filter) 4325 snprintf(m.codec, sizeof(m.codec), " codec=%d", 4326 track->codec.srcbuf.capacity * 4327 frametobyte(&track->codec.srcbuf.fmt, 1)); 4328 snprintf(m.usrbuf, sizeof(m.usrbuf), 4329 " usr=%d", track->usrbuf.capacity); 4330 4331 if (audio_track_is_playback(track)) { 4332 TRACET(0, track, "bufsize%s%s%s%s%s%s", 4333 m.outbuf, m.freq, m.chmix, 4334 m.chvol, m.codec, m.usrbuf); 4335 } else { 4336 TRACET(0, track, "bufsize%s%s%s%s%s%s", 4337 m.freq, m.chmix, m.chvol, 4338 m.codec, m.outbuf, m.usrbuf); 4339 } 4340 } 4341 #endif 4342 return 0; 4343 4344 error: 4345 audio_free_usrbuf(track); 4346 audio_free(track->codec.srcbuf.mem); 4347 audio_free(track->chvol.srcbuf.mem); 4348 audio_free(track->chmix.srcbuf.mem); 4349 audio_free(track->freq.srcbuf.mem); 4350 audio_free(track->outbuf.mem); 4351 return error; 4352 } 4353 4354 /* 4355 * Fill silence frames (as the internal format) up to 1 block 4356 * if the ring is not empty and less than 1 block. 4357 * It returns the number of appended frames. 4358 */ 4359 static int 4360 audio_append_silence(audio_track_t *track, audio_ring_t *ring) 4361 { 4362 int fpb; 4363 int n; 4364 4365 KASSERT(track); 4366 KASSERT(audio_format2_is_internal(&ring->fmt)); 4367 4368 /* XXX is n correct? */ 4369 /* XXX memset uses frametobyte()? */ 4370 4371 if (ring->used == 0) 4372 return 0; 4373 4374 fpb = frame_per_block(track->mixer, &ring->fmt); 4375 if (ring->used >= fpb) 4376 return 0; 4377 4378 n = (ring->capacity - ring->used) % fpb; 4379 4380 KASSERTMSG(auring_get_contig_free(ring) >= n, 4381 "auring_get_contig_free(ring)=%d n=%d", 4382 auring_get_contig_free(ring), n); 4383 4384 memset(auring_tailptr_aint(ring), 0, 4385 n * ring->fmt.channels * sizeof(aint_t)); 4386 auring_push(ring, n); 4387 return n; 4388 } 4389 4390 /* 4391 * Execute the conversion stage. 4392 * It prepares arg from this stage and executes stage->filter. 4393 * It must be called only if stage->filter is not NULL. 4394 * 4395 * For stages other than frequency conversion, the function increments 4396 * src and dst counters here. For frequency conversion stage, on the 4397 * other hand, the function does not touch src and dst counters and 4398 * filter side has to increment them. 4399 */ 4400 static void 4401 audio_apply_stage(audio_track_t *track, audio_stage_t *stage, bool isfreq) 4402 { 4403 audio_filter_arg_t *arg; 4404 int srccount; 4405 int dstcount; 4406 int count; 4407 4408 KASSERT(track); 4409 KASSERT(stage->filter); 4410 4411 srccount = auring_get_contig_used(&stage->srcbuf); 4412 dstcount = auring_get_contig_free(stage->dst); 4413 4414 if (isfreq) { 4415 KASSERTMSG(srccount > 0, "freq but srccount=%d", srccount); 4416 count = uimin(dstcount, track->mixer->frames_per_block); 4417 } else { 4418 count = uimin(srccount, dstcount); 4419 } 4420 4421 if (count > 0) { 4422 arg = &stage->arg; 4423 arg->src = auring_headptr(&stage->srcbuf); 4424 arg->dst = auring_tailptr(stage->dst); 4425 arg->count = count; 4426 4427 stage->filter(arg); 4428 4429 if (!isfreq) { 4430 auring_take(&stage->srcbuf, count); 4431 auring_push(stage->dst, count); 4432 } 4433 } 4434 } 4435 4436 /* 4437 * Produce output buffer for playback from user input buffer. 4438 * It must be called only if usrbuf is not empty and outbuf is 4439 * available at least one free block. 4440 */ 4441 static void 4442 audio_track_play(audio_track_t *track) 4443 { 4444 audio_ring_t *usrbuf; 4445 audio_ring_t *input; 4446 int count; 4447 int framesize; 4448 int bytes; 4449 4450 KASSERT(track); 4451 KASSERT(track->lock); 4452 TRACET(4, track, "start pstate=%d", track->pstate); 4453 4454 /* At this point usrbuf must not be empty. */ 4455 KASSERT(track->usrbuf.used > 0); 4456 /* Also, outbuf must be available at least one block. */ 4457 count = auring_get_contig_free(&track->outbuf); 4458 KASSERTMSG(count >= frame_per_block(track->mixer, &track->outbuf.fmt), 4459 "count=%d fpb=%d", 4460 count, frame_per_block(track->mixer, &track->outbuf.fmt)); 4461 4462 /* XXX TODO: is this necessary for now? */ 4463 int track_count_0 = track->outbuf.used; 4464 4465 usrbuf = &track->usrbuf; 4466 input = track->input; 4467 4468 /* 4469 * framesize is always 1 byte or more since all formats supported as 4470 * usrfmt(=input) have 8bit or more stride. 4471 */ 4472 framesize = frametobyte(&input->fmt, 1); 4473 KASSERT(framesize >= 1); 4474 4475 /* The next stage of usrbuf (=input) must be available. */ 4476 KASSERT(auring_get_contig_free(input) > 0); 4477 4478 /* 4479 * Copy usrbuf up to 1block to input buffer. 4480 * count is the number of frames to copy from usrbuf. 4481 * bytes is the number of bytes to copy from usrbuf. However it is 4482 * not copied less than one frame. 4483 */ 4484 count = uimin(usrbuf->used, track->usrbuf_blksize) / framesize; 4485 bytes = count * framesize; 4486 4487 track->usrbuf_stamp += bytes; 4488 4489 if (usrbuf->head + bytes < usrbuf->capacity) { 4490 memcpy((uint8_t *)input->mem + auring_tail(input) * framesize, 4491 (uint8_t *)usrbuf->mem + usrbuf->head, 4492 bytes); 4493 auring_push(input, count); 4494 auring_take(usrbuf, bytes); 4495 } else { 4496 int bytes1; 4497 int bytes2; 4498 4499 bytes1 = auring_get_contig_used(usrbuf); 4500 KASSERTMSG(bytes1 % framesize == 0, 4501 "bytes1=%d framesize=%d", bytes1, framesize); 4502 memcpy((uint8_t *)input->mem + auring_tail(input) * framesize, 4503 (uint8_t *)usrbuf->mem + usrbuf->head, 4504 bytes1); 4505 auring_push(input, bytes1 / framesize); 4506 auring_take(usrbuf, bytes1); 4507 4508 bytes2 = bytes - bytes1; 4509 memcpy((uint8_t *)input->mem + auring_tail(input) * framesize, 4510 (uint8_t *)usrbuf->mem + usrbuf->head, 4511 bytes2); 4512 auring_push(input, bytes2 / framesize); 4513 auring_take(usrbuf, bytes2); 4514 } 4515 4516 /* Encoding conversion */ 4517 if (track->codec.filter) 4518 audio_apply_stage(track, &track->codec, false); 4519 4520 /* Channel volume */ 4521 if (track->chvol.filter) 4522 audio_apply_stage(track, &track->chvol, false); 4523 4524 /* Channel mix */ 4525 if (track->chmix.filter) 4526 audio_apply_stage(track, &track->chmix, false); 4527 4528 /* Frequency conversion */ 4529 /* 4530 * Since the frequency conversion needs correction for each block, 4531 * it rounds up to 1 block. 4532 */ 4533 if (track->freq.filter) { 4534 int n; 4535 n = audio_append_silence(track, &track->freq.srcbuf); 4536 if (n > 0) { 4537 TRACET(4, track, 4538 "freq.srcbuf add silence %d -> %d/%d/%d", 4539 n, 4540 track->freq.srcbuf.head, 4541 track->freq.srcbuf.used, 4542 track->freq.srcbuf.capacity); 4543 } 4544 if (track->freq.srcbuf.used > 0) { 4545 audio_apply_stage(track, &track->freq, true); 4546 } 4547 } 4548 4549 if (bytes < track->usrbuf_blksize) { 4550 /* 4551 * Clear all conversion buffer pointer if the conversion was 4552 * not exactly one block. These conversion stage buffers are 4553 * certainly circular buffers because of symmetry with the 4554 * previous and next stage buffer. However, since they are 4555 * treated as simple contiguous buffers in operation, so head 4556 * always should point 0. This may happen during drain-age. 4557 */ 4558 TRACET(4, track, "reset stage"); 4559 if (track->codec.filter) { 4560 KASSERT(track->codec.srcbuf.used == 0); 4561 track->codec.srcbuf.head = 0; 4562 } 4563 if (track->chvol.filter) { 4564 KASSERT(track->chvol.srcbuf.used == 0); 4565 track->chvol.srcbuf.head = 0; 4566 } 4567 if (track->chmix.filter) { 4568 KASSERT(track->chmix.srcbuf.used == 0); 4569 track->chmix.srcbuf.head = 0; 4570 } 4571 if (track->freq.filter) { 4572 KASSERT(track->freq.srcbuf.used == 0); 4573 track->freq.srcbuf.head = 0; 4574 } 4575 } 4576 4577 if (track->input == &track->outbuf) { 4578 track->outputcounter = track->inputcounter; 4579 } else { 4580 track->outputcounter += track->outbuf.used - track_count_0; 4581 } 4582 4583 #if defined(AUDIO_DEBUG) 4584 if (audiodebug >= 3) { 4585 struct audio_track_debugbuf m; 4586 audio_track_bufstat(track, &m); 4587 TRACET(0, track, "end%s%s%s%s%s%s", 4588 m.outbuf, m.freq, m.chvol, m.chmix, m.codec, m.usrbuf); 4589 } 4590 #endif 4591 } 4592 4593 /* 4594 * Produce user output buffer for recording from input buffer. 4595 */ 4596 static void 4597 audio_track_record(audio_track_t *track) 4598 { 4599 audio_ring_t *outbuf; 4600 audio_ring_t *usrbuf; 4601 int count; 4602 int bytes; 4603 int framesize; 4604 4605 KASSERT(track); 4606 KASSERT(track->lock); 4607 4608 /* Number of frames to process */ 4609 count = auring_get_contig_used(track->input); 4610 count = uimin(count, track->mixer->frames_per_block); 4611 if (count == 0) { 4612 TRACET(4, track, "count == 0"); 4613 return; 4614 } 4615 4616 /* Frequency conversion */ 4617 if (track->freq.filter) { 4618 if (track->freq.srcbuf.used > 0) { 4619 audio_apply_stage(track, &track->freq, true); 4620 /* XXX should input of freq be from beginning of buf? */ 4621 } 4622 } 4623 4624 /* Channel mix */ 4625 if (track->chmix.filter) 4626 audio_apply_stage(track, &track->chmix, false); 4627 4628 /* Channel volume */ 4629 if (track->chvol.filter) 4630 audio_apply_stage(track, &track->chvol, false); 4631 4632 /* Encoding conversion */ 4633 if (track->codec.filter) 4634 audio_apply_stage(track, &track->codec, false); 4635 4636 /* Copy outbuf to usrbuf */ 4637 outbuf = &track->outbuf; 4638 usrbuf = &track->usrbuf; 4639 /* 4640 * framesize is always 1 byte or more since all formats supported 4641 * as usrfmt(=output) have 8bit or more stride. 4642 */ 4643 framesize = frametobyte(&outbuf->fmt, 1); 4644 KASSERT(framesize >= 1); 4645 /* 4646 * count is the number of frames to copy to usrbuf. 4647 * bytes is the number of bytes to copy to usrbuf. 4648 */ 4649 count = outbuf->used; 4650 count = uimin(count, 4651 (track->usrbuf_usedhigh - usrbuf->used) / framesize); 4652 bytes = count * framesize; 4653 if (auring_tail(usrbuf) + bytes < usrbuf->capacity) { 4654 memcpy((uint8_t *)usrbuf->mem + auring_tail(usrbuf), 4655 (uint8_t *)outbuf->mem + outbuf->head * framesize, 4656 bytes); 4657 auring_push(usrbuf, bytes); 4658 auring_take(outbuf, count); 4659 } else { 4660 int bytes1; 4661 int bytes2; 4662 4663 bytes1 = auring_get_contig_free(usrbuf); 4664 KASSERTMSG(bytes1 % framesize == 0, 4665 "bytes1=%d framesize=%d", bytes1, framesize); 4666 memcpy((uint8_t *)usrbuf->mem + auring_tail(usrbuf), 4667 (uint8_t *)outbuf->mem + outbuf->head * framesize, 4668 bytes1); 4669 auring_push(usrbuf, bytes1); 4670 auring_take(outbuf, bytes1 / framesize); 4671 4672 bytes2 = bytes - bytes1; 4673 memcpy((uint8_t *)usrbuf->mem + auring_tail(usrbuf), 4674 (uint8_t *)outbuf->mem + outbuf->head * framesize, 4675 bytes2); 4676 auring_push(usrbuf, bytes2); 4677 auring_take(outbuf, bytes2 / framesize); 4678 } 4679 4680 /* XXX TODO: any counters here? */ 4681 4682 #if defined(AUDIO_DEBUG) 4683 if (audiodebug >= 3) { 4684 struct audio_track_debugbuf m; 4685 audio_track_bufstat(track, &m); 4686 TRACET(0, track, "end%s%s%s%s%s%s", 4687 m.freq, m.chvol, m.chmix, m.codec, m.outbuf, m.usrbuf); 4688 } 4689 #endif 4690 } 4691 4692 /* 4693 * Calcurate blktime [msec] from mixer(.hwbuf.fmt). 4694 * Must be called with sc_lock held. 4695 */ 4696 static u_int 4697 audio_mixer_calc_blktime(struct audio_softc *sc, audio_trackmixer_t *mixer) 4698 { 4699 audio_format2_t *fmt; 4700 u_int blktime; 4701 u_int frames_per_block; 4702 4703 KASSERT(mutex_owned(sc->sc_lock)); 4704 4705 fmt = &mixer->hwbuf.fmt; 4706 blktime = sc->sc_blk_ms; 4707 4708 /* 4709 * If stride is not multiples of 8, special treatment is necessary. 4710 * For now, it is only x68k's vs(4), 4 bit/sample ADPCM. 4711 */ 4712 if (fmt->stride == 4) { 4713 frames_per_block = fmt->sample_rate * blktime / 1000; 4714 if ((frames_per_block & 1) != 0) 4715 blktime *= 2; 4716 } 4717 #ifdef DIAGNOSTIC 4718 else if (fmt->stride % NBBY != 0) { 4719 panic("unsupported HW stride %d", fmt->stride); 4720 } 4721 #endif 4722 4723 return blktime; 4724 } 4725 4726 /* 4727 * Initialize the mixer corresponding to the mode. 4728 * Set AUMODE_PLAY to the 'mode' for playback or AUMODE_RECORD for recording. 4729 * sc->sc_[pr]mixer (corresponding to the 'mode') must be zero-filled. 4730 * This function returns 0 on successful. Otherwise returns errno. 4731 * Must be called with sc_lock held. 4732 */ 4733 static int 4734 audio_mixer_init(struct audio_softc *sc, int mode, 4735 const audio_format2_t *hwfmt, const audio_filter_reg_t *reg) 4736 { 4737 char codecbuf[64]; 4738 audio_trackmixer_t *mixer; 4739 void (*softint_handler)(void *); 4740 int len; 4741 int blksize; 4742 int capacity; 4743 size_t bufsize; 4744 int hwblks; 4745 int blkms; 4746 int error; 4747 4748 KASSERT(hwfmt != NULL); 4749 KASSERT(reg != NULL); 4750 KASSERT(mutex_owned(sc->sc_lock)); 4751 4752 error = 0; 4753 if (mode == AUMODE_PLAY) 4754 mixer = sc->sc_pmixer; 4755 else 4756 mixer = sc->sc_rmixer; 4757 4758 mixer->sc = sc; 4759 mixer->mode = mode; 4760 4761 mixer->hwbuf.fmt = *hwfmt; 4762 mixer->volume = 256; 4763 mixer->blktime_d = 1000; 4764 mixer->blktime_n = audio_mixer_calc_blktime(sc, mixer); 4765 sc->sc_blk_ms = mixer->blktime_n; 4766 hwblks = NBLKHW; 4767 4768 mixer->frames_per_block = frame_per_block(mixer, &mixer->hwbuf.fmt); 4769 blksize = frametobyte(&mixer->hwbuf.fmt, mixer->frames_per_block); 4770 if (sc->hw_if->round_blocksize) { 4771 int rounded; 4772 audio_params_t p = format2_to_params(&mixer->hwbuf.fmt); 4773 rounded = sc->hw_if->round_blocksize(sc->hw_hdl, blksize, 4774 mode, &p); 4775 TRACE(1, "round_blocksize %d -> %d", blksize, rounded); 4776 if (rounded != blksize) { 4777 if ((rounded * NBBY) % (mixer->hwbuf.fmt.stride * 4778 mixer->hwbuf.fmt.channels) != 0) { 4779 device_printf(sc->sc_dev, 4780 "blksize not configured %d -> %d\n", 4781 blksize, rounded); 4782 return EINVAL; 4783 } 4784 /* Recalculation */ 4785 blksize = rounded; 4786 mixer->frames_per_block = blksize * NBBY / 4787 (mixer->hwbuf.fmt.stride * 4788 mixer->hwbuf.fmt.channels); 4789 } 4790 } 4791 mixer->blktime_n = mixer->frames_per_block; 4792 mixer->blktime_d = mixer->hwbuf.fmt.sample_rate; 4793 4794 capacity = mixer->frames_per_block * hwblks; 4795 bufsize = frametobyte(&mixer->hwbuf.fmt, capacity); 4796 if (sc->hw_if->round_buffersize) { 4797 size_t rounded; 4798 rounded = sc->hw_if->round_buffersize(sc->hw_hdl, mode, 4799 bufsize); 4800 TRACE(1, "round_buffersize %zd -> %zd", bufsize, rounded); 4801 if (rounded < bufsize) { 4802 /* buffersize needs NBLKHW blocks at least. */ 4803 device_printf(sc->sc_dev, 4804 "buffersize too small: buffersize=%zd blksize=%d\n", 4805 rounded, blksize); 4806 return EINVAL; 4807 } 4808 if (rounded % blksize != 0) { 4809 /* buffersize/blksize constraint mismatch? */ 4810 device_printf(sc->sc_dev, 4811 "buffersize must be multiple of blksize: " 4812 "buffersize=%zu blksize=%d\n", 4813 rounded, blksize); 4814 return EINVAL; 4815 } 4816 if (rounded != bufsize) { 4817 /* Recalcuration */ 4818 bufsize = rounded; 4819 hwblks = bufsize / blksize; 4820 capacity = mixer->frames_per_block * hwblks; 4821 } 4822 } 4823 TRACE(1, "buffersize for %s = %zu", 4824 (mode == AUMODE_PLAY) ? "playback" : "recording", 4825 bufsize); 4826 mixer->hwbuf.capacity = capacity; 4827 4828 /* 4829 * XXX need to release sc_lock for compatibility? 4830 */ 4831 if (sc->hw_if->allocm) { 4832 mixer->hwbuf.mem = sc->hw_if->allocm(sc->hw_hdl, mode, bufsize); 4833 if (mixer->hwbuf.mem == NULL) { 4834 device_printf(sc->sc_dev, "%s: allocm(%zu) failed\n", 4835 __func__, bufsize); 4836 return ENOMEM; 4837 } 4838 } else { 4839 mixer->hwbuf.mem = kmem_alloc(bufsize, KM_SLEEP); 4840 } 4841 4842 /* From here, audio_mixer_destroy is necessary to exit. */ 4843 if (mode == AUMODE_PLAY) { 4844 cv_init(&mixer->outcv, "audiowr"); 4845 } else { 4846 cv_init(&mixer->outcv, "audiord"); 4847 } 4848 4849 if (mode == AUMODE_PLAY) { 4850 softint_handler = audio_softintr_wr; 4851 } else { 4852 softint_handler = audio_softintr_rd; 4853 } 4854 mixer->sih = softint_establish(SOFTINT_SERIAL | SOFTINT_MPSAFE, 4855 softint_handler, sc); 4856 if (mixer->sih == NULL) { 4857 device_printf(sc->sc_dev, "softint_establish failed\n"); 4858 goto abort; 4859 } 4860 4861 mixer->track_fmt.encoding = AUDIO_ENCODING_SLINEAR_NE; 4862 mixer->track_fmt.precision = AUDIO_INTERNAL_BITS; 4863 mixer->track_fmt.stride = AUDIO_INTERNAL_BITS; 4864 mixer->track_fmt.channels = mixer->hwbuf.fmt.channels; 4865 mixer->track_fmt.sample_rate = mixer->hwbuf.fmt.sample_rate; 4866 4867 if (mixer->hwbuf.fmt.encoding == AUDIO_ENCODING_SLINEAR_OE && 4868 mixer->hwbuf.fmt.precision == AUDIO_INTERNAL_BITS) { 4869 mixer->swap_endian = true; 4870 TRACE(1, "swap_endian"); 4871 } 4872 4873 if (mode == AUMODE_PLAY) { 4874 /* Mixing buffer */ 4875 mixer->mixfmt = mixer->track_fmt; 4876 mixer->mixfmt.precision *= 2; 4877 mixer->mixfmt.stride *= 2; 4878 /* XXX TODO: use some macros? */ 4879 len = mixer->frames_per_block * mixer->mixfmt.channels * 4880 mixer->mixfmt.stride / NBBY; 4881 mixer->mixsample = audio_realloc(mixer->mixsample, len); 4882 } else { 4883 /* No mixing buffer for recording */ 4884 } 4885 4886 if (reg->codec) { 4887 mixer->codec = reg->codec; 4888 mixer->codecarg.context = reg->context; 4889 if (mode == AUMODE_PLAY) { 4890 mixer->codecarg.srcfmt = &mixer->track_fmt; 4891 mixer->codecarg.dstfmt = &mixer->hwbuf.fmt; 4892 } else { 4893 mixer->codecarg.srcfmt = &mixer->hwbuf.fmt; 4894 mixer->codecarg.dstfmt = &mixer->track_fmt; 4895 } 4896 mixer->codecbuf.fmt = mixer->track_fmt; 4897 mixer->codecbuf.capacity = mixer->frames_per_block; 4898 len = auring_bytelen(&mixer->codecbuf); 4899 mixer->codecbuf.mem = audio_realloc(mixer->codecbuf.mem, len); 4900 if (mixer->codecbuf.mem == NULL) { 4901 device_printf(sc->sc_dev, 4902 "%s: malloc codecbuf(%d) failed\n", 4903 __func__, len); 4904 error = ENOMEM; 4905 goto abort; 4906 } 4907 } 4908 4909 /* Succeeded so display it. */ 4910 codecbuf[0] = '\0'; 4911 if (mixer->codec || mixer->swap_endian) { 4912 snprintf(codecbuf, sizeof(codecbuf), " %s %s:%d", 4913 (mode == AUMODE_PLAY) ? "->" : "<-", 4914 audio_encoding_name(mixer->hwbuf.fmt.encoding), 4915 mixer->hwbuf.fmt.precision); 4916 } 4917 blkms = mixer->blktime_n * 1000 / mixer->blktime_d; 4918 aprint_normal_dev(sc->sc_dev, "%s:%d%s %dch %dHz, blk %dms for %s\n", 4919 audio_encoding_name(mixer->track_fmt.encoding), 4920 mixer->track_fmt.precision, 4921 codecbuf, 4922 mixer->track_fmt.channels, 4923 mixer->track_fmt.sample_rate, 4924 blkms, 4925 (mode == AUMODE_PLAY) ? "playback" : "recording"); 4926 4927 return 0; 4928 4929 abort: 4930 audio_mixer_destroy(sc, mixer); 4931 return error; 4932 } 4933 4934 /* 4935 * Releases all resources of 'mixer'. 4936 * Note that it does not release the memory area of 'mixer' itself. 4937 * Must be called with sc_lock held. 4938 */ 4939 static void 4940 audio_mixer_destroy(struct audio_softc *sc, audio_trackmixer_t *mixer) 4941 { 4942 int bufsize; 4943 4944 KASSERT(mutex_owned(sc->sc_lock)); 4945 4946 bufsize = frametobyte(&mixer->hwbuf.fmt, mixer->hwbuf.capacity); 4947 4948 if (mixer->hwbuf.mem != NULL) { 4949 if (sc->hw_if->freem) { 4950 sc->hw_if->freem(sc->hw_hdl, mixer->hwbuf.mem, bufsize); 4951 } else { 4952 kmem_free(mixer->hwbuf.mem, bufsize); 4953 } 4954 mixer->hwbuf.mem = NULL; 4955 } 4956 4957 audio_free(mixer->codecbuf.mem); 4958 audio_free(mixer->mixsample); 4959 4960 cv_destroy(&mixer->outcv); 4961 4962 if (mixer->sih) { 4963 softint_disestablish(mixer->sih); 4964 mixer->sih = NULL; 4965 } 4966 } 4967 4968 /* 4969 * Starts playback mixer. 4970 * Must be called only if sc_pbusy is false. 4971 * Must be called with sc_lock && sc_exlock held. 4972 * Must not be called from the interrupt context. 4973 */ 4974 static void 4975 audio_pmixer_start(struct audio_softc *sc, bool force) 4976 { 4977 audio_trackmixer_t *mixer; 4978 int minimum; 4979 4980 KASSERT(mutex_owned(sc->sc_lock)); 4981 KASSERT(sc->sc_exlock); 4982 KASSERT(sc->sc_pbusy == false); 4983 4984 mutex_enter(sc->sc_intr_lock); 4985 4986 mixer = sc->sc_pmixer; 4987 TRACE(2, "%smixseq=%d hwseq=%d hwbuf=%d/%d/%d%s", 4988 (audiodebug >= 3) ? "begin " : "", 4989 (int)mixer->mixseq, (int)mixer->hwseq, 4990 mixer->hwbuf.head, mixer->hwbuf.used, mixer->hwbuf.capacity, 4991 force ? " force" : ""); 4992 4993 /* Need two blocks to start normally. */ 4994 minimum = (force) ? 1 : 2; 4995 while (mixer->hwbuf.used < mixer->frames_per_block * minimum) { 4996 audio_pmixer_process(sc); 4997 } 4998 4999 /* Start output */ 5000 audio_pmixer_output(sc); 5001 sc->sc_pbusy = true; 5002 5003 TRACE(3, "end mixseq=%d hwseq=%d hwbuf=%d/%d/%d", 5004 (int)mixer->mixseq, (int)mixer->hwseq, 5005 mixer->hwbuf.head, mixer->hwbuf.used, mixer->hwbuf.capacity); 5006 5007 mutex_exit(sc->sc_intr_lock); 5008 } 5009 5010 /* 5011 * When playing back with MD filter: 5012 * 5013 * track track ... 5014 * v v 5015 * + mix (with aint2_t) 5016 * | master volume (with aint2_t) 5017 * v 5018 * mixsample [::::] wide-int 1 block (ring) buffer 5019 * | 5020 * | convert aint2_t -> aint_t 5021 * v 5022 * codecbuf [....] 1 block (ring) buffer 5023 * | 5024 * | convert to hw format 5025 * v 5026 * hwbuf [............] NBLKHW blocks ring buffer 5027 * 5028 * When playing back without MD filter: 5029 * 5030 * mixsample [::::] wide-int 1 block (ring) buffer 5031 * | 5032 * | convert aint2_t -> aint_t 5033 * | (with byte swap if necessary) 5034 * v 5035 * hwbuf [............] NBLKHW blocks ring buffer 5036 * 5037 * mixsample: slinear_NE, wide internal precision, HW ch, HW freq. 5038 * codecbuf: slinear_NE, internal precision, HW ch, HW freq. 5039 * hwbuf: HW encoding, HW precision, HW ch, HW freq. 5040 */ 5041 5042 /* 5043 * Performs track mixing and converts it to hwbuf. 5044 * Note that this function doesn't transfer hwbuf to hardware. 5045 * Must be called with sc_intr_lock held. 5046 */ 5047 static void 5048 audio_pmixer_process(struct audio_softc *sc) 5049 { 5050 audio_trackmixer_t *mixer; 5051 audio_file_t *f; 5052 int frame_count; 5053 int sample_count; 5054 int mixed; 5055 int i; 5056 aint2_t *m; 5057 aint_t *h; 5058 5059 mixer = sc->sc_pmixer; 5060 5061 frame_count = mixer->frames_per_block; 5062 KASSERTMSG(auring_get_contig_free(&mixer->hwbuf) >= frame_count, 5063 "auring_get_contig_free()=%d frame_count=%d", 5064 auring_get_contig_free(&mixer->hwbuf), frame_count); 5065 sample_count = frame_count * mixer->mixfmt.channels; 5066 5067 mixer->mixseq++; 5068 5069 /* Mix all tracks */ 5070 mixed = 0; 5071 SLIST_FOREACH(f, &sc->sc_files, entry) { 5072 audio_track_t *track = f->ptrack; 5073 5074 if (track == NULL) 5075 continue; 5076 5077 if (track->is_pause) { 5078 TRACET(4, track, "skip; paused"); 5079 continue; 5080 } 5081 5082 /* Skip if the track is used by process context. */ 5083 if (audio_track_lock_tryenter(track) == false) { 5084 TRACET(4, track, "skip; in use"); 5085 continue; 5086 } 5087 5088 /* Emulate mmap'ped track */ 5089 if (track->mmapped) { 5090 auring_push(&track->usrbuf, track->usrbuf_blksize); 5091 TRACET(4, track, "mmap; usr=%d/%d/C%d", 5092 track->usrbuf.head, 5093 track->usrbuf.used, 5094 track->usrbuf.capacity); 5095 } 5096 5097 if (track->outbuf.used < mixer->frames_per_block && 5098 track->usrbuf.used > 0) { 5099 TRACET(4, track, "process"); 5100 audio_track_play(track); 5101 } 5102 5103 if (track->outbuf.used > 0) { 5104 mixed = audio_pmixer_mix_track(mixer, track, mixed); 5105 } else { 5106 TRACET(4, track, "skip; empty"); 5107 } 5108 5109 audio_track_lock_exit(track); 5110 } 5111 5112 if (mixed == 0) { 5113 /* Silence */ 5114 memset(mixer->mixsample, 0, 5115 frametobyte(&mixer->mixfmt, frame_count)); 5116 } else { 5117 if (mixed > 1) { 5118 /* If there are multiple tracks, do auto gain control */ 5119 audio_pmixer_agc(mixer, sample_count); 5120 } 5121 5122 /* Apply master volume */ 5123 if (mixer->volume < 256) { 5124 m = mixer->mixsample; 5125 for (i = 0; i < sample_count; i++) { 5126 *m = AUDIO_SCALEDOWN(*m * mixer->volume, 8); 5127 m++; 5128 } 5129 5130 /* 5131 * Recover the volume gradually at the pace of 5132 * several times per second. If it's too fast, you 5133 * can recognize that the volume changes up and down 5134 * quickly and it's not so comfortable. 5135 */ 5136 mixer->voltimer += mixer->blktime_n; 5137 if (mixer->voltimer * 4 >= mixer->blktime_d) { 5138 mixer->volume++; 5139 mixer->voltimer = 0; 5140 #if defined(AUDIO_DEBUG_AGC) 5141 TRACE(1, "volume recover: %d", mixer->volume); 5142 #endif 5143 } 5144 } 5145 } 5146 5147 /* 5148 * The rest is the hardware part. 5149 */ 5150 5151 if (mixer->codec) { 5152 h = auring_tailptr_aint(&mixer->codecbuf); 5153 } else { 5154 h = auring_tailptr_aint(&mixer->hwbuf); 5155 } 5156 5157 m = mixer->mixsample; 5158 if (mixer->swap_endian) { 5159 for (i = 0; i < sample_count; i++) { 5160 *h++ = bswap16(*m++); 5161 } 5162 } else { 5163 for (i = 0; i < sample_count; i++) { 5164 *h++ = *m++; 5165 } 5166 } 5167 5168 /* Hardware driver's codec */ 5169 if (mixer->codec) { 5170 auring_push(&mixer->codecbuf, frame_count); 5171 mixer->codecarg.src = auring_headptr(&mixer->codecbuf); 5172 mixer->codecarg.dst = auring_tailptr(&mixer->hwbuf); 5173 mixer->codecarg.count = frame_count; 5174 mixer->codec(&mixer->codecarg); 5175 auring_take(&mixer->codecbuf, mixer->codecarg.count); 5176 } 5177 5178 auring_push(&mixer->hwbuf, frame_count); 5179 5180 TRACE(4, "done mixseq=%d hwbuf=%d/%d/%d%s", 5181 (int)mixer->mixseq, 5182 mixer->hwbuf.head, mixer->hwbuf.used, mixer->hwbuf.capacity, 5183 (mixed == 0) ? " silent" : ""); 5184 } 5185 5186 /* 5187 * Do auto gain control. 5188 * Must be called sc_intr_lock held. 5189 */ 5190 static void 5191 audio_pmixer_agc(audio_trackmixer_t *mixer, int sample_count) 5192 { 5193 struct audio_softc *sc __unused; 5194 aint2_t val; 5195 aint2_t maxval; 5196 aint2_t minval; 5197 aint2_t over_plus; 5198 aint2_t over_minus; 5199 aint2_t *m; 5200 int newvol; 5201 int i; 5202 5203 sc = mixer->sc; 5204 5205 /* Overflow detection */ 5206 maxval = AINT_T_MAX; 5207 minval = AINT_T_MIN; 5208 m = mixer->mixsample; 5209 for (i = 0; i < sample_count; i++) { 5210 val = *m++; 5211 if (val > maxval) 5212 maxval = val; 5213 else if (val < minval) 5214 minval = val; 5215 } 5216 5217 /* Absolute value of overflowed amount */ 5218 over_plus = maxval - AINT_T_MAX; 5219 over_minus = AINT_T_MIN - minval; 5220 5221 if (over_plus > 0 || over_minus > 0) { 5222 if (over_plus > over_minus) { 5223 newvol = (int)((aint2_t)AINT_T_MAX * 256 / maxval); 5224 } else { 5225 newvol = (int)((aint2_t)AINT_T_MIN * 256 / minval); 5226 } 5227 5228 /* 5229 * Change the volume only if new one is smaller. 5230 * Reset the timer even if the volume isn't changed. 5231 */ 5232 if (newvol <= mixer->volume) { 5233 mixer->volume = newvol; 5234 mixer->voltimer = 0; 5235 #if defined(AUDIO_DEBUG_AGC) 5236 TRACE(1, "auto volume adjust: %d", mixer->volume); 5237 #endif 5238 } 5239 } 5240 } 5241 5242 /* 5243 * Mix one track. 5244 * 'mixed' specifies the number of tracks mixed so far. 5245 * It returns the number of tracks mixed. In other words, it returns 5246 * mixed + 1 if this track is mixed. 5247 */ 5248 static int 5249 audio_pmixer_mix_track(audio_trackmixer_t *mixer, audio_track_t *track, 5250 int mixed) 5251 { 5252 int count; 5253 int sample_count; 5254 int remain; 5255 int i; 5256 const aint_t *s; 5257 aint2_t *d; 5258 5259 /* XXX TODO: Is this necessary for now? */ 5260 if (mixer->mixseq < track->seq) 5261 return mixed; 5262 5263 count = auring_get_contig_used(&track->outbuf); 5264 count = uimin(count, mixer->frames_per_block); 5265 5266 s = auring_headptr_aint(&track->outbuf); 5267 d = mixer->mixsample; 5268 5269 /* 5270 * Apply track volume with double-sized integer and perform 5271 * additive synthesis. 5272 * 5273 * XXX If you limit the track volume to 1.0 or less (<= 256), 5274 * it would be better to do this in the track conversion stage 5275 * rather than here. However, if you accept the volume to 5276 * be greater than 1.0 (> 256), it's better to do it here. 5277 * Because the operation here is done by double-sized integer. 5278 */ 5279 sample_count = count * mixer->mixfmt.channels; 5280 if (mixed == 0) { 5281 /* If this is the first track, assignment can be used. */ 5282 #if defined(AUDIO_SUPPORT_TRACK_VOLUME) 5283 if (track->volume != 256) { 5284 for (i = 0; i < sample_count; i++) { 5285 aint2_t v; 5286 v = *s++; 5287 *d++ = AUDIO_SCALEDOWN(v * track->volume, 8) 5288 } 5289 } else 5290 #endif 5291 { 5292 for (i = 0; i < sample_count; i++) { 5293 *d++ = ((aint2_t)*s++); 5294 } 5295 } 5296 /* Fill silence if the first track is not filled. */ 5297 for (; i < mixer->frames_per_block * mixer->mixfmt.channels; i++) 5298 *d++ = 0; 5299 } else { 5300 /* If this is the second or later, add it. */ 5301 #if defined(AUDIO_SUPPORT_TRACK_VOLUME) 5302 if (track->volume != 256) { 5303 for (i = 0; i < sample_count; i++) { 5304 aint2_t v; 5305 v = *s++; 5306 *d++ += AUDIO_SCALEDOWN(v * track->volume, 8); 5307 } 5308 } else 5309 #endif 5310 { 5311 for (i = 0; i < sample_count; i++) { 5312 *d++ += ((aint2_t)*s++); 5313 } 5314 } 5315 } 5316 5317 auring_take(&track->outbuf, count); 5318 /* 5319 * The counters have to align block even if outbuf is less than 5320 * one block. XXX Is this still necessary? 5321 */ 5322 remain = mixer->frames_per_block - count; 5323 if (__predict_false(remain != 0)) { 5324 auring_push(&track->outbuf, remain); 5325 auring_take(&track->outbuf, remain); 5326 } 5327 5328 /* 5329 * Update track sequence. 5330 * mixseq has previous value yet at this point. 5331 */ 5332 track->seq = mixer->mixseq + 1; 5333 5334 return mixed + 1; 5335 } 5336 5337 /* 5338 * Output one block from hwbuf to HW. 5339 * Must be called with sc_intr_lock held. 5340 */ 5341 static void 5342 audio_pmixer_output(struct audio_softc *sc) 5343 { 5344 audio_trackmixer_t *mixer; 5345 audio_params_t params; 5346 void *start; 5347 void *end; 5348 int blksize; 5349 int error; 5350 5351 mixer = sc->sc_pmixer; 5352 TRACE(4, "pbusy=%d hwbuf=%d/%d/%d", 5353 sc->sc_pbusy, 5354 mixer->hwbuf.head, mixer->hwbuf.used, mixer->hwbuf.capacity); 5355 KASSERTMSG(mixer->hwbuf.used >= mixer->frames_per_block, 5356 "mixer->hwbuf.used=%d mixer->frames_per_block=%d", 5357 mixer->hwbuf.used, mixer->frames_per_block); 5358 5359 blksize = frametobyte(&mixer->hwbuf.fmt, mixer->frames_per_block); 5360 5361 if (sc->hw_if->trigger_output) { 5362 /* trigger (at once) */ 5363 if (!sc->sc_pbusy) { 5364 start = mixer->hwbuf.mem; 5365 end = (uint8_t *)start + auring_bytelen(&mixer->hwbuf); 5366 params = format2_to_params(&mixer->hwbuf.fmt); 5367 5368 error = sc->hw_if->trigger_output(sc->hw_hdl, 5369 start, end, blksize, audio_pintr, sc, ¶ms); 5370 if (error) { 5371 device_printf(sc->sc_dev, 5372 "trigger_output failed with %d\n", error); 5373 return; 5374 } 5375 } 5376 } else { 5377 /* start (everytime) */ 5378 start = auring_headptr(&mixer->hwbuf); 5379 5380 error = sc->hw_if->start_output(sc->hw_hdl, 5381 start, blksize, audio_pintr, sc); 5382 if (error) { 5383 device_printf(sc->sc_dev, 5384 "start_output failed with %d\n", error); 5385 return; 5386 } 5387 } 5388 } 5389 5390 /* 5391 * This is an interrupt handler for playback. 5392 * It is called with sc_intr_lock held. 5393 * 5394 * It is usually called from hardware interrupt. However, note that 5395 * for some drivers (e.g. uaudio) it is called from software interrupt. 5396 */ 5397 static void 5398 audio_pintr(void *arg) 5399 { 5400 struct audio_softc *sc; 5401 audio_trackmixer_t *mixer; 5402 5403 sc = arg; 5404 KASSERT(mutex_owned(sc->sc_intr_lock)); 5405 5406 if (sc->sc_dying) 5407 return; 5408 if (sc->sc_pbusy == false) { 5409 #if defined(DIAGNOSTIC) 5410 device_printf(sc->sc_dev, "stray interrupt\n"); 5411 #endif 5412 return; 5413 } 5414 5415 mixer = sc->sc_pmixer; 5416 mixer->hw_complete_counter += mixer->frames_per_block; 5417 mixer->hwseq++; 5418 5419 auring_take(&mixer->hwbuf, mixer->frames_per_block); 5420 5421 TRACE(4, 5422 "HW_INT ++hwseq=%" PRIu64 " cmplcnt=%" PRIu64 " hwbuf=%d/%d/%d", 5423 mixer->hwseq, mixer->hw_complete_counter, 5424 mixer->hwbuf.head, mixer->hwbuf.used, mixer->hwbuf.capacity); 5425 5426 #if defined(AUDIO_HW_SINGLE_BUFFER) 5427 /* 5428 * Create a new block here and output it immediately. 5429 * It makes a latency lower but needs machine power. 5430 */ 5431 audio_pmixer_process(sc); 5432 audio_pmixer_output(sc); 5433 #else 5434 /* 5435 * It is called when block N output is done. 5436 * Output immediately block N+1 created by the last interrupt. 5437 * And then create block N+2 for the next interrupt. 5438 * This method makes playback robust even on slower machines. 5439 * Instead the latency is increased by one block. 5440 */ 5441 5442 /* At first, output ready block. */ 5443 if (mixer->hwbuf.used >= mixer->frames_per_block) { 5444 audio_pmixer_output(sc); 5445 } 5446 5447 bool later = false; 5448 5449 if (mixer->hwbuf.used < mixer->frames_per_block) { 5450 later = true; 5451 } 5452 5453 /* Then, process next block. */ 5454 audio_pmixer_process(sc); 5455 5456 if (later) { 5457 audio_pmixer_output(sc); 5458 } 5459 #endif 5460 5461 /* 5462 * When this interrupt is the real hardware interrupt, disabling 5463 * preemption here is not necessary. But some drivers (e.g. uaudio) 5464 * emulate it by software interrupt, so kpreempt_disable is necessary. 5465 */ 5466 kpreempt_disable(); 5467 softint_schedule(mixer->sih); 5468 kpreempt_enable(); 5469 } 5470 5471 /* 5472 * Starts record mixer. 5473 * Must be called only if sc_rbusy is false. 5474 * Must be called with sc_lock && sc_exlock held. 5475 * Must not be called from the interrupt context. 5476 */ 5477 static void 5478 audio_rmixer_start(struct audio_softc *sc) 5479 { 5480 5481 KASSERT(mutex_owned(sc->sc_lock)); 5482 KASSERT(sc->sc_exlock); 5483 KASSERT(sc->sc_rbusy == false); 5484 5485 mutex_enter(sc->sc_intr_lock); 5486 5487 TRACE(2, "%s", (audiodebug >= 3) ? "begin" : ""); 5488 audio_rmixer_input(sc); 5489 sc->sc_rbusy = true; 5490 TRACE(3, "end"); 5491 5492 mutex_exit(sc->sc_intr_lock); 5493 } 5494 5495 /* 5496 * When recording with MD filter: 5497 * 5498 * hwbuf [............] NBLKHW blocks ring buffer 5499 * | 5500 * | convert from hw format 5501 * v 5502 * codecbuf [....] 1 block (ring) buffer 5503 * | | 5504 * v v 5505 * track track ... 5506 * 5507 * When recording without MD filter: 5508 * 5509 * hwbuf [............] NBLKHW blocks ring buffer 5510 * | | 5511 * v v 5512 * track track ... 5513 * 5514 * hwbuf: HW encoding, HW precision, HW ch, HW freq. 5515 * codecbuf: slinear_NE, internal precision, HW ch, HW freq. 5516 */ 5517 5518 /* 5519 * Distribute a recorded block to all recording tracks. 5520 */ 5521 static void 5522 audio_rmixer_process(struct audio_softc *sc) 5523 { 5524 audio_trackmixer_t *mixer; 5525 audio_ring_t *mixersrc; 5526 audio_file_t *f; 5527 aint_t *p; 5528 int count; 5529 int bytes; 5530 int i; 5531 5532 mixer = sc->sc_rmixer; 5533 5534 /* 5535 * count is the number of frames to be retrieved this time. 5536 * count should be one block. 5537 */ 5538 count = auring_get_contig_used(&mixer->hwbuf); 5539 count = uimin(count, mixer->frames_per_block); 5540 if (count <= 0) { 5541 TRACE(4, "count %d: too short", count); 5542 return; 5543 } 5544 bytes = frametobyte(&mixer->track_fmt, count); 5545 5546 /* Hardware driver's codec */ 5547 if (mixer->codec) { 5548 mixer->codecarg.src = auring_headptr(&mixer->hwbuf); 5549 mixer->codecarg.dst = auring_tailptr(&mixer->codecbuf); 5550 mixer->codecarg.count = count; 5551 mixer->codec(&mixer->codecarg); 5552 auring_take(&mixer->hwbuf, mixer->codecarg.count); 5553 auring_push(&mixer->codecbuf, mixer->codecarg.count); 5554 mixersrc = &mixer->codecbuf; 5555 } else { 5556 mixersrc = &mixer->hwbuf; 5557 } 5558 5559 if (mixer->swap_endian) { 5560 /* inplace conversion */ 5561 p = auring_headptr_aint(mixersrc); 5562 for (i = 0; i < count * mixer->track_fmt.channels; i++, p++) { 5563 *p = bswap16(*p); 5564 } 5565 } 5566 5567 /* Distribute to all tracks. */ 5568 SLIST_FOREACH(f, &sc->sc_files, entry) { 5569 audio_track_t *track = f->rtrack; 5570 audio_ring_t *input; 5571 5572 if (track == NULL) 5573 continue; 5574 5575 if (track->is_pause) { 5576 TRACET(4, track, "skip; paused"); 5577 continue; 5578 } 5579 5580 if (audio_track_lock_tryenter(track) == false) { 5581 TRACET(4, track, "skip; in use"); 5582 continue; 5583 } 5584 5585 /* If the track buffer is full, discard the oldest one? */ 5586 input = track->input; 5587 if (input->capacity - input->used < mixer->frames_per_block) { 5588 int drops = mixer->frames_per_block - 5589 (input->capacity - input->used); 5590 track->dropframes += drops; 5591 TRACET(4, track, "drop %d frames: inp=%d/%d/%d", 5592 drops, 5593 input->head, input->used, input->capacity); 5594 auring_take(input, drops); 5595 } 5596 KASSERTMSG(input->used % mixer->frames_per_block == 0, 5597 "input->used=%d mixer->frames_per_block=%d", 5598 input->used, mixer->frames_per_block); 5599 5600 memcpy(auring_tailptr_aint(input), 5601 auring_headptr_aint(mixersrc), 5602 bytes); 5603 auring_push(input, count); 5604 5605 /* XXX sequence counter? */ 5606 5607 audio_track_lock_exit(track); 5608 } 5609 5610 auring_take(mixersrc, count); 5611 } 5612 5613 /* 5614 * Input one block from HW to hwbuf. 5615 * Must be called with sc_intr_lock held. 5616 */ 5617 static void 5618 audio_rmixer_input(struct audio_softc *sc) 5619 { 5620 audio_trackmixer_t *mixer; 5621 audio_params_t params; 5622 void *start; 5623 void *end; 5624 int blksize; 5625 int error; 5626 5627 mixer = sc->sc_rmixer; 5628 blksize = frametobyte(&mixer->hwbuf.fmt, mixer->frames_per_block); 5629 5630 if (sc->hw_if->trigger_input) { 5631 /* trigger (at once) */ 5632 if (!sc->sc_rbusy) { 5633 start = mixer->hwbuf.mem; 5634 end = (uint8_t *)start + auring_bytelen(&mixer->hwbuf); 5635 params = format2_to_params(&mixer->hwbuf.fmt); 5636 5637 error = sc->hw_if->trigger_input(sc->hw_hdl, 5638 start, end, blksize, audio_rintr, sc, ¶ms); 5639 if (error) { 5640 device_printf(sc->sc_dev, 5641 "trigger_input failed with %d\n", error); 5642 return; 5643 } 5644 } 5645 } else { 5646 /* start (everytime) */ 5647 start = auring_tailptr(&mixer->hwbuf); 5648 5649 error = sc->hw_if->start_input(sc->hw_hdl, 5650 start, blksize, audio_rintr, sc); 5651 if (error) { 5652 device_printf(sc->sc_dev, 5653 "start_input failed with %d\n", error); 5654 return; 5655 } 5656 } 5657 } 5658 5659 /* 5660 * This is an interrupt handler for recording. 5661 * It is called with sc_intr_lock. 5662 * 5663 * It is usually called from hardware interrupt. However, note that 5664 * for some drivers (e.g. uaudio) it is called from software interrupt. 5665 */ 5666 static void 5667 audio_rintr(void *arg) 5668 { 5669 struct audio_softc *sc; 5670 audio_trackmixer_t *mixer; 5671 5672 sc = arg; 5673 KASSERT(mutex_owned(sc->sc_intr_lock)); 5674 5675 if (sc->sc_dying) 5676 return; 5677 if (sc->sc_rbusy == false) { 5678 #if defined(DIAGNOSTIC) 5679 device_printf(sc->sc_dev, "stray interrupt\n"); 5680 #endif 5681 return; 5682 } 5683 5684 mixer = sc->sc_rmixer; 5685 mixer->hw_complete_counter += mixer->frames_per_block; 5686 mixer->hwseq++; 5687 5688 auring_push(&mixer->hwbuf, mixer->frames_per_block); 5689 5690 TRACE(4, 5691 "HW_INT ++hwseq=%" PRIu64 " cmplcnt=%" PRIu64 " hwbuf=%d/%d/%d", 5692 mixer->hwseq, mixer->hw_complete_counter, 5693 mixer->hwbuf.head, mixer->hwbuf.used, mixer->hwbuf.capacity); 5694 5695 /* Distrubute recorded block */ 5696 audio_rmixer_process(sc); 5697 5698 /* Request next block */ 5699 audio_rmixer_input(sc); 5700 5701 /* 5702 * When this interrupt is the real hardware interrupt, disabling 5703 * preemption here is not necessary. But some drivers (e.g. uaudio) 5704 * emulate it by software interrupt, so kpreempt_disable is necessary. 5705 */ 5706 kpreempt_disable(); 5707 softint_schedule(mixer->sih); 5708 kpreempt_enable(); 5709 } 5710 5711 /* 5712 * Halts playback mixer. 5713 * This function also clears related parameters, so call this function 5714 * instead of calling halt_output directly. 5715 * Must be called only if sc_pbusy is true. 5716 * Must be called with sc_lock && sc_exlock held. 5717 */ 5718 static int 5719 audio_pmixer_halt(struct audio_softc *sc) 5720 { 5721 int error; 5722 5723 TRACE(2, ""); 5724 KASSERT(mutex_owned(sc->sc_lock)); 5725 KASSERT(sc->sc_exlock); 5726 5727 mutex_enter(sc->sc_intr_lock); 5728 error = sc->hw_if->halt_output(sc->hw_hdl); 5729 5730 /* Halts anyway even if some error has occurred. */ 5731 sc->sc_pbusy = false; 5732 sc->sc_pmixer->hwbuf.head = 0; 5733 sc->sc_pmixer->hwbuf.used = 0; 5734 sc->sc_pmixer->mixseq = 0; 5735 sc->sc_pmixer->hwseq = 0; 5736 mutex_exit(sc->sc_intr_lock); 5737 5738 return error; 5739 } 5740 5741 /* 5742 * Halts recording mixer. 5743 * This function also clears related parameters, so call this function 5744 * instead of calling halt_input directly. 5745 * Must be called only if sc_rbusy is true. 5746 * Must be called with sc_lock && sc_exlock held. 5747 */ 5748 static int 5749 audio_rmixer_halt(struct audio_softc *sc) 5750 { 5751 int error; 5752 5753 TRACE(2, ""); 5754 KASSERT(mutex_owned(sc->sc_lock)); 5755 KASSERT(sc->sc_exlock); 5756 5757 mutex_enter(sc->sc_intr_lock); 5758 error = sc->hw_if->halt_input(sc->hw_hdl); 5759 5760 /* Halts anyway even if some error has occurred. */ 5761 sc->sc_rbusy = false; 5762 sc->sc_rmixer->hwbuf.head = 0; 5763 sc->sc_rmixer->hwbuf.used = 0; 5764 sc->sc_rmixer->mixseq = 0; 5765 sc->sc_rmixer->hwseq = 0; 5766 mutex_exit(sc->sc_intr_lock); 5767 5768 return error; 5769 } 5770 5771 /* 5772 * Flush this track. 5773 * Halts all operations, clears all buffers, reset error counters. 5774 * XXX I'm not sure... 5775 */ 5776 static void 5777 audio_track_clear(struct audio_softc *sc, audio_track_t *track) 5778 { 5779 5780 KASSERT(track); 5781 TRACET(3, track, "clear"); 5782 5783 audio_track_lock_enter(track); 5784 5785 track->usrbuf.used = 0; 5786 /* Clear all internal parameters. */ 5787 if (track->codec.filter) { 5788 track->codec.srcbuf.used = 0; 5789 track->codec.srcbuf.head = 0; 5790 } 5791 if (track->chvol.filter) { 5792 track->chvol.srcbuf.used = 0; 5793 track->chvol.srcbuf.head = 0; 5794 } 5795 if (track->chmix.filter) { 5796 track->chmix.srcbuf.used = 0; 5797 track->chmix.srcbuf.head = 0; 5798 } 5799 if (track->freq.filter) { 5800 track->freq.srcbuf.used = 0; 5801 track->freq.srcbuf.head = 0; 5802 if (track->freq_step < 65536) 5803 track->freq_current = 65536; 5804 else 5805 track->freq_current = 0; 5806 memset(track->freq_prev, 0, sizeof(track->freq_prev)); 5807 memset(track->freq_curr, 0, sizeof(track->freq_curr)); 5808 } 5809 /* Clear buffer, then operation halts naturally. */ 5810 track->outbuf.used = 0; 5811 5812 /* Clear counters. */ 5813 track->dropframes = 0; 5814 5815 audio_track_lock_exit(track); 5816 } 5817 5818 /* 5819 * Drain the track. 5820 * track must be present and for playback. 5821 * If successful, it returns 0. Otherwise returns errno. 5822 * Must be called with sc_lock held. 5823 */ 5824 static int 5825 audio_track_drain(struct audio_softc *sc, audio_track_t *track) 5826 { 5827 audio_trackmixer_t *mixer; 5828 int done; 5829 int error; 5830 5831 KASSERT(track); 5832 TRACET(3, track, "start"); 5833 mixer = track->mixer; 5834 KASSERT(mutex_owned(sc->sc_lock)); 5835 5836 /* Ignore them if pause. */ 5837 if (track->is_pause) { 5838 TRACET(3, track, "pause -> clear"); 5839 track->pstate = AUDIO_STATE_CLEAR; 5840 } 5841 /* Terminate early here if there is no data in the track. */ 5842 if (track->pstate == AUDIO_STATE_CLEAR) { 5843 TRACET(3, track, "no need to drain"); 5844 return 0; 5845 } 5846 track->pstate = AUDIO_STATE_DRAINING; 5847 5848 for (;;) { 5849 /* I want to display it before condition evaluation. */ 5850 TRACET(3, track, "pid=%d.%d trkseq=%d hwseq=%d out=%d/%d/%d", 5851 (int)curproc->p_pid, (int)curlwp->l_lid, 5852 (int)track->seq, (int)mixer->hwseq, 5853 track->outbuf.head, track->outbuf.used, 5854 track->outbuf.capacity); 5855 5856 /* Condition to terminate */ 5857 audio_track_lock_enter(track); 5858 done = (track->usrbuf.used < frametobyte(&track->inputfmt, 1) && 5859 track->outbuf.used == 0 && 5860 track->seq <= mixer->hwseq); 5861 audio_track_lock_exit(track); 5862 if (done) 5863 break; 5864 5865 TRACET(3, track, "sleep"); 5866 error = audio_track_waitio(sc, track); 5867 if (error) 5868 return error; 5869 5870 /* XXX call audio_track_play here ? */ 5871 } 5872 5873 track->pstate = AUDIO_STATE_CLEAR; 5874 TRACET(3, track, "done trk_inp=%d trk_out=%d", 5875 (int)track->inputcounter, (int)track->outputcounter); 5876 return 0; 5877 } 5878 5879 /* 5880 * Send signal to process. 5881 * This is intended to be called only from audio_softintr_{rd,wr}. 5882 * Must be called with sc_lock && sc_intr_lock held. 5883 */ 5884 static inline void 5885 audio_psignal(struct audio_softc *sc, pid_t pid, int signum) 5886 { 5887 proc_t *p; 5888 5889 KASSERT(mutex_owned(sc->sc_lock)); 5890 KASSERT(mutex_owned(sc->sc_intr_lock)); 5891 KASSERT(pid != 0); 5892 5893 /* 5894 * psignal() must be called without spin lock held. 5895 * So leave intr_lock temporarily here. 5896 */ 5897 mutex_exit(sc->sc_intr_lock); 5898 5899 mutex_enter(proc_lock); 5900 p = proc_find(pid); 5901 if (p) 5902 psignal(p, signum); 5903 mutex_exit(proc_lock); 5904 5905 /* Enter intr_lock again */ 5906 mutex_enter(sc->sc_intr_lock); 5907 } 5908 5909 /* 5910 * This is software interrupt handler for record. 5911 * It is called from recording hardware interrupt everytime. 5912 * It does: 5913 * - Deliver SIGIO for all async processes. 5914 * - Notify to audio_read() that data has arrived. 5915 * - selnotify() for select/poll-ing processes. 5916 */ 5917 /* 5918 * XXX If a process issues FIOASYNC between hardware interrupt and 5919 * software interrupt, (stray) SIGIO will be sent to the process 5920 * despite the fact that it has not receive recorded data yet. 5921 */ 5922 static void 5923 audio_softintr_rd(void *cookie) 5924 { 5925 struct audio_softc *sc = cookie; 5926 audio_file_t *f; 5927 pid_t pid; 5928 5929 mutex_enter(sc->sc_lock); 5930 mutex_enter(sc->sc_intr_lock); 5931 5932 SLIST_FOREACH(f, &sc->sc_files, entry) { 5933 audio_track_t *track = f->rtrack; 5934 5935 if (track == NULL) 5936 continue; 5937 5938 TRACET(4, track, "broadcast; inp=%d/%d/%d", 5939 track->input->head, 5940 track->input->used, 5941 track->input->capacity); 5942 5943 pid = f->async_audio; 5944 if (pid != 0) { 5945 TRACEF(4, f, "sending SIGIO %d", pid); 5946 audio_psignal(sc, pid, SIGIO); 5947 } 5948 } 5949 mutex_exit(sc->sc_intr_lock); 5950 5951 /* Notify that data has arrived. */ 5952 selnotify(&sc->sc_rsel, 0, NOTE_SUBMIT); 5953 KNOTE(&sc->sc_rsel.sel_klist, 0); 5954 cv_broadcast(&sc->sc_rmixer->outcv); 5955 5956 mutex_exit(sc->sc_lock); 5957 } 5958 5959 /* 5960 * This is software interrupt handler for playback. 5961 * It is called from playback hardware interrupt everytime. 5962 * It does: 5963 * - Deliver SIGIO for all async and writable (used < lowat) processes. 5964 * - Notify to audio_write() that outbuf block available. 5965 * - selnotify() for select/poll-ing processes if there are any writable 5966 * (used < lowat) processes. Checking each descriptor will be done by 5967 * filt_audiowrite_event(). 5968 */ 5969 static void 5970 audio_softintr_wr(void *cookie) 5971 { 5972 struct audio_softc *sc = cookie; 5973 audio_file_t *f; 5974 bool found; 5975 pid_t pid; 5976 5977 TRACE(4, "called"); 5978 found = false; 5979 5980 mutex_enter(sc->sc_lock); 5981 mutex_enter(sc->sc_intr_lock); 5982 5983 SLIST_FOREACH(f, &sc->sc_files, entry) { 5984 audio_track_t *track = f->ptrack; 5985 5986 if (track == NULL) 5987 continue; 5988 5989 TRACET(4, track, "broadcast; trseq=%d out=%d/%d/%d", 5990 (int)track->seq, 5991 track->outbuf.head, 5992 track->outbuf.used, 5993 track->outbuf.capacity); 5994 5995 /* 5996 * Send a signal if the process is async mode and 5997 * used is lower than lowat. 5998 */ 5999 if (track->usrbuf.used <= track->usrbuf_usedlow && 6000 !track->is_pause) { 6001 /* For selnotify */ 6002 found = true; 6003 /* For SIGIO */ 6004 pid = f->async_audio; 6005 if (pid != 0) { 6006 TRACEF(4, f, "sending SIGIO %d", pid); 6007 audio_psignal(sc, pid, SIGIO); 6008 } 6009 } 6010 } 6011 mutex_exit(sc->sc_intr_lock); 6012 6013 /* 6014 * Notify for select/poll when someone become writable. 6015 * It needs sc_lock (and not sc_intr_lock). 6016 */ 6017 if (found) { 6018 TRACE(4, "selnotify"); 6019 selnotify(&sc->sc_wsel, 0, NOTE_SUBMIT); 6020 KNOTE(&sc->sc_wsel.sel_klist, 0); 6021 } 6022 6023 /* Notify to audio_write() that outbuf available. */ 6024 cv_broadcast(&sc->sc_pmixer->outcv); 6025 6026 mutex_exit(sc->sc_lock); 6027 } 6028 6029 /* 6030 * Check (and convert) the format *p came from userland. 6031 * If successful, it writes back the converted format to *p if necessary 6032 * and returns 0. Otherwise returns errno (*p may change even this case). 6033 */ 6034 static int 6035 audio_check_params(audio_format2_t *p) 6036 { 6037 6038 /* Convert obsoleted AUDIO_ENCODING_PCM* */ 6039 /* XXX Is this conversion right? */ 6040 if (p->encoding == AUDIO_ENCODING_PCM16) { 6041 if (p->precision == 8) 6042 p->encoding = AUDIO_ENCODING_ULINEAR; 6043 else 6044 p->encoding = AUDIO_ENCODING_SLINEAR; 6045 } else if (p->encoding == AUDIO_ENCODING_PCM8) { 6046 if (p->precision == 8) 6047 p->encoding = AUDIO_ENCODING_ULINEAR; 6048 else 6049 return EINVAL; 6050 } 6051 6052 /* 6053 * Convert obsoleted AUDIO_ENCODING_[SU]LINEAR without endianness 6054 * suffix. 6055 */ 6056 if (p->encoding == AUDIO_ENCODING_SLINEAR) 6057 p->encoding = AUDIO_ENCODING_SLINEAR_NE; 6058 if (p->encoding == AUDIO_ENCODING_ULINEAR) 6059 p->encoding = AUDIO_ENCODING_ULINEAR_NE; 6060 6061 switch (p->encoding) { 6062 case AUDIO_ENCODING_ULAW: 6063 case AUDIO_ENCODING_ALAW: 6064 if (p->precision != 8) 6065 return EINVAL; 6066 break; 6067 case AUDIO_ENCODING_ADPCM: 6068 if (p->precision != 4 && p->precision != 8) 6069 return EINVAL; 6070 break; 6071 case AUDIO_ENCODING_SLINEAR_LE: 6072 case AUDIO_ENCODING_SLINEAR_BE: 6073 case AUDIO_ENCODING_ULINEAR_LE: 6074 case AUDIO_ENCODING_ULINEAR_BE: 6075 if (p->precision != 8 && p->precision != 16 && 6076 p->precision != 24 && p->precision != 32) 6077 return EINVAL; 6078 6079 /* 8bit format does not have endianness. */ 6080 if (p->precision == 8) { 6081 if (p->encoding == AUDIO_ENCODING_SLINEAR_OE) 6082 p->encoding = AUDIO_ENCODING_SLINEAR_NE; 6083 if (p->encoding == AUDIO_ENCODING_ULINEAR_OE) 6084 p->encoding = AUDIO_ENCODING_ULINEAR_NE; 6085 } 6086 6087 if (p->precision > p->stride) 6088 return EINVAL; 6089 break; 6090 case AUDIO_ENCODING_MPEG_L1_STREAM: 6091 case AUDIO_ENCODING_MPEG_L1_PACKETS: 6092 case AUDIO_ENCODING_MPEG_L1_SYSTEM: 6093 case AUDIO_ENCODING_MPEG_L2_STREAM: 6094 case AUDIO_ENCODING_MPEG_L2_PACKETS: 6095 case AUDIO_ENCODING_MPEG_L2_SYSTEM: 6096 case AUDIO_ENCODING_AC3: 6097 break; 6098 default: 6099 return EINVAL; 6100 } 6101 6102 /* sanity check # of channels*/ 6103 if (p->channels < 1 || p->channels > AUDIO_MAX_CHANNELS) 6104 return EINVAL; 6105 6106 return 0; 6107 } 6108 6109 /* 6110 * Initialize playback and record mixers. 6111 * mode (AUMODE_{PLAY,RECORD}) indicates the mixer to be initialized. 6112 * phwfmt and rhwfmt indicate the hardware format. pfil and rfil indicate 6113 * the filter registration information. These four must not be NULL. 6114 * If successful returns 0. Otherwise returns errno. 6115 * Must be called with sc_lock held. 6116 * Must not be called if there are any tracks. 6117 * Caller should check that the initialization succeed by whether 6118 * sc_[pr]mixer is not NULL. 6119 */ 6120 static int 6121 audio_mixers_init(struct audio_softc *sc, int mode, 6122 const audio_format2_t *phwfmt, const audio_format2_t *rhwfmt, 6123 const audio_filter_reg_t *pfil, const audio_filter_reg_t *rfil) 6124 { 6125 int error; 6126 6127 KASSERT(phwfmt != NULL); 6128 KASSERT(rhwfmt != NULL); 6129 KASSERT(pfil != NULL); 6130 KASSERT(rfil != NULL); 6131 KASSERT(mutex_owned(sc->sc_lock)); 6132 6133 if ((mode & AUMODE_PLAY)) { 6134 if (sc->sc_pmixer == NULL) { 6135 sc->sc_pmixer = kmem_zalloc(sizeof(*sc->sc_pmixer), 6136 KM_SLEEP); 6137 } else { 6138 /* destroy() doesn't free memory. */ 6139 audio_mixer_destroy(sc, sc->sc_pmixer); 6140 memset(sc->sc_pmixer, 0, sizeof(*sc->sc_pmixer)); 6141 } 6142 error = audio_mixer_init(sc, AUMODE_PLAY, phwfmt, pfil); 6143 if (error) { 6144 device_printf(sc->sc_dev, 6145 "configuring playback mode failed with %d\n", 6146 error); 6147 kmem_free(sc->sc_pmixer, sizeof(*sc->sc_pmixer)); 6148 sc->sc_pmixer = NULL; 6149 return error; 6150 } 6151 } 6152 if ((mode & AUMODE_RECORD)) { 6153 if (sc->sc_rmixer == NULL) { 6154 sc->sc_rmixer = kmem_zalloc(sizeof(*sc->sc_rmixer), 6155 KM_SLEEP); 6156 } else { 6157 /* destroy() doesn't free memory. */ 6158 audio_mixer_destroy(sc, sc->sc_rmixer); 6159 memset(sc->sc_rmixer, 0, sizeof(*sc->sc_rmixer)); 6160 } 6161 error = audio_mixer_init(sc, AUMODE_RECORD, rhwfmt, rfil); 6162 if (error) { 6163 device_printf(sc->sc_dev, 6164 "configuring record mode failed with %d\n", 6165 error); 6166 kmem_free(sc->sc_rmixer, sizeof(*sc->sc_rmixer)); 6167 sc->sc_rmixer = NULL; 6168 return error; 6169 } 6170 } 6171 6172 return 0; 6173 } 6174 6175 /* 6176 * Select a frequency. 6177 * Prioritize 48kHz and 44.1kHz. Otherwise choose the highest one. 6178 * XXX Better algorithm? 6179 */ 6180 static int 6181 audio_select_freq(const struct audio_format *fmt) 6182 { 6183 int freq; 6184 int high; 6185 int low; 6186 int j; 6187 6188 if (fmt->frequency_type == 0) { 6189 low = fmt->frequency[0]; 6190 high = fmt->frequency[1]; 6191 freq = 48000; 6192 if (low <= freq && freq <= high) { 6193 return freq; 6194 } 6195 freq = 44100; 6196 if (low <= freq && freq <= high) { 6197 return freq; 6198 } 6199 return high; 6200 } else { 6201 for (j = 0; j < fmt->frequency_type; j++) { 6202 if (fmt->frequency[j] == 48000) { 6203 return fmt->frequency[j]; 6204 } 6205 } 6206 high = 0; 6207 for (j = 0; j < fmt->frequency_type; j++) { 6208 if (fmt->frequency[j] == 44100) { 6209 return fmt->frequency[j]; 6210 } 6211 if (fmt->frequency[j] > high) { 6212 high = fmt->frequency[j]; 6213 } 6214 } 6215 return high; 6216 } 6217 } 6218 6219 /* 6220 * Choose the most preferred hardware format. 6221 * If successful, it will store the chosen format into *cand and return 0. 6222 * Otherwise, return errno. 6223 * Must be called without sc_lock held. 6224 */ 6225 static int 6226 audio_hw_probe(struct audio_softc *sc, audio_format2_t *cand, int mode) 6227 { 6228 audio_format_query_t query; 6229 int cand_score; 6230 int score; 6231 int i; 6232 int error; 6233 6234 /* 6235 * Score each formats and choose the highest one. 6236 * 6237 * +---- priority(0-3) 6238 * |+--- encoding/precision 6239 * ||+-- channels 6240 * score = 0x000000PEC 6241 */ 6242 6243 cand_score = 0; 6244 for (i = 0; ; i++) { 6245 memset(&query, 0, sizeof(query)); 6246 query.index = i; 6247 6248 mutex_enter(sc->sc_lock); 6249 error = sc->hw_if->query_format(sc->hw_hdl, &query); 6250 mutex_exit(sc->sc_lock); 6251 if (error == EINVAL) 6252 break; 6253 if (error) 6254 return error; 6255 6256 #if defined(AUDIO_DEBUG) 6257 DPRINTF(1, "fmt[%d] %c%c pri=%d %s,%d/%dbit,%dch,", i, 6258 (query.fmt.mode & AUMODE_PLAY) ? 'P' : '-', 6259 (query.fmt.mode & AUMODE_RECORD) ? 'R' : '-', 6260 query.fmt.priority, 6261 audio_encoding_name(query.fmt.encoding), 6262 query.fmt.validbits, 6263 query.fmt.precision, 6264 query.fmt.channels); 6265 if (query.fmt.frequency_type == 0) { 6266 DPRINTF(1, "{%d-%d", 6267 query.fmt.frequency[0], query.fmt.frequency[1]); 6268 } else { 6269 int j; 6270 for (j = 0; j < query.fmt.frequency_type; j++) { 6271 DPRINTF(1, "%c%d", 6272 (j == 0) ? '{' : ',', 6273 query.fmt.frequency[j]); 6274 } 6275 } 6276 DPRINTF(1, "}\n"); 6277 #endif 6278 6279 if ((query.fmt.mode & mode) == 0) { 6280 DPRINTF(1, "fmt[%d] skip; mode not match %d\n", i, 6281 mode); 6282 continue; 6283 } 6284 6285 if (query.fmt.priority < 0) { 6286 DPRINTF(1, "fmt[%d] skip; unsupported encoding\n", i); 6287 continue; 6288 } 6289 6290 /* Score */ 6291 score = (query.fmt.priority & 3) * 0x100; 6292 if (query.fmt.encoding == AUDIO_ENCODING_SLINEAR_NE && 6293 query.fmt.validbits == AUDIO_INTERNAL_BITS && 6294 query.fmt.precision == AUDIO_INTERNAL_BITS) { 6295 score += 0x20; 6296 } else if (query.fmt.encoding == AUDIO_ENCODING_SLINEAR_OE && 6297 query.fmt.validbits == AUDIO_INTERNAL_BITS && 6298 query.fmt.precision == AUDIO_INTERNAL_BITS) { 6299 score += 0x10; 6300 } 6301 score += query.fmt.channels; 6302 6303 if (score < cand_score) { 6304 DPRINTF(1, "fmt[%d] skip; score 0x%x < 0x%x\n", i, 6305 score, cand_score); 6306 continue; 6307 } 6308 6309 /* Update candidate */ 6310 cand_score = score; 6311 cand->encoding = query.fmt.encoding; 6312 cand->precision = query.fmt.validbits; 6313 cand->stride = query.fmt.precision; 6314 cand->channels = query.fmt.channels; 6315 cand->sample_rate = audio_select_freq(&query.fmt); 6316 DPRINTF(1, "fmt[%d] candidate (score=0x%x)" 6317 " pri=%d %s,%d/%d,%dch,%dHz\n", i, 6318 cand_score, query.fmt.priority, 6319 audio_encoding_name(query.fmt.encoding), 6320 cand->precision, cand->stride, 6321 cand->channels, cand->sample_rate); 6322 } 6323 6324 if (cand_score == 0) { 6325 DPRINTF(1, "%s no fmt\n", __func__); 6326 return ENXIO; 6327 } 6328 DPRINTF(1, "%s selected: %s,%d/%d,%dch,%dHz\n", __func__, 6329 audio_encoding_name(cand->encoding), 6330 cand->precision, cand->stride, cand->channels, cand->sample_rate); 6331 return 0; 6332 } 6333 6334 /* 6335 * Validate fmt with query_format. 6336 * If fmt is included in the result of query_format, returns 0. 6337 * Otherwise returns EINVAL. 6338 * Must be called with sc_lock held. 6339 */ 6340 static int 6341 audio_hw_validate_format(struct audio_softc *sc, int mode, 6342 const audio_format2_t *fmt) 6343 { 6344 audio_format_query_t query; 6345 struct audio_format *q; 6346 int index; 6347 int error; 6348 int j; 6349 6350 KASSERT(mutex_owned(sc->sc_lock)); 6351 6352 for (index = 0; ; index++) { 6353 query.index = index; 6354 error = sc->hw_if->query_format(sc->hw_hdl, &query); 6355 if (error == EINVAL) 6356 break; 6357 if (error) 6358 return error; 6359 6360 q = &query.fmt; 6361 /* 6362 * Note that fmt is audio_format2_t (precision/stride) but 6363 * q is audio_format_t (validbits/precision). 6364 */ 6365 if ((q->mode & mode) == 0) { 6366 continue; 6367 } 6368 if (fmt->encoding != q->encoding) { 6369 continue; 6370 } 6371 if (fmt->precision != q->validbits) { 6372 continue; 6373 } 6374 if (fmt->stride != q->precision) { 6375 continue; 6376 } 6377 if (fmt->channels != q->channels) { 6378 continue; 6379 } 6380 if (q->frequency_type == 0) { 6381 if (fmt->sample_rate < q->frequency[0] || 6382 fmt->sample_rate > q->frequency[1]) { 6383 continue; 6384 } 6385 } else { 6386 for (j = 0; j < q->frequency_type; j++) { 6387 if (fmt->sample_rate == q->frequency[j]) 6388 break; 6389 } 6390 if (j == query.fmt.frequency_type) { 6391 continue; 6392 } 6393 } 6394 6395 /* Matched. */ 6396 return 0; 6397 } 6398 6399 return EINVAL; 6400 } 6401 6402 /* 6403 * Set track mixer's format depending on ai->mode. 6404 * If AUMODE_PLAY is set in ai->mode, it set up the playback mixer 6405 * with ai.play.*. 6406 * If AUMODE_RECORD is set in ai->mode, it set up the recording mixer 6407 * with ai.record.*. 6408 * All other fields in ai are ignored. 6409 * If successful returns 0. Otherwise returns errno. 6410 * This function does not roll back even if it fails. 6411 * Must be called with sc_lock held. 6412 */ 6413 static int 6414 audio_mixers_set_format(struct audio_softc *sc, const struct audio_info *ai) 6415 { 6416 audio_format2_t phwfmt; 6417 audio_format2_t rhwfmt; 6418 audio_filter_reg_t pfil; 6419 audio_filter_reg_t rfil; 6420 int mode; 6421 int error; 6422 6423 KASSERT(mutex_owned(sc->sc_lock)); 6424 6425 /* 6426 * Even when setting either one of playback and recording, 6427 * both must be halted. 6428 */ 6429 if (sc->sc_popens + sc->sc_ropens > 0) 6430 return EBUSY; 6431 6432 if (!SPECIFIED(ai->mode) || ai->mode == 0) 6433 return ENOTTY; 6434 6435 mode = ai->mode; 6436 if ((mode & AUMODE_PLAY)) { 6437 phwfmt.encoding = ai->play.encoding; 6438 phwfmt.precision = ai->play.precision; 6439 phwfmt.stride = ai->play.precision; 6440 phwfmt.channels = ai->play.channels; 6441 phwfmt.sample_rate = ai->play.sample_rate; 6442 } 6443 if ((mode & AUMODE_RECORD)) { 6444 rhwfmt.encoding = ai->record.encoding; 6445 rhwfmt.precision = ai->record.precision; 6446 rhwfmt.stride = ai->record.precision; 6447 rhwfmt.channels = ai->record.channels; 6448 rhwfmt.sample_rate = ai->record.sample_rate; 6449 } 6450 6451 /* On non-independent devices, use the same format for both. */ 6452 if ((sc->sc_props & AUDIO_PROP_INDEPENDENT) == 0) { 6453 if (mode == AUMODE_RECORD) { 6454 phwfmt = rhwfmt; 6455 } else { 6456 rhwfmt = phwfmt; 6457 } 6458 mode = AUMODE_PLAY | AUMODE_RECORD; 6459 } 6460 6461 /* Then, unset the direction not exist on the hardware. */ 6462 if ((sc->sc_props & AUDIO_PROP_PLAYBACK) == 0) 6463 mode &= ~AUMODE_PLAY; 6464 if ((sc->sc_props & AUDIO_PROP_CAPTURE) == 0) 6465 mode &= ~AUMODE_RECORD; 6466 6467 /* debug */ 6468 if ((mode & AUMODE_PLAY)) { 6469 TRACE(1, "play=%s/%d/%d/%dch/%dHz", 6470 audio_encoding_name(phwfmt.encoding), 6471 phwfmt.precision, 6472 phwfmt.stride, 6473 phwfmt.channels, 6474 phwfmt.sample_rate); 6475 } 6476 if ((mode & AUMODE_RECORD)) { 6477 TRACE(1, "rec =%s/%d/%d/%dch/%dHz", 6478 audio_encoding_name(rhwfmt.encoding), 6479 rhwfmt.precision, 6480 rhwfmt.stride, 6481 rhwfmt.channels, 6482 rhwfmt.sample_rate); 6483 } 6484 6485 /* Check the format */ 6486 if ((mode & AUMODE_PLAY)) { 6487 if (audio_hw_validate_format(sc, AUMODE_PLAY, &phwfmt)) { 6488 TRACE(1, "invalid format"); 6489 return EINVAL; 6490 } 6491 } 6492 if ((mode & AUMODE_RECORD)) { 6493 if (audio_hw_validate_format(sc, AUMODE_RECORD, &rhwfmt)) { 6494 TRACE(1, "invalid format"); 6495 return EINVAL; 6496 } 6497 } 6498 6499 /* Configure the mixers. */ 6500 memset(&pfil, 0, sizeof(pfil)); 6501 memset(&rfil, 0, sizeof(rfil)); 6502 error = audio_hw_set_format(sc, mode, &phwfmt, &rhwfmt, &pfil, &rfil); 6503 if (error) 6504 return error; 6505 6506 error = audio_mixers_init(sc, mode, &phwfmt, &rhwfmt, &pfil, &rfil); 6507 if (error) 6508 return error; 6509 6510 return 0; 6511 } 6512 6513 /* 6514 * Store current mixers format into *ai. 6515 */ 6516 static void 6517 audio_mixers_get_format(struct audio_softc *sc, struct audio_info *ai) 6518 { 6519 /* 6520 * There is no stride information in audio_info but it doesn't matter. 6521 * trackmixer always treats stride and precision as the same. 6522 */ 6523 AUDIO_INITINFO(ai); 6524 ai->mode = 0; 6525 if (sc->sc_pmixer) { 6526 audio_format2_t *fmt = &sc->sc_pmixer->track_fmt; 6527 ai->play.encoding = fmt->encoding; 6528 ai->play.precision = fmt->precision; 6529 ai->play.channels = fmt->channels; 6530 ai->play.sample_rate = fmt->sample_rate; 6531 ai->mode |= AUMODE_PLAY; 6532 } 6533 if (sc->sc_rmixer) { 6534 audio_format2_t *fmt = &sc->sc_rmixer->track_fmt; 6535 ai->record.encoding = fmt->encoding; 6536 ai->record.precision = fmt->precision; 6537 ai->record.channels = fmt->channels; 6538 ai->record.sample_rate = fmt->sample_rate; 6539 ai->mode |= AUMODE_RECORD; 6540 } 6541 } 6542 6543 /* 6544 * audio_info details: 6545 * 6546 * ai.{play,record}.sample_rate (R/W) 6547 * ai.{play,record}.encoding (R/W) 6548 * ai.{play,record}.precision (R/W) 6549 * ai.{play,record}.channels (R/W) 6550 * These specify the playback or recording format. 6551 * Ignore members within an inactive track. 6552 * 6553 * ai.mode (R/W) 6554 * It specifies the playback or recording mode, AUMODE_*. 6555 * Currently, a mode change operation by ai.mode after opening is 6556 * prohibited. In addition, AUMODE_PLAY_ALL no longer makes sense. 6557 * However, it's possible to get or to set for backward compatibility. 6558 * 6559 * ai.{hiwat,lowat} (R/W) 6560 * These specify the high water mark and low water mark for playback 6561 * track. The unit is block. 6562 * 6563 * ai.{play,record}.gain (R/W) 6564 * It specifies the HW mixer volume in 0-255. 6565 * It is historical reason that the gain is connected to HW mixer. 6566 * 6567 * ai.{play,record}.balance (R/W) 6568 * It specifies the left-right balance of HW mixer in 0-64. 6569 * 32 means the center. 6570 * It is historical reason that the balance is connected to HW mixer. 6571 * 6572 * ai.{play,record}.port (R/W) 6573 * It specifies the input/output port of HW mixer. 6574 * 6575 * ai.monitor_gain (R/W) 6576 * It specifies the recording monitor gain(?) of HW mixer. 6577 * 6578 * ai.{play,record}.pause (R/W) 6579 * Non-zero means the track is paused. 6580 * 6581 * ai.play.seek (R/-) 6582 * It indicates the number of bytes written but not processed. 6583 * ai.record.seek (R/-) 6584 * It indicates the number of bytes to be able to read. 6585 * 6586 * ai.{play,record}.avail_ports (R/-) 6587 * Mixer info. 6588 * 6589 * ai.{play,record}.buffer_size (R/-) 6590 * It indicates the buffer size in bytes. Internally it means usrbuf. 6591 * 6592 * ai.{play,record}.samples (R/-) 6593 * It indicates the total number of bytes played or recorded. 6594 * 6595 * ai.{play,record}.eof (R/-) 6596 * It indicates the number of times reached EOF(?). 6597 * 6598 * ai.{play,record}.error (R/-) 6599 * Non-zero indicates overflow/underflow has occured. 6600 * 6601 * ai.{play,record}.waiting (R/-) 6602 * Non-zero indicates that other process waits to open. 6603 * It will never happen anymore. 6604 * 6605 * ai.{play,record}.open (R/-) 6606 * Non-zero indicates the direction is opened by this process(?). 6607 * XXX Is this better to indicate that "the device is opened by 6608 * at least one process"? 6609 * 6610 * ai.{play,record}.active (R/-) 6611 * Non-zero indicates that I/O is currently active. 6612 * 6613 * ai.blocksize (R/-) 6614 * It indicates the block size in bytes. 6615 * XXX The blocksize of playback and recording may be different. 6616 */ 6617 6618 /* 6619 * Pause consideration: 6620 * 6621 * The introduction of these two behavior makes pause/unpause operation 6622 * simple. 6623 * 1. The first read/write access of the first track makes mixer start. 6624 * 2. A pause of the last track doesn't make mixer stop. 6625 */ 6626 6627 /* 6628 * Set both track's parameters within a file depending on ai. 6629 * Update sc_sound_[pr]* if set. 6630 * Must be called with sc_lock and sc_exlock held. 6631 */ 6632 static int 6633 audio_file_setinfo(struct audio_softc *sc, audio_file_t *file, 6634 const struct audio_info *ai) 6635 { 6636 const struct audio_prinfo *pi; 6637 const struct audio_prinfo *ri; 6638 audio_track_t *ptrack; 6639 audio_track_t *rtrack; 6640 audio_format2_t pfmt; 6641 audio_format2_t rfmt; 6642 int pchanges; 6643 int rchanges; 6644 int mode; 6645 struct audio_info saved_ai; 6646 audio_format2_t saved_pfmt; 6647 audio_format2_t saved_rfmt; 6648 int error; 6649 6650 KASSERT(mutex_owned(sc->sc_lock)); 6651 KASSERT(sc->sc_exlock); 6652 6653 pi = &ai->play; 6654 ri = &ai->record; 6655 pchanges = 0; 6656 rchanges = 0; 6657 6658 ptrack = file->ptrack; 6659 rtrack = file->rtrack; 6660 6661 #if defined(AUDIO_DEBUG) 6662 if (audiodebug >= 2) { 6663 char buf[256]; 6664 char p[64]; 6665 int buflen; 6666 int plen; 6667 #define SPRINTF(var, fmt...) do { \ 6668 var##len += snprintf(var + var##len, sizeof(var) - var##len, fmt); \ 6669 } while (0) 6670 6671 buflen = 0; 6672 plen = 0; 6673 if (SPECIFIED(pi->encoding)) 6674 SPRINTF(p, "/%s", audio_encoding_name(pi->encoding)); 6675 if (SPECIFIED(pi->precision)) 6676 SPRINTF(p, "/%dbit", pi->precision); 6677 if (SPECIFIED(pi->channels)) 6678 SPRINTF(p, "/%dch", pi->channels); 6679 if (SPECIFIED(pi->sample_rate)) 6680 SPRINTF(p, "/%dHz", pi->sample_rate); 6681 if (plen > 0) 6682 SPRINTF(buf, ",play.param=%s", p + 1); 6683 6684 plen = 0; 6685 if (SPECIFIED(ri->encoding)) 6686 SPRINTF(p, "/%s", audio_encoding_name(ri->encoding)); 6687 if (SPECIFIED(ri->precision)) 6688 SPRINTF(p, "/%dbit", ri->precision); 6689 if (SPECIFIED(ri->channels)) 6690 SPRINTF(p, "/%dch", ri->channels); 6691 if (SPECIFIED(ri->sample_rate)) 6692 SPRINTF(p, "/%dHz", ri->sample_rate); 6693 if (plen > 0) 6694 SPRINTF(buf, ",record.param=%s", p + 1); 6695 6696 if (SPECIFIED(ai->mode)) 6697 SPRINTF(buf, ",mode=%d", ai->mode); 6698 if (SPECIFIED(ai->hiwat)) 6699 SPRINTF(buf, ",hiwat=%d", ai->hiwat); 6700 if (SPECIFIED(ai->lowat)) 6701 SPRINTF(buf, ",lowat=%d", ai->lowat); 6702 if (SPECIFIED(ai->play.gain)) 6703 SPRINTF(buf, ",play.gain=%d", ai->play.gain); 6704 if (SPECIFIED(ai->record.gain)) 6705 SPRINTF(buf, ",record.gain=%d", ai->record.gain); 6706 if (SPECIFIED_CH(ai->play.balance)) 6707 SPRINTF(buf, ",play.balance=%d", ai->play.balance); 6708 if (SPECIFIED_CH(ai->record.balance)) 6709 SPRINTF(buf, ",record.balance=%d", ai->record.balance); 6710 if (SPECIFIED(ai->play.port)) 6711 SPRINTF(buf, ",play.port=%d", ai->play.port); 6712 if (SPECIFIED(ai->record.port)) 6713 SPRINTF(buf, ",record.port=%d", ai->record.port); 6714 if (SPECIFIED(ai->monitor_gain)) 6715 SPRINTF(buf, ",monitor_gain=%d", ai->monitor_gain); 6716 if (SPECIFIED_CH(ai->play.pause)) 6717 SPRINTF(buf, ",play.pause=%d", ai->play.pause); 6718 if (SPECIFIED_CH(ai->record.pause)) 6719 SPRINTF(buf, ",record.pause=%d", ai->record.pause); 6720 6721 if (buflen > 0) 6722 TRACE(2, "specified %s", buf + 1); 6723 } 6724 #endif 6725 6726 AUDIO_INITINFO(&saved_ai); 6727 /* XXX shut up gcc */ 6728 memset(&saved_pfmt, 0, sizeof(saved_pfmt)); 6729 memset(&saved_rfmt, 0, sizeof(saved_rfmt)); 6730 6731 /* Set default value and save current parameters */ 6732 if (ptrack) { 6733 pfmt = ptrack->usrbuf.fmt; 6734 saved_pfmt = ptrack->usrbuf.fmt; 6735 saved_ai.play.pause = ptrack->is_pause; 6736 } 6737 if (rtrack) { 6738 rfmt = rtrack->usrbuf.fmt; 6739 saved_rfmt = rtrack->usrbuf.fmt; 6740 saved_ai.record.pause = rtrack->is_pause; 6741 } 6742 saved_ai.mode = file->mode; 6743 6744 /* Overwrite if specified */ 6745 mode = file->mode; 6746 if (SPECIFIED(ai->mode)) { 6747 /* 6748 * Setting ai->mode no longer does anything because it's 6749 * prohibited to change playback/recording mode after open 6750 * and AUMODE_PLAY_ALL is obsoleted. However, it still 6751 * keeps the state of AUMODE_PLAY_ALL itself for backward 6752 * compatibility. 6753 * In the internal, only file->mode has the state of 6754 * AUMODE_PLAY_ALL flag and track->mode in both track does 6755 * not have. 6756 */ 6757 if ((file->mode & AUMODE_PLAY)) { 6758 mode = (file->mode & (AUMODE_PLAY | AUMODE_RECORD)) 6759 | (ai->mode & AUMODE_PLAY_ALL); 6760 } 6761 } 6762 6763 if (ptrack) { 6764 pchanges = audio_track_setinfo_check(&pfmt, pi, 6765 &sc->sc_pmixer->hwbuf.fmt); 6766 if (pchanges == -1) { 6767 #if defined(AUDIO_DEBUG) 6768 char fmtbuf[64]; 6769 audio_format2_tostr(fmtbuf, sizeof(fmtbuf), &pfmt); 6770 TRACET(1, ptrack, "check play.params failed: %s", 6771 fmtbuf); 6772 #endif 6773 return EINVAL; 6774 } 6775 if (SPECIFIED(ai->mode)) 6776 pchanges = 1; 6777 } 6778 if (rtrack) { 6779 rchanges = audio_track_setinfo_check(&rfmt, ri, 6780 &sc->sc_rmixer->hwbuf.fmt); 6781 if (rchanges == -1) { 6782 #if defined(AUDIO_DEBUG) 6783 char fmtbuf[64]; 6784 audio_format2_tostr(fmtbuf, sizeof(fmtbuf), &rfmt); 6785 TRACET(1, rtrack, "check record.params failed: %s", 6786 fmtbuf); 6787 #endif 6788 return EINVAL; 6789 } 6790 if (SPECIFIED(ai->mode)) 6791 rchanges = 1; 6792 } 6793 6794 /* 6795 * Even when setting either one of playback and recording, 6796 * both track must be halted. 6797 */ 6798 if (pchanges || rchanges) { 6799 audio_file_clear(sc, file); 6800 #if defined(AUDIO_DEBUG) 6801 char fmtbuf[64]; 6802 if (pchanges) { 6803 audio_format2_tostr(fmtbuf, sizeof(fmtbuf), &pfmt); 6804 DPRINTF(1, "audio track#%d play mode: %s\n", 6805 ptrack->id, fmtbuf); 6806 } 6807 if (rchanges) { 6808 audio_format2_tostr(fmtbuf, sizeof(fmtbuf), &rfmt); 6809 DPRINTF(1, "audio track#%d rec mode: %s\n", 6810 rtrack->id, fmtbuf); 6811 } 6812 #endif 6813 } 6814 6815 /* Set mixer parameters */ 6816 error = audio_hw_setinfo(sc, ai, &saved_ai); 6817 if (error) 6818 goto abort1; 6819 6820 /* Set to track and update sticky parameters */ 6821 error = 0; 6822 file->mode = mode; 6823 if (ptrack) { 6824 if (SPECIFIED_CH(pi->pause)) { 6825 ptrack->is_pause = pi->pause; 6826 sc->sc_sound_ppause = pi->pause; 6827 } 6828 if (pchanges) { 6829 audio_track_lock_enter(ptrack); 6830 error = audio_track_set_format(ptrack, &pfmt); 6831 audio_track_lock_exit(ptrack); 6832 if (error) { 6833 TRACET(1, ptrack, "set play.params failed"); 6834 goto abort2; 6835 } 6836 sc->sc_sound_pparams = pfmt; 6837 } 6838 /* Change water marks after initializing the buffers. */ 6839 if (SPECIFIED(ai->hiwat) || SPECIFIED(ai->lowat)) 6840 audio_track_setinfo_water(ptrack, ai); 6841 } 6842 if (rtrack) { 6843 if (SPECIFIED_CH(ri->pause)) { 6844 rtrack->is_pause = ri->pause; 6845 sc->sc_sound_rpause = ri->pause; 6846 } 6847 if (rchanges) { 6848 audio_track_lock_enter(rtrack); 6849 error = audio_track_set_format(rtrack, &rfmt); 6850 audio_track_lock_exit(rtrack); 6851 if (error) { 6852 TRACET(1, rtrack, "set record.params failed"); 6853 goto abort3; 6854 } 6855 sc->sc_sound_rparams = rfmt; 6856 } 6857 } 6858 6859 return 0; 6860 6861 /* Rollback */ 6862 abort3: 6863 if (error != ENOMEM) { 6864 rtrack->is_pause = saved_ai.record.pause; 6865 audio_track_lock_enter(rtrack); 6866 audio_track_set_format(rtrack, &saved_rfmt); 6867 audio_track_lock_exit(rtrack); 6868 } 6869 abort2: 6870 if (ptrack && error != ENOMEM) { 6871 ptrack->is_pause = saved_ai.play.pause; 6872 audio_track_lock_enter(ptrack); 6873 audio_track_set_format(ptrack, &saved_pfmt); 6874 audio_track_lock_exit(ptrack); 6875 sc->sc_sound_pparams = saved_pfmt; 6876 sc->sc_sound_ppause = saved_ai.play.pause; 6877 } 6878 file->mode = saved_ai.mode; 6879 abort1: 6880 audio_hw_setinfo(sc, &saved_ai, NULL); 6881 6882 return error; 6883 } 6884 6885 /* 6886 * Write SPECIFIED() parameters within info back to fmt. 6887 * Return value of 1 indicates that fmt is modified. 6888 * Return value of 0 indicates that fmt is not modified. 6889 * Return value of -1 indicates that error EINVAL has occurred. 6890 */ 6891 static int 6892 audio_track_setinfo_check(audio_format2_t *fmt, const struct audio_prinfo *info, 6893 const audio_format2_t *hwfmt) 6894 { 6895 int changes; 6896 6897 changes = 0; 6898 if (SPECIFIED(info->sample_rate)) { 6899 if (info->sample_rate < AUDIO_MIN_FREQUENCY) 6900 return -1; 6901 if (info->sample_rate > AUDIO_MAX_FREQUENCY) 6902 return -1; 6903 fmt->sample_rate = info->sample_rate; 6904 changes = 1; 6905 } 6906 if (SPECIFIED(info->encoding)) { 6907 fmt->encoding = info->encoding; 6908 changes = 1; 6909 } 6910 if (SPECIFIED(info->precision)) { 6911 fmt->precision = info->precision; 6912 /* we don't have API to specify stride */ 6913 fmt->stride = info->precision; 6914 changes = 1; 6915 } 6916 if (SPECIFIED(info->channels)) { 6917 /* 6918 * We can convert between monaural and stereo each other. 6919 * We can reduce than the number of channels that the hardware 6920 * supports. 6921 */ 6922 if (info->channels > 2 && info->channels > hwfmt->channels) 6923 return -1; 6924 fmt->channels = info->channels; 6925 changes = 1; 6926 } 6927 6928 if (changes) { 6929 if (audio_check_params(fmt) != 0) 6930 return -1; 6931 } 6932 6933 return changes; 6934 } 6935 6936 /* 6937 * Change water marks for playback track if specfied. 6938 */ 6939 static void 6940 audio_track_setinfo_water(audio_track_t *track, const struct audio_info *ai) 6941 { 6942 u_int blks; 6943 u_int maxblks; 6944 u_int blksize; 6945 6946 KASSERT(audio_track_is_playback(track)); 6947 6948 blksize = track->usrbuf_blksize; 6949 maxblks = track->usrbuf.capacity / blksize; 6950 6951 if (SPECIFIED(ai->hiwat)) { 6952 blks = ai->hiwat; 6953 if (blks > maxblks) 6954 blks = maxblks; 6955 if (blks < 2) 6956 blks = 2; 6957 track->usrbuf_usedhigh = blks * blksize; 6958 } 6959 if (SPECIFIED(ai->lowat)) { 6960 blks = ai->lowat; 6961 if (blks > maxblks - 1) 6962 blks = maxblks - 1; 6963 track->usrbuf_usedlow = blks * blksize; 6964 } 6965 if (SPECIFIED(ai->hiwat) || SPECIFIED(ai->lowat)) { 6966 if (track->usrbuf_usedlow > track->usrbuf_usedhigh - blksize) { 6967 track->usrbuf_usedlow = track->usrbuf_usedhigh - 6968 blksize; 6969 } 6970 } 6971 } 6972 6973 /* 6974 * Set hardware part of *newai. 6975 * The parameters handled here are *.port, *.gain, *.balance and monitor_gain. 6976 * If oldai is specified, previous parameters are stored. 6977 * This function itself does not roll back if error occurred. 6978 * Must be called with sc_lock and sc_exlock held. 6979 */ 6980 static int 6981 audio_hw_setinfo(struct audio_softc *sc, const struct audio_info *newai, 6982 struct audio_info *oldai) 6983 { 6984 const struct audio_prinfo *newpi; 6985 const struct audio_prinfo *newri; 6986 struct audio_prinfo *oldpi; 6987 struct audio_prinfo *oldri; 6988 u_int pgain; 6989 u_int rgain; 6990 u_char pbalance; 6991 u_char rbalance; 6992 int error; 6993 6994 KASSERT(mutex_owned(sc->sc_lock)); 6995 KASSERT(sc->sc_exlock); 6996 6997 /* XXX shut up gcc */ 6998 oldpi = NULL; 6999 oldri = NULL; 7000 7001 newpi = &newai->play; 7002 newri = &newai->record; 7003 if (oldai) { 7004 oldpi = &oldai->play; 7005 oldri = &oldai->record; 7006 } 7007 error = 0; 7008 7009 /* 7010 * It looks like unnecessary to halt HW mixers to set HW mixers. 7011 * mixer_ioctl(MIXER_WRITE) also doesn't halt. 7012 */ 7013 7014 if (SPECIFIED(newpi->port)) { 7015 if (oldai) 7016 oldpi->port = au_get_port(sc, &sc->sc_outports); 7017 error = au_set_port(sc, &sc->sc_outports, newpi->port); 7018 if (error) { 7019 device_printf(sc->sc_dev, 7020 "setting play.port=%d failed with %d\n", 7021 newpi->port, error); 7022 goto abort; 7023 } 7024 } 7025 if (SPECIFIED(newri->port)) { 7026 if (oldai) 7027 oldri->port = au_get_port(sc, &sc->sc_inports); 7028 error = au_set_port(sc, &sc->sc_inports, newri->port); 7029 if (error) { 7030 device_printf(sc->sc_dev, 7031 "setting record.port=%d failed with %d\n", 7032 newri->port, error); 7033 goto abort; 7034 } 7035 } 7036 7037 /* Backup play.{gain,balance} */ 7038 if (SPECIFIED(newpi->gain) || SPECIFIED_CH(newpi->balance)) { 7039 au_get_gain(sc, &sc->sc_outports, &pgain, &pbalance); 7040 if (oldai) { 7041 oldpi->gain = pgain; 7042 oldpi->balance = pbalance; 7043 } 7044 } 7045 /* Backup record.{gain,balance} */ 7046 if (SPECIFIED(newri->gain) || SPECIFIED_CH(newri->balance)) { 7047 au_get_gain(sc, &sc->sc_inports, &rgain, &rbalance); 7048 if (oldai) { 7049 oldri->gain = rgain; 7050 oldri->balance = rbalance; 7051 } 7052 } 7053 if (SPECIFIED(newpi->gain)) { 7054 error = au_set_gain(sc, &sc->sc_outports, 7055 newpi->gain, pbalance); 7056 if (error) { 7057 device_printf(sc->sc_dev, 7058 "setting play.gain=%d failed with %d\n", 7059 newpi->gain, error); 7060 goto abort; 7061 } 7062 } 7063 if (SPECIFIED(newri->gain)) { 7064 error = au_set_gain(sc, &sc->sc_inports, 7065 newri->gain, rbalance); 7066 if (error) { 7067 device_printf(sc->sc_dev, 7068 "setting record.gain=%d failed with %d\n", 7069 newri->gain, error); 7070 goto abort; 7071 } 7072 } 7073 if (SPECIFIED_CH(newpi->balance)) { 7074 error = au_set_gain(sc, &sc->sc_outports, 7075 pgain, newpi->balance); 7076 if (error) { 7077 device_printf(sc->sc_dev, 7078 "setting play.balance=%d failed with %d\n", 7079 newpi->balance, error); 7080 goto abort; 7081 } 7082 } 7083 if (SPECIFIED_CH(newri->balance)) { 7084 error = au_set_gain(sc, &sc->sc_inports, 7085 rgain, newri->balance); 7086 if (error) { 7087 device_printf(sc->sc_dev, 7088 "setting record.balance=%d failed with %d\n", 7089 newri->balance, error); 7090 goto abort; 7091 } 7092 } 7093 7094 if (SPECIFIED(newai->monitor_gain) && sc->sc_monitor_port != -1) { 7095 if (oldai) 7096 oldai->monitor_gain = au_get_monitor_gain(sc); 7097 error = au_set_monitor_gain(sc, newai->monitor_gain); 7098 if (error) { 7099 device_printf(sc->sc_dev, 7100 "setting monitor_gain=%d failed with %d\n", 7101 newai->monitor_gain, error); 7102 goto abort; 7103 } 7104 } 7105 7106 /* XXX TODO */ 7107 /* sc->sc_ai = *ai; */ 7108 7109 error = 0; 7110 abort: 7111 return error; 7112 } 7113 7114 /* 7115 * Setup the hardware with mixer format phwfmt, rhwfmt. 7116 * The arguments have following restrictions: 7117 * - setmode is the direction you want to set, AUMODE_PLAY or AUMODE_RECORD, 7118 * or both. 7119 * - phwfmt and rhwfmt must not be NULL regardless of setmode. 7120 * - On non-independent devices, phwfmt and rhwfmt must have the same 7121 * parameters. 7122 * - pfil and rfil must be zero-filled. 7123 * If successful, 7124 * - pfil, rfil will be filled with filter information specified by the 7125 * hardware driver. 7126 * and then returns 0. Otherwise returns errno. 7127 * Must be called with sc_lock held. 7128 */ 7129 static int 7130 audio_hw_set_format(struct audio_softc *sc, int setmode, 7131 const audio_format2_t *phwfmt, const audio_format2_t *rhwfmt, 7132 audio_filter_reg_t *pfil, audio_filter_reg_t *rfil) 7133 { 7134 audio_params_t pp, rp; 7135 int error; 7136 7137 KASSERT(mutex_owned(sc->sc_lock)); 7138 KASSERT(phwfmt != NULL); 7139 KASSERT(rhwfmt != NULL); 7140 7141 pp = format2_to_params(phwfmt); 7142 rp = format2_to_params(rhwfmt); 7143 7144 error = sc->hw_if->set_format(sc->hw_hdl, setmode, 7145 &pp, &rp, pfil, rfil); 7146 if (error) { 7147 device_printf(sc->sc_dev, 7148 "set_format failed with %d\n", error); 7149 return error; 7150 } 7151 7152 if (sc->hw_if->commit_settings) { 7153 error = sc->hw_if->commit_settings(sc->hw_hdl); 7154 if (error) { 7155 device_printf(sc->sc_dev, 7156 "commit_settings failed with %d\n", error); 7157 return error; 7158 } 7159 } 7160 7161 return 0; 7162 } 7163 7164 /* 7165 * Fill audio_info structure. If need_mixerinfo is true, it will also 7166 * fill the hardware mixer information. 7167 * Must be called with sc_lock held. 7168 * Must be called with sc_exlock held, in addition, if need_mixerinfo is 7169 * true. 7170 */ 7171 static int 7172 audiogetinfo(struct audio_softc *sc, struct audio_info *ai, int need_mixerinfo, 7173 audio_file_t *file) 7174 { 7175 struct audio_prinfo *ri, *pi; 7176 audio_track_t *track; 7177 audio_track_t *ptrack; 7178 audio_track_t *rtrack; 7179 int gain; 7180 7181 KASSERT(mutex_owned(sc->sc_lock)); 7182 7183 ri = &ai->record; 7184 pi = &ai->play; 7185 ptrack = file->ptrack; 7186 rtrack = file->rtrack; 7187 7188 memset(ai, 0, sizeof(*ai)); 7189 7190 if (ptrack) { 7191 pi->sample_rate = ptrack->usrbuf.fmt.sample_rate; 7192 pi->channels = ptrack->usrbuf.fmt.channels; 7193 pi->precision = ptrack->usrbuf.fmt.precision; 7194 pi->encoding = ptrack->usrbuf.fmt.encoding; 7195 } else { 7196 /* Set default parameters if the track is not available. */ 7197 if (ISDEVAUDIO(file->dev)) { 7198 pi->sample_rate = audio_default.sample_rate; 7199 pi->channels = audio_default.channels; 7200 pi->precision = audio_default.precision; 7201 pi->encoding = audio_default.encoding; 7202 } else { 7203 pi->sample_rate = sc->sc_sound_pparams.sample_rate; 7204 pi->channels = sc->sc_sound_pparams.channels; 7205 pi->precision = sc->sc_sound_pparams.precision; 7206 pi->encoding = sc->sc_sound_pparams.encoding; 7207 } 7208 } 7209 if (rtrack) { 7210 ri->sample_rate = rtrack->usrbuf.fmt.sample_rate; 7211 ri->channels = rtrack->usrbuf.fmt.channels; 7212 ri->precision = rtrack->usrbuf.fmt.precision; 7213 ri->encoding = rtrack->usrbuf.fmt.encoding; 7214 } else { 7215 /* Set default parameters if the track is not available. */ 7216 if (ISDEVAUDIO(file->dev)) { 7217 ri->sample_rate = audio_default.sample_rate; 7218 ri->channels = audio_default.channels; 7219 ri->precision = audio_default.precision; 7220 ri->encoding = audio_default.encoding; 7221 } else { 7222 ri->sample_rate = sc->sc_sound_rparams.sample_rate; 7223 ri->channels = sc->sc_sound_rparams.channels; 7224 ri->precision = sc->sc_sound_rparams.precision; 7225 ri->encoding = sc->sc_sound_rparams.encoding; 7226 } 7227 } 7228 7229 if (ptrack) { 7230 pi->seek = ptrack->usrbuf.used; 7231 pi->samples = ptrack->usrbuf_stamp; 7232 pi->eof = ptrack->eofcounter; 7233 pi->pause = ptrack->is_pause; 7234 pi->error = (ptrack->dropframes != 0) ? 1 : 0; 7235 pi->waiting = 0; /* open never hangs */ 7236 pi->open = 1; 7237 pi->active = sc->sc_pbusy; 7238 pi->buffer_size = ptrack->usrbuf.capacity; 7239 } 7240 if (rtrack) { 7241 ri->seek = rtrack->usrbuf.used; 7242 ri->samples = rtrack->usrbuf_stamp; 7243 ri->eof = 0; 7244 ri->pause = rtrack->is_pause; 7245 ri->error = (rtrack->dropframes != 0) ? 1 : 0; 7246 ri->waiting = 0; /* open never hangs */ 7247 ri->open = 1; 7248 ri->active = sc->sc_rbusy; 7249 ri->buffer_size = rtrack->usrbuf.capacity; 7250 } 7251 7252 /* 7253 * XXX There may be different number of channels between playback 7254 * and recording, so that blocksize also may be different. 7255 * But struct audio_info has an united blocksize... 7256 * Here, I use play info precedencely if ptrack is available, 7257 * otherwise record info. 7258 * 7259 * XXX hiwat/lowat is a playback-only parameter. What should I 7260 * return for a record-only descriptor? 7261 */ 7262 track = ptrack ? ptrack : rtrack; 7263 if (track) { 7264 ai->blocksize = track->usrbuf_blksize; 7265 ai->hiwat = track->usrbuf_usedhigh / track->usrbuf_blksize; 7266 ai->lowat = track->usrbuf_usedlow / track->usrbuf_blksize; 7267 } 7268 ai->mode = file->mode; 7269 7270 if (need_mixerinfo) { 7271 KASSERT(sc->sc_exlock); 7272 7273 pi->port = au_get_port(sc, &sc->sc_outports); 7274 ri->port = au_get_port(sc, &sc->sc_inports); 7275 7276 pi->avail_ports = sc->sc_outports.allports; 7277 ri->avail_ports = sc->sc_inports.allports; 7278 7279 au_get_gain(sc, &sc->sc_outports, &pi->gain, &pi->balance); 7280 au_get_gain(sc, &sc->sc_inports, &ri->gain, &ri->balance); 7281 7282 if (sc->sc_monitor_port != -1) { 7283 gain = au_get_monitor_gain(sc); 7284 if (gain != -1) 7285 ai->monitor_gain = gain; 7286 } 7287 } 7288 7289 return 0; 7290 } 7291 7292 /* 7293 * Return true if playback is configured. 7294 * This function can be used after audioattach. 7295 */ 7296 static bool 7297 audio_can_playback(struct audio_softc *sc) 7298 { 7299 7300 return (sc->sc_pmixer != NULL); 7301 } 7302 7303 /* 7304 * Return true if recording is configured. 7305 * This function can be used after audioattach. 7306 */ 7307 static bool 7308 audio_can_capture(struct audio_softc *sc) 7309 { 7310 7311 return (sc->sc_rmixer != NULL); 7312 } 7313 7314 /* 7315 * Get the afp->index'th item from the valid one of format[]. 7316 * If found, stores it to afp->fmt and returns 0. Otherwise return EINVAL. 7317 * 7318 * This is common routines for query_format. 7319 * If your hardware driver has struct audio_format[], the simplest case 7320 * you can write your query_format interface as follows: 7321 * 7322 * struct audio_format foo_format[] = { ... }; 7323 * 7324 * int 7325 * foo_query_format(void *hdl, audio_format_query_t *afp) 7326 * { 7327 * return audio_query_format(foo_format, __arraycount(foo_format), afp); 7328 * } 7329 */ 7330 int 7331 audio_query_format(const struct audio_format *format, int nformats, 7332 audio_format_query_t *afp) 7333 { 7334 const struct audio_format *f; 7335 int idx; 7336 int i; 7337 7338 idx = 0; 7339 for (i = 0; i < nformats; i++) { 7340 f = &format[i]; 7341 if (!AUFMT_IS_VALID(f)) 7342 continue; 7343 if (afp->index == idx) { 7344 afp->fmt = *f; 7345 return 0; 7346 } 7347 idx++; 7348 } 7349 return EINVAL; 7350 } 7351 7352 /* 7353 * This function is provided for the hardware driver's set_format() to 7354 * find index matches with 'param' from array of audio_format_t 'formats'. 7355 * 'mode' is either of AUMODE_PLAY or AUMODE_RECORD. 7356 * It returns the matched index and never fails. Because param passed to 7357 * set_format() is selected from query_format(). 7358 * This function will be an alternative to auconv_set_converter() to 7359 * find index. 7360 */ 7361 int 7362 audio_indexof_format(const struct audio_format *formats, int nformats, 7363 int mode, const audio_params_t *param) 7364 { 7365 const struct audio_format *f; 7366 int index; 7367 int j; 7368 7369 for (index = 0; index < nformats; index++) { 7370 f = &formats[index]; 7371 7372 if (!AUFMT_IS_VALID(f)) 7373 continue; 7374 if ((f->mode & mode) == 0) 7375 continue; 7376 if (f->encoding != param->encoding) 7377 continue; 7378 if (f->validbits != param->precision) 7379 continue; 7380 if (f->channels != param->channels) 7381 continue; 7382 7383 if (f->frequency_type == 0) { 7384 if (param->sample_rate < f->frequency[0] || 7385 param->sample_rate > f->frequency[1]) 7386 continue; 7387 } else { 7388 for (j = 0; j < f->frequency_type; j++) { 7389 if (param->sample_rate == f->frequency[j]) 7390 break; 7391 } 7392 if (j == f->frequency_type) 7393 continue; 7394 } 7395 7396 /* Then, matched */ 7397 return index; 7398 } 7399 7400 /* Not matched. This should not be happened. */ 7401 panic("%s: cannot find matched format\n", __func__); 7402 } 7403 7404 /* 7405 * Get or set hardware blocksize in msec. 7406 * XXX It's for debug. 7407 */ 7408 static int 7409 audio_sysctl_blk_ms(SYSCTLFN_ARGS) 7410 { 7411 struct sysctlnode node; 7412 struct audio_softc *sc; 7413 audio_format2_t phwfmt; 7414 audio_format2_t rhwfmt; 7415 audio_filter_reg_t pfil; 7416 audio_filter_reg_t rfil; 7417 int t; 7418 int old_blk_ms; 7419 int mode; 7420 int error; 7421 7422 node = *rnode; 7423 sc = node.sysctl_data; 7424 7425 mutex_enter(sc->sc_lock); 7426 7427 old_blk_ms = sc->sc_blk_ms; 7428 t = old_blk_ms; 7429 node.sysctl_data = &t; 7430 error = sysctl_lookup(SYSCTLFN_CALL(&node)); 7431 if (error || newp == NULL) 7432 goto abort; 7433 7434 if (t < 0) { 7435 error = EINVAL; 7436 goto abort; 7437 } 7438 7439 if (sc->sc_popens + sc->sc_ropens > 0) { 7440 error = EBUSY; 7441 goto abort; 7442 } 7443 sc->sc_blk_ms = t; 7444 mode = 0; 7445 if (sc->sc_pmixer) { 7446 mode |= AUMODE_PLAY; 7447 phwfmt = sc->sc_pmixer->hwbuf.fmt; 7448 } 7449 if (sc->sc_rmixer) { 7450 mode |= AUMODE_RECORD; 7451 rhwfmt = sc->sc_rmixer->hwbuf.fmt; 7452 } 7453 7454 /* re-init hardware */ 7455 memset(&pfil, 0, sizeof(pfil)); 7456 memset(&rfil, 0, sizeof(rfil)); 7457 error = audio_hw_set_format(sc, mode, &phwfmt, &rhwfmt, &pfil, &rfil); 7458 if (error) { 7459 goto abort; 7460 } 7461 7462 /* re-init track mixer */ 7463 error = audio_mixers_init(sc, mode, &phwfmt, &rhwfmt, &pfil, &rfil); 7464 if (error) { 7465 /* Rollback */ 7466 sc->sc_blk_ms = old_blk_ms; 7467 audio_mixers_init(sc, mode, &phwfmt, &rhwfmt, &pfil, &rfil); 7468 goto abort; 7469 } 7470 error = 0; 7471 abort: 7472 mutex_exit(sc->sc_lock); 7473 return error; 7474 } 7475 7476 /* 7477 * Get or set multiuser mode. 7478 */ 7479 static int 7480 audio_sysctl_multiuser(SYSCTLFN_ARGS) 7481 { 7482 struct sysctlnode node; 7483 struct audio_softc *sc; 7484 bool t; 7485 int error; 7486 7487 node = *rnode; 7488 sc = node.sysctl_data; 7489 7490 mutex_enter(sc->sc_lock); 7491 7492 t = sc->sc_multiuser; 7493 node.sysctl_data = &t; 7494 error = sysctl_lookup(SYSCTLFN_CALL(&node)); 7495 if (error || newp == NULL) 7496 goto abort; 7497 7498 sc->sc_multiuser = t; 7499 error = 0; 7500 abort: 7501 mutex_exit(sc->sc_lock); 7502 return error; 7503 } 7504 7505 #if defined(AUDIO_DEBUG) 7506 /* 7507 * Get or set debug verbose level. (0..4) 7508 * XXX It's for debug. 7509 * XXX It is not separated per device. 7510 */ 7511 static int 7512 audio_sysctl_debug(SYSCTLFN_ARGS) 7513 { 7514 struct sysctlnode node; 7515 int t; 7516 int error; 7517 7518 node = *rnode; 7519 t = audiodebug; 7520 node.sysctl_data = &t; 7521 error = sysctl_lookup(SYSCTLFN_CALL(&node)); 7522 if (error || newp == NULL) 7523 return error; 7524 7525 if (t < 0 || t > 4) 7526 return EINVAL; 7527 audiodebug = t; 7528 printf("audio: audiodebug = %d\n", audiodebug); 7529 return 0; 7530 } 7531 #endif /* AUDIO_DEBUG */ 7532 7533 #ifdef AUDIO_PM_IDLE 7534 static void 7535 audio_idle(void *arg) 7536 { 7537 device_t dv = arg; 7538 struct audio_softc *sc = device_private(dv); 7539 7540 #ifdef PNP_DEBUG 7541 extern int pnp_debug_idle; 7542 if (pnp_debug_idle) 7543 printf("%s: idle handler called\n", device_xname(dv)); 7544 #endif 7545 7546 sc->sc_idle = true; 7547 7548 /* XXX joerg Make pmf_device_suspend handle children? */ 7549 if (!pmf_device_suspend(dv, PMF_Q_SELF)) 7550 return; 7551 7552 if (!pmf_device_suspend(sc->hw_dev, PMF_Q_SELF)) 7553 pmf_device_resume(dv, PMF_Q_SELF); 7554 } 7555 7556 static void 7557 audio_activity(device_t dv, devactive_t type) 7558 { 7559 struct audio_softc *sc = device_private(dv); 7560 7561 if (type != DVA_SYSTEM) 7562 return; 7563 7564 callout_schedule(&sc->sc_idle_counter, audio_idle_timeout * hz); 7565 7566 sc->sc_idle = false; 7567 if (!device_is_active(dv)) { 7568 /* XXX joerg How to deal with a failing resume... */ 7569 pmf_device_resume(sc->hw_dev, PMF_Q_SELF); 7570 pmf_device_resume(dv, PMF_Q_SELF); 7571 } 7572 } 7573 #endif 7574 7575 static bool 7576 audio_suspend(device_t dv, const pmf_qual_t *qual) 7577 { 7578 struct audio_softc *sc = device_private(dv); 7579 int error; 7580 7581 error = audio_enter_exclusive(sc); 7582 if (error) 7583 return error; 7584 audio_mixer_capture(sc); 7585 7586 /* Halts mixers but don't clear busy flag for resume */ 7587 if (sc->sc_pbusy) { 7588 audio_pmixer_halt(sc); 7589 sc->sc_pbusy = true; 7590 } 7591 if (sc->sc_rbusy) { 7592 audio_rmixer_halt(sc); 7593 sc->sc_rbusy = true; 7594 } 7595 7596 #ifdef AUDIO_PM_IDLE 7597 callout_halt(&sc->sc_idle_counter, sc->sc_lock); 7598 #endif 7599 audio_exit_exclusive(sc); 7600 7601 return true; 7602 } 7603 7604 static bool 7605 audio_resume(device_t dv, const pmf_qual_t *qual) 7606 { 7607 struct audio_softc *sc = device_private(dv); 7608 struct audio_info ai; 7609 int error; 7610 7611 error = audio_enter_exclusive(sc); 7612 if (error) 7613 return error; 7614 7615 audio_mixer_restore(sc); 7616 /* XXX ? */ 7617 AUDIO_INITINFO(&ai); 7618 audio_hw_setinfo(sc, &ai, NULL); 7619 7620 if (sc->sc_pbusy) 7621 audio_pmixer_start(sc, true); 7622 if (sc->sc_rbusy) 7623 audio_rmixer_start(sc); 7624 7625 audio_exit_exclusive(sc); 7626 7627 return true; 7628 } 7629 7630 #if defined(AUDIO_DEBUG) 7631 static void 7632 audio_format2_tostr(char *buf, size_t bufsize, const audio_format2_t *fmt) 7633 { 7634 int n; 7635 7636 n = 0; 7637 n += snprintf(buf + n, bufsize - n, "%s", 7638 audio_encoding_name(fmt->encoding)); 7639 if (fmt->precision == fmt->stride) { 7640 n += snprintf(buf + n, bufsize - n, " %dbit", fmt->precision); 7641 } else { 7642 n += snprintf(buf + n, bufsize - n, " %d/%dbit", 7643 fmt->precision, fmt->stride); 7644 } 7645 7646 snprintf(buf + n, bufsize - n, " %uch %uHz", 7647 fmt->channels, fmt->sample_rate); 7648 } 7649 #endif 7650 7651 #if defined(AUDIO_DEBUG) 7652 static void 7653 audio_print_format2(const char *s, const audio_format2_t *fmt) 7654 { 7655 char fmtstr[64]; 7656 7657 audio_format2_tostr(fmtstr, sizeof(fmtstr), fmt); 7658 printf("%s %s\n", s, fmtstr); 7659 } 7660 #endif 7661 7662 #ifdef DIAGNOSTIC 7663 void 7664 audio_diagnostic_format2(const char *where, const audio_format2_t *fmt) 7665 { 7666 7667 KASSERTMSG(fmt, "called from %s", where); 7668 7669 /* XXX MSM6258 vs(4) only has 4bit stride format. */ 7670 if (fmt->encoding == AUDIO_ENCODING_ADPCM) { 7671 KASSERTMSG(fmt->stride == 4 || fmt->stride == 8, 7672 "called from %s: fmt->stride=%d", where, fmt->stride); 7673 } else { 7674 KASSERTMSG(fmt->stride % NBBY == 0, 7675 "called from %s: fmt->stride=%d", where, fmt->stride); 7676 } 7677 KASSERTMSG(fmt->precision <= fmt->stride, 7678 "called from %s: fmt->precision=%d fmt->stride=%d", 7679 where, fmt->precision, fmt->stride); 7680 KASSERTMSG(1 <= fmt->channels && fmt->channels <= AUDIO_MAX_CHANNELS, 7681 "called from %s: fmt->channels=%d", where, fmt->channels); 7682 7683 /* XXX No check for encodings? */ 7684 } 7685 7686 void 7687 audio_diagnostic_filter_arg(const char *where, const audio_filter_arg_t *arg) 7688 { 7689 7690 KASSERT(arg != NULL); 7691 KASSERT(arg->src != NULL); 7692 KASSERT(arg->dst != NULL); 7693 audio_diagnostic_format2(where, arg->srcfmt); 7694 audio_diagnostic_format2(where, arg->dstfmt); 7695 KASSERT(arg->count > 0); 7696 } 7697 7698 void 7699 audio_diagnostic_ring(const char *where, const audio_ring_t *ring) 7700 { 7701 7702 KASSERTMSG(ring, "called from %s", where); 7703 audio_diagnostic_format2(where, &ring->fmt); 7704 KASSERTMSG(0 <= ring->capacity && ring->capacity < INT_MAX / 2, 7705 "called from %s: ring->capacity=%d", where, ring->capacity); 7706 KASSERTMSG(0 <= ring->used && ring->used <= ring->capacity, 7707 "called from %s: ring->used=%d ring->capacity=%d", 7708 where, ring->used, ring->capacity); 7709 if (ring->capacity == 0) { 7710 KASSERTMSG(ring->mem == NULL, 7711 "called from %s: capacity == 0 but mem != NULL", where); 7712 } else { 7713 KASSERTMSG(ring->mem != NULL, 7714 "called from %s: capacity != 0 but mem == NULL", where); 7715 KASSERTMSG(0 <= ring->head && ring->head < ring->capacity, 7716 "called from %s: ring->head=%d ring->capacity=%d", 7717 where, ring->head, ring->capacity); 7718 } 7719 } 7720 #endif /* DIAGNOSTIC */ 7721 7722 7723 /* 7724 * Mixer driver 7725 */ 7726 int 7727 mixer_open(dev_t dev, struct audio_softc *sc, int flags, int ifmt, 7728 struct lwp *l) 7729 { 7730 struct file *fp; 7731 audio_file_t *af; 7732 int error, fd; 7733 7734 KASSERT(mutex_owned(sc->sc_lock)); 7735 7736 TRACE(1, "flags=0x%x", flags); 7737 7738 error = fd_allocfile(&fp, &fd); 7739 if (error) 7740 return error; 7741 7742 af = kmem_zalloc(sizeof(*af), KM_SLEEP); 7743 af->sc = sc; 7744 af->dev = dev; 7745 7746 error = fd_clone(fp, fd, flags, &audio_fileops, af); 7747 KASSERT(error == EMOVEFD); 7748 7749 return error; 7750 } 7751 7752 /* 7753 * Add a process to those to be signalled on mixer activity. 7754 * If the process has already been added, do nothing. 7755 * Must be called with sc_lock held. 7756 */ 7757 static void 7758 mixer_async_add(struct audio_softc *sc, pid_t pid) 7759 { 7760 int i; 7761 7762 KASSERT(mutex_owned(sc->sc_lock)); 7763 7764 /* If already exists, returns without doing anything. */ 7765 for (i = 0; i < sc->sc_am_used; i++) { 7766 if (sc->sc_am[i] == pid) 7767 return; 7768 } 7769 7770 /* Extend array if necessary. */ 7771 if (sc->sc_am_used >= sc->sc_am_capacity) { 7772 sc->sc_am_capacity += AM_CAPACITY; 7773 sc->sc_am = kern_realloc(sc->sc_am, 7774 sc->sc_am_capacity * sizeof(pid_t), M_WAITOK); 7775 TRACE(2, "realloc am_capacity=%d", sc->sc_am_capacity); 7776 } 7777 7778 TRACE(2, "am[%d]=%d", sc->sc_am_used, (int)pid); 7779 sc->sc_am[sc->sc_am_used++] = pid; 7780 } 7781 7782 /* 7783 * Remove a process from those to be signalled on mixer activity. 7784 * If the process has not been added, do nothing. 7785 * Must be called with sc_lock held. 7786 */ 7787 static void 7788 mixer_async_remove(struct audio_softc *sc, pid_t pid) 7789 { 7790 int i; 7791 7792 KASSERT(mutex_owned(sc->sc_lock)); 7793 7794 for (i = 0; i < sc->sc_am_used; i++) { 7795 if (sc->sc_am[i] == pid) { 7796 sc->sc_am[i] = sc->sc_am[--sc->sc_am_used]; 7797 TRACE(2, "am[%d](%d) removed, used=%d", 7798 i, (int)pid, sc->sc_am_used); 7799 7800 /* Empty array if no longer necessary. */ 7801 if (sc->sc_am_used == 0) { 7802 kern_free(sc->sc_am); 7803 sc->sc_am = NULL; 7804 sc->sc_am_capacity = 0; 7805 TRACE(2, "released"); 7806 } 7807 return; 7808 } 7809 } 7810 } 7811 7812 /* 7813 * Signal all processes waiting for the mixer. 7814 * Must be called with sc_lock held. 7815 */ 7816 static void 7817 mixer_signal(struct audio_softc *sc) 7818 { 7819 proc_t *p; 7820 int i; 7821 7822 KASSERT(mutex_owned(sc->sc_lock)); 7823 7824 for (i = 0; i < sc->sc_am_used; i++) { 7825 mutex_enter(proc_lock); 7826 p = proc_find(sc->sc_am[i]); 7827 if (p) 7828 psignal(p, SIGIO); 7829 mutex_exit(proc_lock); 7830 } 7831 } 7832 7833 /* 7834 * Close a mixer device 7835 */ 7836 int 7837 mixer_close(struct audio_softc *sc, audio_file_t *file) 7838 { 7839 7840 mutex_enter(sc->sc_lock); 7841 TRACE(1, ""); 7842 mixer_async_remove(sc, curproc->p_pid); 7843 mutex_exit(sc->sc_lock); 7844 7845 return 0; 7846 } 7847 7848 /* 7849 * Must be called without sc_lock nor sc_exlock held. 7850 */ 7851 int 7852 mixer_ioctl(struct audio_softc *sc, u_long cmd, void *addr, int flag, 7853 struct lwp *l) 7854 { 7855 mixer_devinfo_t *mi; 7856 mixer_ctrl_t *mc; 7857 int error; 7858 7859 TRACE(2, "(%lu,'%c',%lu)", 7860 IOCPARM_LEN(cmd), (char)IOCGROUP(cmd), cmd & 0xff); 7861 error = EINVAL; 7862 7863 /* we can return cached values if we are sleeping */ 7864 if (cmd != AUDIO_MIXER_READ) { 7865 mutex_enter(sc->sc_lock); 7866 device_active(sc->sc_dev, DVA_SYSTEM); 7867 mutex_exit(sc->sc_lock); 7868 } 7869 7870 switch (cmd) { 7871 case FIOASYNC: 7872 mutex_enter(sc->sc_lock); 7873 if (*(int *)addr) { 7874 mixer_async_add(sc, curproc->p_pid); 7875 } else { 7876 mixer_async_remove(sc, curproc->p_pid); 7877 } 7878 mutex_exit(sc->sc_lock); 7879 error = 0; 7880 break; 7881 7882 case AUDIO_GETDEV: 7883 TRACE(2, "AUDIO_GETDEV"); 7884 error = audio_enter_exclusive(sc); 7885 if (error) 7886 break; 7887 error = sc->hw_if->getdev(sc->hw_hdl, (audio_device_t *)addr); 7888 audio_exit_exclusive(sc); 7889 break; 7890 7891 case AUDIO_MIXER_DEVINFO: 7892 TRACE(2, "AUDIO_MIXER_DEVINFO"); 7893 mi = (mixer_devinfo_t *)addr; 7894 7895 mi->un.v.delta = 0; /* default */ 7896 mutex_enter(sc->sc_lock); 7897 error = audio_query_devinfo(sc, mi); 7898 mutex_exit(sc->sc_lock); 7899 break; 7900 7901 case AUDIO_MIXER_READ: 7902 TRACE(2, "AUDIO_MIXER_READ"); 7903 mc = (mixer_ctrl_t *)addr; 7904 7905 error = audio_enter_exclusive(sc); 7906 if (error) 7907 break; 7908 if (device_is_active(sc->hw_dev)) 7909 error = audio_get_port(sc, mc); 7910 else if (mc->dev < 0 || mc->dev >= sc->sc_nmixer_states) 7911 error = ENXIO; 7912 else { 7913 int dev = mc->dev; 7914 memcpy(mc, &sc->sc_mixer_state[dev], 7915 sizeof(mixer_ctrl_t)); 7916 error = 0; 7917 } 7918 audio_exit_exclusive(sc); 7919 break; 7920 7921 case AUDIO_MIXER_WRITE: 7922 TRACE(2, "AUDIO_MIXER_WRITE"); 7923 error = audio_enter_exclusive(sc); 7924 if (error) 7925 break; 7926 error = audio_set_port(sc, (mixer_ctrl_t *)addr); 7927 if (error) { 7928 audio_exit_exclusive(sc); 7929 break; 7930 } 7931 7932 if (sc->hw_if->commit_settings) { 7933 error = sc->hw_if->commit_settings(sc->hw_hdl); 7934 if (error) { 7935 audio_exit_exclusive(sc); 7936 break; 7937 } 7938 } 7939 mixer_signal(sc); 7940 audio_exit_exclusive(sc); 7941 break; 7942 7943 default: 7944 if (sc->hw_if->dev_ioctl) { 7945 error = audio_enter_exclusive(sc); 7946 if (error) 7947 break; 7948 error = sc->hw_if->dev_ioctl(sc->hw_hdl, 7949 cmd, addr, flag, l); 7950 audio_exit_exclusive(sc); 7951 } else 7952 error = EINVAL; 7953 break; 7954 } 7955 TRACE(2, "(%lu,'%c',%lu) result %d", 7956 IOCPARM_LEN(cmd), (char)IOCGROUP(cmd), cmd & 0xff, error); 7957 return error; 7958 } 7959 7960 /* 7961 * Must be called with sc_lock held. 7962 */ 7963 int 7964 au_portof(struct audio_softc *sc, char *name, int class) 7965 { 7966 mixer_devinfo_t mi; 7967 7968 KASSERT(mutex_owned(sc->sc_lock)); 7969 7970 for (mi.index = 0; audio_query_devinfo(sc, &mi) == 0; mi.index++) { 7971 if (mi.mixer_class == class && strcmp(mi.label.name, name) == 0) 7972 return mi.index; 7973 } 7974 return -1; 7975 } 7976 7977 /* 7978 * Must be called with sc_lock held. 7979 */ 7980 void 7981 au_setup_ports(struct audio_softc *sc, struct au_mixer_ports *ports, 7982 mixer_devinfo_t *mi, const struct portname *tbl) 7983 { 7984 int i, j; 7985 7986 KASSERT(mutex_owned(sc->sc_lock)); 7987 7988 ports->index = mi->index; 7989 if (mi->type == AUDIO_MIXER_ENUM) { 7990 ports->isenum = true; 7991 for(i = 0; tbl[i].name; i++) 7992 for(j = 0; j < mi->un.e.num_mem; j++) 7993 if (strcmp(mi->un.e.member[j].label.name, 7994 tbl[i].name) == 0) { 7995 ports->allports |= tbl[i].mask; 7996 ports->aumask[ports->nports] = tbl[i].mask; 7997 ports->misel[ports->nports] = 7998 mi->un.e.member[j].ord; 7999 ports->miport[ports->nports] = 8000 au_portof(sc, mi->un.e.member[j].label.name, 8001 mi->mixer_class); 8002 if (ports->mixerout != -1 && 8003 ports->miport[ports->nports] != -1) 8004 ports->isdual = true; 8005 ++ports->nports; 8006 } 8007 } else if (mi->type == AUDIO_MIXER_SET) { 8008 for(i = 0; tbl[i].name; i++) 8009 for(j = 0; j < mi->un.s.num_mem; j++) 8010 if (strcmp(mi->un.s.member[j].label.name, 8011 tbl[i].name) == 0) { 8012 ports->allports |= tbl[i].mask; 8013 ports->aumask[ports->nports] = tbl[i].mask; 8014 ports->misel[ports->nports] = 8015 mi->un.s.member[j].mask; 8016 ports->miport[ports->nports] = 8017 au_portof(sc, mi->un.s.member[j].label.name, 8018 mi->mixer_class); 8019 ++ports->nports; 8020 } 8021 } 8022 } 8023 8024 /* 8025 * Must be called with sc_lock && sc_exlock held. 8026 */ 8027 int 8028 au_set_lr_value(struct audio_softc *sc, mixer_ctrl_t *ct, int l, int r) 8029 { 8030 8031 KASSERT(mutex_owned(sc->sc_lock)); 8032 KASSERT(sc->sc_exlock); 8033 8034 ct->type = AUDIO_MIXER_VALUE; 8035 ct->un.value.num_channels = 2; 8036 ct->un.value.level[AUDIO_MIXER_LEVEL_LEFT] = l; 8037 ct->un.value.level[AUDIO_MIXER_LEVEL_RIGHT] = r; 8038 if (audio_set_port(sc, ct) == 0) 8039 return 0; 8040 ct->un.value.num_channels = 1; 8041 ct->un.value.level[AUDIO_MIXER_LEVEL_MONO] = (l+r)/2; 8042 return audio_set_port(sc, ct); 8043 } 8044 8045 /* 8046 * Must be called with sc_lock && sc_exlock held. 8047 */ 8048 int 8049 au_get_lr_value(struct audio_softc *sc, mixer_ctrl_t *ct, int *l, int *r) 8050 { 8051 int error; 8052 8053 KASSERT(mutex_owned(sc->sc_lock)); 8054 KASSERT(sc->sc_exlock); 8055 8056 ct->un.value.num_channels = 2; 8057 if (audio_get_port(sc, ct) == 0) { 8058 *l = ct->un.value.level[AUDIO_MIXER_LEVEL_LEFT]; 8059 *r = ct->un.value.level[AUDIO_MIXER_LEVEL_RIGHT]; 8060 } else { 8061 ct->un.value.num_channels = 1; 8062 error = audio_get_port(sc, ct); 8063 if (error) 8064 return error; 8065 *r = *l = ct->un.value.level[AUDIO_MIXER_LEVEL_MONO]; 8066 } 8067 return 0; 8068 } 8069 8070 /* 8071 * Must be called with sc_lock && sc_exlock held. 8072 */ 8073 int 8074 au_set_gain(struct audio_softc *sc, struct au_mixer_ports *ports, 8075 int gain, int balance) 8076 { 8077 mixer_ctrl_t ct; 8078 int i, error; 8079 int l, r; 8080 u_int mask; 8081 int nset; 8082 8083 KASSERT(mutex_owned(sc->sc_lock)); 8084 KASSERT(sc->sc_exlock); 8085 8086 if (balance == AUDIO_MID_BALANCE) { 8087 l = r = gain; 8088 } else if (balance < AUDIO_MID_BALANCE) { 8089 l = gain; 8090 r = (balance * gain) / AUDIO_MID_BALANCE; 8091 } else { 8092 r = gain; 8093 l = ((AUDIO_RIGHT_BALANCE - balance) * gain) 8094 / AUDIO_MID_BALANCE; 8095 } 8096 TRACE(2, "gain=%d balance=%d, l=%d r=%d", gain, balance, l, r); 8097 8098 if (ports->index == -1) { 8099 usemaster: 8100 if (ports->master == -1) 8101 return 0; /* just ignore it silently */ 8102 ct.dev = ports->master; 8103 error = au_set_lr_value(sc, &ct, l, r); 8104 } else { 8105 ct.dev = ports->index; 8106 if (ports->isenum) { 8107 ct.type = AUDIO_MIXER_ENUM; 8108 error = audio_get_port(sc, &ct); 8109 if (error) 8110 return error; 8111 if (ports->isdual) { 8112 if (ports->cur_port == -1) 8113 ct.dev = ports->master; 8114 else 8115 ct.dev = ports->miport[ports->cur_port]; 8116 error = au_set_lr_value(sc, &ct, l, r); 8117 } else { 8118 for(i = 0; i < ports->nports; i++) 8119 if (ports->misel[i] == ct.un.ord) { 8120 ct.dev = ports->miport[i]; 8121 if (ct.dev == -1 || 8122 au_set_lr_value(sc, &ct, l, r)) 8123 goto usemaster; 8124 else 8125 break; 8126 } 8127 } 8128 } else { 8129 ct.type = AUDIO_MIXER_SET; 8130 error = audio_get_port(sc, &ct); 8131 if (error) 8132 return error; 8133 mask = ct.un.mask; 8134 nset = 0; 8135 for(i = 0; i < ports->nports; i++) { 8136 if (ports->misel[i] & mask) { 8137 ct.dev = ports->miport[i]; 8138 if (ct.dev != -1 && 8139 au_set_lr_value(sc, &ct, l, r) == 0) 8140 nset++; 8141 } 8142 } 8143 if (nset == 0) 8144 goto usemaster; 8145 } 8146 } 8147 if (!error) 8148 mixer_signal(sc); 8149 return error; 8150 } 8151 8152 /* 8153 * Must be called with sc_lock && sc_exlock held. 8154 */ 8155 void 8156 au_get_gain(struct audio_softc *sc, struct au_mixer_ports *ports, 8157 u_int *pgain, u_char *pbalance) 8158 { 8159 mixer_ctrl_t ct; 8160 int i, l, r, n; 8161 int lgain, rgain; 8162 8163 KASSERT(mutex_owned(sc->sc_lock)); 8164 KASSERT(sc->sc_exlock); 8165 8166 lgain = AUDIO_MAX_GAIN / 2; 8167 rgain = AUDIO_MAX_GAIN / 2; 8168 if (ports->index == -1) { 8169 usemaster: 8170 if (ports->master == -1) 8171 goto bad; 8172 ct.dev = ports->master; 8173 ct.type = AUDIO_MIXER_VALUE; 8174 if (au_get_lr_value(sc, &ct, &lgain, &rgain)) 8175 goto bad; 8176 } else { 8177 ct.dev = ports->index; 8178 if (ports->isenum) { 8179 ct.type = AUDIO_MIXER_ENUM; 8180 if (audio_get_port(sc, &ct)) 8181 goto bad; 8182 ct.type = AUDIO_MIXER_VALUE; 8183 if (ports->isdual) { 8184 if (ports->cur_port == -1) 8185 ct.dev = ports->master; 8186 else 8187 ct.dev = ports->miport[ports->cur_port]; 8188 au_get_lr_value(sc, &ct, &lgain, &rgain); 8189 } else { 8190 for(i = 0; i < ports->nports; i++) 8191 if (ports->misel[i] == ct.un.ord) { 8192 ct.dev = ports->miport[i]; 8193 if (ct.dev == -1 || 8194 au_get_lr_value(sc, &ct, 8195 &lgain, &rgain)) 8196 goto usemaster; 8197 else 8198 break; 8199 } 8200 } 8201 } else { 8202 ct.type = AUDIO_MIXER_SET; 8203 if (audio_get_port(sc, &ct)) 8204 goto bad; 8205 ct.type = AUDIO_MIXER_VALUE; 8206 lgain = rgain = n = 0; 8207 for(i = 0; i < ports->nports; i++) { 8208 if (ports->misel[i] & ct.un.mask) { 8209 ct.dev = ports->miport[i]; 8210 if (ct.dev == -1 || 8211 au_get_lr_value(sc, &ct, &l, &r)) 8212 goto usemaster; 8213 else { 8214 lgain += l; 8215 rgain += r; 8216 n++; 8217 } 8218 } 8219 } 8220 if (n != 0) { 8221 lgain /= n; 8222 rgain /= n; 8223 } 8224 } 8225 } 8226 bad: 8227 if (lgain == rgain) { /* handles lgain==rgain==0 */ 8228 *pgain = lgain; 8229 *pbalance = AUDIO_MID_BALANCE; 8230 } else if (lgain < rgain) { 8231 *pgain = rgain; 8232 /* balance should be > AUDIO_MID_BALANCE */ 8233 *pbalance = AUDIO_RIGHT_BALANCE - 8234 (AUDIO_MID_BALANCE * lgain) / rgain; 8235 } else /* lgain > rgain */ { 8236 *pgain = lgain; 8237 /* balance should be < AUDIO_MID_BALANCE */ 8238 *pbalance = (AUDIO_MID_BALANCE * rgain) / lgain; 8239 } 8240 } 8241 8242 /* 8243 * Must be called with sc_lock && sc_exlock held. 8244 */ 8245 int 8246 au_set_port(struct audio_softc *sc, struct au_mixer_ports *ports, u_int port) 8247 { 8248 mixer_ctrl_t ct; 8249 int i, error, use_mixerout; 8250 8251 KASSERT(mutex_owned(sc->sc_lock)); 8252 KASSERT(sc->sc_exlock); 8253 8254 use_mixerout = 1; 8255 if (port == 0) { 8256 if (ports->allports == 0) 8257 return 0; /* Allow this special case. */ 8258 else if (ports->isdual) { 8259 if (ports->cur_port == -1) { 8260 return 0; 8261 } else { 8262 port = ports->aumask[ports->cur_port]; 8263 ports->cur_port = -1; 8264 use_mixerout = 0; 8265 } 8266 } 8267 } 8268 if (ports->index == -1) 8269 return EINVAL; 8270 ct.dev = ports->index; 8271 if (ports->isenum) { 8272 if (port & (port-1)) 8273 return EINVAL; /* Only one port allowed */ 8274 ct.type = AUDIO_MIXER_ENUM; 8275 error = EINVAL; 8276 for(i = 0; i < ports->nports; i++) 8277 if (ports->aumask[i] == port) { 8278 if (ports->isdual && use_mixerout) { 8279 ct.un.ord = ports->mixerout; 8280 ports->cur_port = i; 8281 } else { 8282 ct.un.ord = ports->misel[i]; 8283 } 8284 error = audio_set_port(sc, &ct); 8285 break; 8286 } 8287 } else { 8288 ct.type = AUDIO_MIXER_SET; 8289 ct.un.mask = 0; 8290 for(i = 0; i < ports->nports; i++) 8291 if (ports->aumask[i] & port) 8292 ct.un.mask |= ports->misel[i]; 8293 if (port != 0 && ct.un.mask == 0) 8294 error = EINVAL; 8295 else 8296 error = audio_set_port(sc, &ct); 8297 } 8298 if (!error) 8299 mixer_signal(sc); 8300 return error; 8301 } 8302 8303 /* 8304 * Must be called with sc_lock && sc_exlock held. 8305 */ 8306 int 8307 au_get_port(struct audio_softc *sc, struct au_mixer_ports *ports) 8308 { 8309 mixer_ctrl_t ct; 8310 int i, aumask; 8311 8312 KASSERT(mutex_owned(sc->sc_lock)); 8313 KASSERT(sc->sc_exlock); 8314 8315 if (ports->index == -1) 8316 return 0; 8317 ct.dev = ports->index; 8318 ct.type = ports->isenum ? AUDIO_MIXER_ENUM : AUDIO_MIXER_SET; 8319 if (audio_get_port(sc, &ct)) 8320 return 0; 8321 aumask = 0; 8322 if (ports->isenum) { 8323 if (ports->isdual && ports->cur_port != -1) { 8324 if (ports->mixerout == ct.un.ord) 8325 aumask = ports->aumask[ports->cur_port]; 8326 else 8327 ports->cur_port = -1; 8328 } 8329 if (aumask == 0) 8330 for(i = 0; i < ports->nports; i++) 8331 if (ports->misel[i] == ct.un.ord) 8332 aumask = ports->aumask[i]; 8333 } else { 8334 for(i = 0; i < ports->nports; i++) 8335 if (ct.un.mask & ports->misel[i]) 8336 aumask |= ports->aumask[i]; 8337 } 8338 return aumask; 8339 } 8340 8341 /* 8342 * It returns 0 if success, otherwise errno. 8343 * Must be called only if sc->sc_monitor_port != -1. 8344 * Must be called with sc_lock && sc_exlock held. 8345 */ 8346 static int 8347 au_set_monitor_gain(struct audio_softc *sc, int monitor_gain) 8348 { 8349 mixer_ctrl_t ct; 8350 8351 KASSERT(mutex_owned(sc->sc_lock)); 8352 KASSERT(sc->sc_exlock); 8353 8354 ct.dev = sc->sc_monitor_port; 8355 ct.type = AUDIO_MIXER_VALUE; 8356 ct.un.value.num_channels = 1; 8357 ct.un.value.level[AUDIO_MIXER_LEVEL_MONO] = monitor_gain; 8358 return audio_set_port(sc, &ct); 8359 } 8360 8361 /* 8362 * It returns monitor gain if success, otherwise -1. 8363 * Must be called only if sc->sc_monitor_port != -1. 8364 * Must be called with sc_lock && sc_exlock held. 8365 */ 8366 static int 8367 au_get_monitor_gain(struct audio_softc *sc) 8368 { 8369 mixer_ctrl_t ct; 8370 8371 KASSERT(mutex_owned(sc->sc_lock)); 8372 KASSERT(sc->sc_exlock); 8373 8374 ct.dev = sc->sc_monitor_port; 8375 ct.type = AUDIO_MIXER_VALUE; 8376 ct.un.value.num_channels = 1; 8377 if (audio_get_port(sc, &ct)) 8378 return -1; 8379 return ct.un.value.level[AUDIO_MIXER_LEVEL_MONO]; 8380 } 8381 8382 /* 8383 * Must be called with sc_lock && sc_exlock held. 8384 */ 8385 static int 8386 audio_set_port(struct audio_softc *sc, mixer_ctrl_t *mc) 8387 { 8388 8389 KASSERT(mutex_owned(sc->sc_lock)); 8390 KASSERT(sc->sc_exlock); 8391 8392 return sc->hw_if->set_port(sc->hw_hdl, mc); 8393 } 8394 8395 /* 8396 * Must be called with sc_lock && sc_exlock held. 8397 */ 8398 static int 8399 audio_get_port(struct audio_softc *sc, mixer_ctrl_t *mc) 8400 { 8401 8402 KASSERT(mutex_owned(sc->sc_lock)); 8403 KASSERT(sc->sc_exlock); 8404 8405 return sc->hw_if->get_port(sc->hw_hdl, mc); 8406 } 8407 8408 /* 8409 * Must be called with sc_lock && sc_exlock held. 8410 */ 8411 static void 8412 audio_mixer_capture(struct audio_softc *sc) 8413 { 8414 mixer_devinfo_t mi; 8415 mixer_ctrl_t *mc; 8416 8417 KASSERT(mutex_owned(sc->sc_lock)); 8418 KASSERT(sc->sc_exlock); 8419 8420 for (mi.index = 0;; mi.index++) { 8421 if (audio_query_devinfo(sc, &mi) != 0) 8422 break; 8423 KASSERT(mi.index < sc->sc_nmixer_states); 8424 if (mi.type == AUDIO_MIXER_CLASS) 8425 continue; 8426 mc = &sc->sc_mixer_state[mi.index]; 8427 mc->dev = mi.index; 8428 mc->type = mi.type; 8429 mc->un.value.num_channels = mi.un.v.num_channels; 8430 (void)audio_get_port(sc, mc); 8431 } 8432 8433 return; 8434 } 8435 8436 /* 8437 * Must be called with sc_lock && sc_exlock held. 8438 */ 8439 static void 8440 audio_mixer_restore(struct audio_softc *sc) 8441 { 8442 mixer_devinfo_t mi; 8443 mixer_ctrl_t *mc; 8444 8445 KASSERT(mutex_owned(sc->sc_lock)); 8446 KASSERT(sc->sc_exlock); 8447 8448 for (mi.index = 0; ; mi.index++) { 8449 if (audio_query_devinfo(sc, &mi) != 0) 8450 break; 8451 if (mi.type == AUDIO_MIXER_CLASS) 8452 continue; 8453 mc = &sc->sc_mixer_state[mi.index]; 8454 (void)audio_set_port(sc, mc); 8455 } 8456 if (sc->hw_if->commit_settings) 8457 sc->hw_if->commit_settings(sc->hw_hdl); 8458 8459 return; 8460 } 8461 8462 static void 8463 audio_volume_down(device_t dv) 8464 { 8465 struct audio_softc *sc = device_private(dv); 8466 mixer_devinfo_t mi; 8467 int newgain; 8468 u_int gain; 8469 u_char balance; 8470 8471 if (audio_enter_exclusive(sc) != 0) 8472 return; 8473 if (sc->sc_outports.index == -1 && sc->sc_outports.master != -1) { 8474 mi.index = sc->sc_outports.master; 8475 mi.un.v.delta = 0; 8476 if (audio_query_devinfo(sc, &mi) == 0) { 8477 au_get_gain(sc, &sc->sc_outports, &gain, &balance); 8478 newgain = gain - mi.un.v.delta; 8479 if (newgain < AUDIO_MIN_GAIN) 8480 newgain = AUDIO_MIN_GAIN; 8481 au_set_gain(sc, &sc->sc_outports, newgain, balance); 8482 } 8483 } 8484 audio_exit_exclusive(sc); 8485 } 8486 8487 static void 8488 audio_volume_up(device_t dv) 8489 { 8490 struct audio_softc *sc = device_private(dv); 8491 mixer_devinfo_t mi; 8492 u_int gain, newgain; 8493 u_char balance; 8494 8495 if (audio_enter_exclusive(sc) != 0) 8496 return; 8497 if (sc->sc_outports.index == -1 && sc->sc_outports.master != -1) { 8498 mi.index = sc->sc_outports.master; 8499 mi.un.v.delta = 0; 8500 if (audio_query_devinfo(sc, &mi) == 0) { 8501 au_get_gain(sc, &sc->sc_outports, &gain, &balance); 8502 newgain = gain + mi.un.v.delta; 8503 if (newgain > AUDIO_MAX_GAIN) 8504 newgain = AUDIO_MAX_GAIN; 8505 au_set_gain(sc, &sc->sc_outports, newgain, balance); 8506 } 8507 } 8508 audio_exit_exclusive(sc); 8509 } 8510 8511 static void 8512 audio_volume_toggle(device_t dv) 8513 { 8514 struct audio_softc *sc = device_private(dv); 8515 u_int gain, newgain; 8516 u_char balance; 8517 8518 if (audio_enter_exclusive(sc) != 0) 8519 return; 8520 au_get_gain(sc, &sc->sc_outports, &gain, &balance); 8521 if (gain != 0) { 8522 sc->sc_lastgain = gain; 8523 newgain = 0; 8524 } else 8525 newgain = sc->sc_lastgain; 8526 au_set_gain(sc, &sc->sc_outports, newgain, balance); 8527 audio_exit_exclusive(sc); 8528 } 8529 8530 static int 8531 audio_query_devinfo(struct audio_softc *sc, mixer_devinfo_t *di) 8532 { 8533 8534 KASSERT(mutex_owned(sc->sc_lock)); 8535 8536 return sc->hw_if->query_devinfo(sc->hw_hdl, di); 8537 } 8538 8539 #endif /* NAUDIO > 0 */ 8540 8541 #if NAUDIO == 0 && (NMIDI > 0 || NMIDIBUS > 0) 8542 #include <sys/param.h> 8543 #include <sys/systm.h> 8544 #include <sys/device.h> 8545 #include <sys/audioio.h> 8546 #include <dev/audio/audio_if.h> 8547 #endif 8548 8549 #if NAUDIO > 0 || (NMIDI > 0 || NMIDIBUS > 0) 8550 int 8551 audioprint(void *aux, const char *pnp) 8552 { 8553 struct audio_attach_args *arg; 8554 const char *type; 8555 8556 if (pnp != NULL) { 8557 arg = aux; 8558 switch (arg->type) { 8559 case AUDIODEV_TYPE_AUDIO: 8560 type = "audio"; 8561 break; 8562 case AUDIODEV_TYPE_MIDI: 8563 type = "midi"; 8564 break; 8565 case AUDIODEV_TYPE_OPL: 8566 type = "opl"; 8567 break; 8568 case AUDIODEV_TYPE_MPU: 8569 type = "mpu"; 8570 break; 8571 default: 8572 panic("audioprint: unknown type %d", arg->type); 8573 } 8574 aprint_normal("%s at %s", type, pnp); 8575 } 8576 return UNCONF; 8577 } 8578 8579 #endif /* NAUDIO > 0 || (NMIDI > 0 || NMIDIBUS > 0) */ 8580 8581 #ifdef _MODULE 8582 8583 devmajor_t audio_bmajor = -1, audio_cmajor = -1; 8584 8585 #include "ioconf.c" 8586 8587 #endif 8588 8589 MODULE(MODULE_CLASS_DRIVER, audio, NULL); 8590 8591 static int 8592 audio_modcmd(modcmd_t cmd, void *arg) 8593 { 8594 int error = 0; 8595 8596 switch (cmd) { 8597 case MODULE_CMD_INIT: 8598 /* XXX interrupt level? */ 8599 audio_psref_class = psref_class_create("audio", IPL_SOFTSERIAL); 8600 #ifdef _MODULE 8601 error = devsw_attach(audio_cd.cd_name, NULL, &audio_bmajor, 8602 &audio_cdevsw, &audio_cmajor); 8603 if (error) 8604 break; 8605 8606 error = config_init_component(cfdriver_ioconf_audio, 8607 cfattach_ioconf_audio, cfdata_ioconf_audio); 8608 if (error) { 8609 devsw_detach(NULL, &audio_cdevsw); 8610 } 8611 #endif 8612 break; 8613 case MODULE_CMD_FINI: 8614 #ifdef _MODULE 8615 devsw_detach(NULL, &audio_cdevsw); 8616 error = config_fini_component(cfdriver_ioconf_audio, 8617 cfattach_ioconf_audio, cfdata_ioconf_audio); 8618 if (error) 8619 devsw_attach(audio_cd.cd_name, NULL, &audio_bmajor, 8620 &audio_cdevsw, &audio_cmajor); 8621 #endif 8622 psref_class_destroy(audio_psref_class); 8623 break; 8624 default: 8625 error = ENOTTY; 8626 break; 8627 } 8628 8629 return error; 8630 } 8631
Indexes created Mon Sep 29 03:10:08 GMT 2025