audio.c revision 1.142
1 /* $NetBSD: audio.c,v 1.142 2023/04/23 08:38:53 mlelstv 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 * Terminology: "sample", "channel", "frame", "block", "track": 67 * 68 * channel frame 69 * | ........ 70 * v : : \ 71 * +------:------:------:- -+------+ : +------+-.. | 72 * #0(L) |sample|sample|sample| .. |sample| : |sample| | 73 * +------:------:------:- -+------+ : +------+-.. | 74 * #1(R) |sample|sample|sample| .. |sample| : |sample| | 75 * +------:------:------:- -+------+ : +------+-.. | track 76 * : : : : | 77 * +------:------:------:- -+------+ : +------+-.. | 78 * |sample|sample|sample| .. |sample| : |sample| | 79 * +------:------:------:- -+------+ : +------+-.. | 80 * : : / 81 * ........ 82 * 83 * \--------------------------------/ \--------.. 84 * block 85 * 86 * - A "frame" is the minimum unit in the time axis direction, and consists 87 * of samples for the number of channels. 88 * - A "block" is basic length of processing. The audio layer basically 89 * handles audio data stream block by block, asks underlying hardware to 90 * process them block by block, and then the hardware raises interrupt by 91 * each block. 92 * - A "track" is single completed audio stream. 93 * 94 * For example, the hardware block is assumed to be 10 msec, and your audio 95 * track consists of 2.1(=3) channels 44.1kHz 16bit PCM, 96 * 97 * "channel" = 3 98 * "sample" = 2 [bytes] 99 * "frame" = 2 [bytes/sample] * 3 [channels] = 6 [bytes] 100 * "block" = 44100 [Hz] * (10/1000) [seconds] * 6 [bytes/frame] = 2646 [bytes] 101 * 102 * The terminologies shown here are only for this MI audio layer. Note that 103 * different terminologies may be used in each manufacturer's datasheet, and 104 * each MD driver may follow it. For example, what we call a "block" is 105 * called a "frame" in sys/dev/pci/yds.c. 106 */ 107 108 /* 109 * Locking: there are three locks per device. 110 * 111 * - sc_lock, provided by the underlying driver. This is an adaptive lock, 112 * returned in the second parameter to hw_if->get_locks(). It is known 113 * as the "thread lock". 114 * 115 * It serializes access to state in all places except the 116 * driver's interrupt service routine. This lock is taken from process 117 * context (example: access to /dev/audio). It is also taken from soft 118 * interrupt handlers in this module, primarily to serialize delivery of 119 * wakeups. This lock may be used/provided by modules external to the 120 * audio subsystem, so take care not to introduce a lock order problem. 121 * LONG TERM SLEEPS MUST NOT OCCUR WITH THIS LOCK HELD. 122 * 123 * - sc_intr_lock, provided by the underlying driver. This may be either a 124 * spinlock (at IPL_SCHED or IPL_VM) or an adaptive lock (IPL_NONE or 125 * IPL_SOFT*), returned in the first parameter to hw_if->get_locks(). It 126 * is known as the "interrupt lock". 127 * 128 * It provides atomic access to the device's hardware state, and to audio 129 * channel data that may be accessed by the hardware driver's ISR. 130 * In all places outside the ISR, sc_lock must be held before taking 131 * sc_intr_lock. This is to ensure that groups of hardware operations are 132 * made atomically. SLEEPS CANNOT OCCUR WITH THIS LOCK HELD. 133 * 134 * - sc_exlock, private to this module. This is a variable protected by 135 * sc_lock. It is known as the "critical section". 136 * Some operations release sc_lock in order to allocate memory, to wait 137 * for in-flight I/O to complete, to copy to/from user context, etc. 138 * sc_exlock provides a critical section even under the circumstance. 139 * "+" in following list indicates the interfaces which necessary to be 140 * protected by sc_exlock. 141 * 142 * List of hardware interface methods, and which locks are held when each 143 * is called by this module: 144 * 145 * METHOD INTR THREAD NOTES 146 * ----------------------- ------- ------- ------------------------- 147 * open x x + 148 * close x x + 149 * query_format - x 150 * set_format - x 151 * round_blocksize - x 152 * commit_settings - x 153 * init_output x x 154 * init_input x x 155 * start_output x x + 156 * start_input x x + 157 * halt_output x x + 158 * halt_input x x + 159 * speaker_ctl x x 160 * getdev - - 161 * set_port - x + 162 * get_port - x + 163 * query_devinfo - x 164 * allocm - - + 165 * freem - - + 166 * round_buffersize - x 167 * get_props - - Called at attach time 168 * trigger_output x x + 169 * trigger_input x x + 170 * dev_ioctl - x 171 * get_locks - - Called at attach time 172 * 173 * In addition, there is an additional lock. 174 * 175 * - track->lock. This is an atomic variable and is similar to the 176 * "interrupt lock". This is one for each track. If any thread context 177 * (and software interrupt context) and hardware interrupt context who 178 * want to access some variables on this track, they must acquire this 179 * lock before. It protects track's consistency between hardware 180 * interrupt context and others. 181 */ 182 183 #include <sys/cdefs.h> 184 __KERNEL_RCSID(0, "$NetBSD: audio.c,v 1.142 2023/04/23 08:38:53 mlelstv Exp $"); 185 186 #ifdef _KERNEL_OPT 187 #include "audio.h" 188 #include "midi.h" 189 #endif 190 191 #if NAUDIO > 0 192 193 #include <sys/types.h> 194 #include <sys/param.h> 195 #include <sys/atomic.h> 196 #include <sys/audioio.h> 197 #include <sys/conf.h> 198 #include <sys/cpu.h> 199 #include <sys/device.h> 200 #include <sys/fcntl.h> 201 #include <sys/file.h> 202 #include <sys/filedesc.h> 203 #include <sys/intr.h> 204 #include <sys/ioctl.h> 205 #include <sys/kauth.h> 206 #include <sys/kernel.h> 207 #include <sys/kmem.h> 208 #include <sys/lock.h> 209 #include <sys/malloc.h> 210 #include <sys/mman.h> 211 #include <sys/module.h> 212 #include <sys/poll.h> 213 #include <sys/proc.h> 214 #include <sys/queue.h> 215 #include <sys/select.h> 216 #include <sys/signalvar.h> 217 #include <sys/stat.h> 218 #include <sys/sysctl.h> 219 #include <sys/systm.h> 220 #include <sys/syslog.h> 221 #include <sys/vnode.h> 222 223 #include <dev/audio/audio_if.h> 224 #include <dev/audio/audiovar.h> 225 #include <dev/audio/audiodef.h> 226 #include <dev/audio/linear.h> 227 #include <dev/audio/mulaw.h> 228 229 #include <machine/endian.h> 230 231 #include <uvm/uvm_extern.h> 232 233 #include "ioconf.h" 234 235 /* 236 * 0: No debug logs 237 * 1: action changes like open/close/set_format/mmap... 238 * 2: + normal operations like read/write/ioctl... 239 * 3: + TRACEs except interrupt 240 * 4: + TRACEs including interrupt 241 */ 242 //#define AUDIO_DEBUG 1 243 244 #if defined(AUDIO_DEBUG) 245 246 int audiodebug = AUDIO_DEBUG; 247 static void audio_vtrace(struct audio_softc *sc, const char *, const char *, 248 const char *, va_list); 249 static void audio_trace(struct audio_softc *sc, const char *, const char *, ...) 250 __printflike(3, 4); 251 static void audio_tracet(const char *, audio_track_t *, const char *, ...) 252 __printflike(3, 4); 253 static void audio_tracef(const char *, audio_file_t *, const char *, ...) 254 __printflike(3, 4); 255 256 /* XXX sloppy memory logger */ 257 static void audio_mlog_init(void); 258 static void audio_mlog_free(void); 259 static void audio_mlog_softintr(void *); 260 extern void audio_mlog_flush(void); 261 extern void audio_mlog_printf(const char *, ...); 262 263 static int mlog_refs; /* reference counter */ 264 static char *mlog_buf[2]; /* double buffer */ 265 static int mlog_buflen; /* buffer length */ 266 static int mlog_used; /* used length */ 267 static int mlog_full; /* number of dropped lines by buffer full */ 268 static int mlog_drop; /* number of dropped lines by busy */ 269 static volatile uint32_t mlog_inuse; /* in-use */ 270 static int mlog_wpage; /* active page */ 271 static void *mlog_sih; /* softint handle */ 272 273 static void 274 audio_mlog_init(void) 275 { 276 mlog_refs++; 277 if (mlog_refs > 1) 278 return; 279 mlog_buflen = 4096; 280 mlog_buf[0] = kmem_zalloc(mlog_buflen, KM_SLEEP); 281 mlog_buf[1] = kmem_zalloc(mlog_buflen, KM_SLEEP); 282 mlog_used = 0; 283 mlog_full = 0; 284 mlog_drop = 0; 285 mlog_inuse = 0; 286 mlog_wpage = 0; 287 mlog_sih = softint_establish(SOFTINT_SERIAL, audio_mlog_softintr, NULL); 288 if (mlog_sih == NULL) 289 printf("%s: softint_establish failed\n", __func__); 290 } 291 292 static void 293 audio_mlog_free(void) 294 { 295 mlog_refs--; 296 if (mlog_refs > 0) 297 return; 298 299 audio_mlog_flush(); 300 if (mlog_sih) 301 softint_disestablish(mlog_sih); 302 kmem_free(mlog_buf[0], mlog_buflen); 303 kmem_free(mlog_buf[1], mlog_buflen); 304 } 305 306 /* 307 * Flush memory buffer. 308 * It must not be called from hardware interrupt context. 309 */ 310 void 311 audio_mlog_flush(void) 312 { 313 if (mlog_refs == 0) 314 return; 315 316 /* Nothing to do if already in use ? */ 317 if (atomic_swap_32(&mlog_inuse, 1) == 1) 318 return; 319 membar_acquire(); 320 321 int rpage = mlog_wpage; 322 mlog_wpage ^= 1; 323 mlog_buf[mlog_wpage][0] = '\0'; 324 mlog_used = 0; 325 326 atomic_store_release(&mlog_inuse, 0); 327 328 if (mlog_buf[rpage][0] != '\0') { 329 printf("%s", mlog_buf[rpage]); 330 if (mlog_drop > 0) 331 printf("mlog_drop %d\n", mlog_drop); 332 if (mlog_full > 0) 333 printf("mlog_full %d\n", mlog_full); 334 } 335 mlog_full = 0; 336 mlog_drop = 0; 337 } 338 339 static void 340 audio_mlog_softintr(void *cookie) 341 { 342 audio_mlog_flush(); 343 } 344 345 void 346 audio_mlog_printf(const char *fmt, ...) 347 { 348 int len; 349 va_list ap; 350 351 if (atomic_swap_32(&mlog_inuse, 1) == 1) { 352 /* already inuse */ 353 mlog_drop++; 354 return; 355 } 356 membar_acquire(); 357 358 va_start(ap, fmt); 359 len = vsnprintf( 360 mlog_buf[mlog_wpage] + mlog_used, 361 mlog_buflen - mlog_used, 362 fmt, ap); 363 va_end(ap); 364 365 mlog_used += len; 366 if (mlog_buflen - mlog_used <= 1) { 367 mlog_full++; 368 } 369 370 atomic_store_release(&mlog_inuse, 0); 371 372 if (mlog_sih) 373 softint_schedule(mlog_sih); 374 } 375 376 /* trace functions */ 377 static void 378 audio_vtrace(struct audio_softc *sc, const char *funcname, const char *header, 379 const char *fmt, va_list ap) 380 { 381 char buf[256]; 382 int n; 383 384 n = 0; 385 buf[0] = '\0'; 386 n += snprintf(buf + n, sizeof(buf) - n, "%s@%d %s", 387 funcname, device_unit(sc->sc_dev), header); 388 n += vsnprintf(buf + n, sizeof(buf) - n, fmt, ap); 389 390 if (cpu_intr_p()) { 391 audio_mlog_printf("%s\n", buf); 392 } else { 393 audio_mlog_flush(); 394 printf("%s\n", buf); 395 } 396 } 397 398 static void 399 audio_trace(struct audio_softc *sc, const char *funcname, const char *fmt, ...) 400 { 401 va_list ap; 402 403 va_start(ap, fmt); 404 audio_vtrace(sc, funcname, "", fmt, ap); 405 va_end(ap); 406 } 407 408 static void 409 audio_tracet(const char *funcname, audio_track_t *track, const char *fmt, ...) 410 { 411 char hdr[16]; 412 va_list ap; 413 414 snprintf(hdr, sizeof(hdr), "#%d ", track->id); 415 va_start(ap, fmt); 416 audio_vtrace(track->mixer->sc, funcname, hdr, fmt, ap); 417 va_end(ap); 418 } 419 420 static void 421 audio_tracef(const char *funcname, audio_file_t *file, const char *fmt, ...) 422 { 423 char hdr[32]; 424 char phdr[16], rhdr[16]; 425 va_list ap; 426 427 phdr[0] = '\0'; 428 rhdr[0] = '\0'; 429 if (file->ptrack) 430 snprintf(phdr, sizeof(phdr), "#%d", file->ptrack->id); 431 if (file->rtrack) 432 snprintf(rhdr, sizeof(rhdr), "#%d", file->rtrack->id); 433 snprintf(hdr, sizeof(hdr), "{%s,%s} ", phdr, rhdr); 434 435 va_start(ap, fmt); 436 audio_vtrace(file->sc, funcname, hdr, fmt, ap); 437 va_end(ap); 438 } 439 440 #define DPRINTF(n, fmt...) do { \ 441 if (audiodebug >= (n)) { \ 442 audio_mlog_flush(); \ 443 printf(fmt); \ 444 } \ 445 } while (0) 446 #define TRACE(n, fmt...) do { \ 447 if (audiodebug >= (n)) audio_trace(sc, __func__, fmt); \ 448 } while (0) 449 #define TRACET(n, t, fmt...) do { \ 450 if (audiodebug >= (n)) audio_tracet(__func__, t, fmt); \ 451 } while (0) 452 #define TRACEF(n, f, fmt...) do { \ 453 if (audiodebug >= (n)) audio_tracef(__func__, f, fmt); \ 454 } while (0) 455 456 struct audio_track_debugbuf { 457 char usrbuf[32]; 458 char codec[32]; 459 char chvol[32]; 460 char chmix[32]; 461 char freq[32]; 462 char outbuf[32]; 463 }; 464 465 static void 466 audio_track_bufstat(audio_track_t *track, struct audio_track_debugbuf *buf) 467 { 468 469 memset(buf, 0, sizeof(*buf)); 470 471 snprintf(buf->outbuf, sizeof(buf->outbuf), " out=%d/%d/%d", 472 track->outbuf.head, track->outbuf.used, track->outbuf.capacity); 473 if (track->freq.filter) 474 snprintf(buf->freq, sizeof(buf->freq), " f=%d/%d/%d", 475 track->freq.srcbuf.head, 476 track->freq.srcbuf.used, 477 track->freq.srcbuf.capacity); 478 if (track->chmix.filter) 479 snprintf(buf->chmix, sizeof(buf->chmix), " m=%d", 480 track->chmix.srcbuf.used); 481 if (track->chvol.filter) 482 snprintf(buf->chvol, sizeof(buf->chvol), " v=%d", 483 track->chvol.srcbuf.used); 484 if (track->codec.filter) 485 snprintf(buf->codec, sizeof(buf->codec), " e=%d", 486 track->codec.srcbuf.used); 487 snprintf(buf->usrbuf, sizeof(buf->usrbuf), " usr=%d/%d/H%d", 488 track->usrbuf.head, track->usrbuf.used, track->usrbuf_usedhigh); 489 } 490 #else 491 #define DPRINTF(n, fmt...) do { } while (0) 492 #define TRACE(n, fmt, ...) do { } while (0) 493 #define TRACET(n, t, fmt, ...) do { } while (0) 494 #define TRACEF(n, f, fmt, ...) do { } while (0) 495 #endif 496 497 #define SPECIFIED(x) ((x) != ~0) 498 #define SPECIFIED_CH(x) ((x) != (u_char)~0) 499 500 /* 501 * Default hardware blocksize in msec. 502 * 503 * We use 10 msec for most modern platforms. This period is good enough to 504 * play audio and video synchronizely. 505 * In contrast, for very old platforms, this is usually too short and too 506 * severe. Also such platforms usually can not play video confortably, so 507 * it's not so important to make the blocksize shorter. If the platform 508 * defines its own value as __AUDIO_BLK_MS in its <machine/param.h>, it 509 * uses this instead. 510 * 511 * In either case, you can overwrite AUDIO_BLK_MS by your kernel 512 * configuration file if you wish. 513 */ 514 #if !defined(AUDIO_BLK_MS) 515 # if defined(__AUDIO_BLK_MS) 516 # define AUDIO_BLK_MS __AUDIO_BLK_MS 517 # else 518 # define AUDIO_BLK_MS (10) 519 # endif 520 #endif 521 522 /* Device timeout in msec */ 523 #define AUDIO_TIMEOUT (3000) 524 525 /* #define AUDIO_PM_IDLE */ 526 #ifdef AUDIO_PM_IDLE 527 int audio_idle_timeout = 30; 528 #endif 529 530 /* Number of elements of async mixer's pid */ 531 #define AM_CAPACITY (4) 532 533 struct portname { 534 const char *name; 535 int mask; 536 }; 537 538 static int audiomatch(device_t, cfdata_t, void *); 539 static void audioattach(device_t, device_t, void *); 540 static int audiodetach(device_t, int); 541 static int audioactivate(device_t, enum devact); 542 static void audiochilddet(device_t, device_t); 543 static int audiorescan(device_t, const char *, const int *); 544 545 static int audio_modcmd(modcmd_t, void *); 546 547 #ifdef AUDIO_PM_IDLE 548 static void audio_idle(void *); 549 static void audio_activity(device_t, devactive_t); 550 #endif 551 552 static bool audio_suspend(device_t dv, const pmf_qual_t *); 553 static bool audio_resume(device_t dv, const pmf_qual_t *); 554 static void audio_volume_down(device_t); 555 static void audio_volume_up(device_t); 556 static void audio_volume_toggle(device_t); 557 558 static void audio_mixer_capture(struct audio_softc *); 559 static void audio_mixer_restore(struct audio_softc *); 560 561 static void audio_softintr_rd(void *); 562 static void audio_softintr_wr(void *); 563 564 static int audio_properties(struct audio_softc *); 565 static void audio_printf(struct audio_softc *, const char *, ...) 566 __printflike(2, 3); 567 static int audio_exlock_mutex_enter(struct audio_softc *); 568 static void audio_exlock_mutex_exit(struct audio_softc *); 569 static int audio_exlock_enter(struct audio_softc *); 570 static void audio_exlock_exit(struct audio_softc *); 571 static struct audio_softc *audio_sc_acquire_fromfile(audio_file_t *, 572 struct psref *); 573 static void audio_sc_release(struct audio_softc *, struct psref *); 574 static int audio_track_waitio(struct audio_softc *, audio_track_t *, 575 const char *mess); 576 577 static int audioclose(struct file *); 578 static int audioread(struct file *, off_t *, struct uio *, kauth_cred_t, int); 579 static int audiowrite(struct file *, off_t *, struct uio *, kauth_cred_t, int); 580 static int audioioctl(struct file *, u_long, void *); 581 static int audiopoll(struct file *, int); 582 static int audiokqfilter(struct file *, struct knote *); 583 static int audiommap(struct file *, off_t *, size_t, int, int *, int *, 584 struct uvm_object **, int *); 585 static int audiostat(struct file *, struct stat *); 586 587 static void filt_audiowrite_detach(struct knote *); 588 static int filt_audiowrite_event(struct knote *, long); 589 static void filt_audioread_detach(struct knote *); 590 static int filt_audioread_event(struct knote *, long); 591 592 static int audio_open(dev_t, struct audio_softc *, int, int, struct lwp *, 593 audio_file_t **); 594 static int audio_close(struct audio_softc *, audio_file_t *); 595 static void audio_unlink(struct audio_softc *, audio_file_t *); 596 static int audio_read(struct audio_softc *, struct uio *, int, audio_file_t *); 597 static int audio_write(struct audio_softc *, struct uio *, int, audio_file_t *); 598 static void audio_file_clear(struct audio_softc *, audio_file_t *); 599 static int audio_ioctl(dev_t, struct audio_softc *, u_long, void *, int, 600 struct lwp *, audio_file_t *); 601 static int audio_poll(struct audio_softc *, int, struct lwp *, audio_file_t *); 602 static int audio_kqfilter(struct audio_softc *, audio_file_t *, struct knote *); 603 static int audio_mmap(struct audio_softc *, off_t *, size_t, int, int *, int *, 604 struct uvm_object **, int *, audio_file_t *); 605 606 static int audioctl_open(dev_t, struct audio_softc *, int, int, struct lwp *); 607 608 static void audio_pintr(void *); 609 static void audio_rintr(void *); 610 611 static int audio_query_devinfo(struct audio_softc *, mixer_devinfo_t *); 612 613 static int audio_track_inputblk_as_usrbyte(const audio_track_t *, int); 614 static int audio_track_readablebytes(const audio_track_t *); 615 static int audio_file_setinfo(struct audio_softc *, audio_file_t *, 616 const struct audio_info *); 617 static int audio_track_setinfo_check(audio_track_t *, 618 audio_format2_t *, const struct audio_prinfo *); 619 static void audio_track_setinfo_water(audio_track_t *, 620 const struct audio_info *); 621 static int audio_hw_setinfo(struct audio_softc *, const struct audio_info *, 622 struct audio_info *); 623 static int audio_hw_set_format(struct audio_softc *, int, 624 const audio_format2_t *, const audio_format2_t *, 625 audio_filter_reg_t *, audio_filter_reg_t *); 626 static int audiogetinfo(struct audio_softc *, struct audio_info *, int, 627 audio_file_t *); 628 static bool audio_can_playback(struct audio_softc *); 629 static bool audio_can_capture(struct audio_softc *); 630 static int audio_check_params(audio_format2_t *); 631 static int audio_mixers_init(struct audio_softc *sc, int, 632 const audio_format2_t *, const audio_format2_t *, 633 const audio_filter_reg_t *, const audio_filter_reg_t *); 634 static int audio_select_freq(const struct audio_format *); 635 static int audio_hw_probe(struct audio_softc *, audio_format2_t *, int); 636 static int audio_hw_validate_format(struct audio_softc *, int, 637 const audio_format2_t *); 638 static int audio_mixers_set_format(struct audio_softc *, 639 const struct audio_info *); 640 static void audio_mixers_get_format(struct audio_softc *, struct audio_info *); 641 static int audio_sysctl_blk_ms(SYSCTLFN_PROTO); 642 static int audio_sysctl_multiuser(SYSCTLFN_PROTO); 643 #if defined(AUDIO_DEBUG) 644 static int audio_sysctl_debug(SYSCTLFN_PROTO); 645 static void audio_format2_tostr(char *, size_t, const audio_format2_t *); 646 static void audio_print_format2(const char *, const audio_format2_t *) __unused; 647 #endif 648 649 static void *audio_realloc(void *, size_t); 650 static void audio_free_usrbuf(audio_track_t *); 651 652 static audio_track_t *audio_track_create(struct audio_softc *, 653 audio_trackmixer_t *); 654 static void audio_track_destroy(audio_track_t *); 655 static audio_filter_t audio_track_get_codec(audio_track_t *, 656 const audio_format2_t *, const audio_format2_t *); 657 static int audio_track_set_format(audio_track_t *, audio_format2_t *); 658 static void audio_track_play(audio_track_t *); 659 static int audio_track_drain(struct audio_softc *, audio_track_t *); 660 static void audio_track_record(audio_track_t *); 661 static void audio_track_clear(struct audio_softc *, audio_track_t *); 662 663 static int audio_mixer_init(struct audio_softc *, int, 664 const audio_format2_t *, const audio_filter_reg_t *); 665 static void audio_mixer_destroy(struct audio_softc *, audio_trackmixer_t *); 666 static void audio_pmixer_start(struct audio_softc *, bool); 667 static void audio_pmixer_process(struct audio_softc *); 668 static void audio_pmixer_agc(audio_trackmixer_t *, int); 669 static int audio_pmixer_mix_track(audio_trackmixer_t *, audio_track_t *, int); 670 static void audio_pmixer_output(struct audio_softc *); 671 static int audio_pmixer_halt(struct audio_softc *); 672 static void audio_rmixer_start(struct audio_softc *); 673 static void audio_rmixer_process(struct audio_softc *); 674 static void audio_rmixer_input(struct audio_softc *); 675 static int audio_rmixer_halt(struct audio_softc *); 676 677 static void mixer_init(struct audio_softc *); 678 static int mixer_open(dev_t, struct audio_softc *, int, int, struct lwp *); 679 static int mixer_close(struct audio_softc *, audio_file_t *); 680 static int mixer_ioctl(struct audio_softc *, u_long, void *, int, struct lwp *); 681 static void mixer_async_add(struct audio_softc *, pid_t); 682 static void mixer_async_remove(struct audio_softc *, pid_t); 683 static void mixer_signal(struct audio_softc *); 684 685 static int au_portof(struct audio_softc *, char *, int); 686 687 static void au_setup_ports(struct audio_softc *, struct au_mixer_ports *, 688 mixer_devinfo_t *, const struct portname *); 689 static int au_set_lr_value(struct audio_softc *, mixer_ctrl_t *, int, int); 690 static int au_get_lr_value(struct audio_softc *, mixer_ctrl_t *, int *, int *); 691 static int au_set_gain(struct audio_softc *, struct au_mixer_ports *, int, int); 692 static void au_get_gain(struct audio_softc *, struct au_mixer_ports *, 693 u_int *, u_char *); 694 static int au_set_port(struct audio_softc *, struct au_mixer_ports *, u_int); 695 static int au_get_port(struct audio_softc *, struct au_mixer_ports *); 696 static int au_set_monitor_gain(struct audio_softc *, int); 697 static int au_get_monitor_gain(struct audio_softc *); 698 static int audio_get_port(struct audio_softc *, mixer_ctrl_t *); 699 static int audio_set_port(struct audio_softc *, mixer_ctrl_t *); 700 701 void audio_mixsample_to_linear(audio_filter_arg_t *); 702 703 static __inline struct audio_params 704 format2_to_params(const audio_format2_t *f2) 705 { 706 audio_params_t p; 707 708 /* validbits/precision <-> precision/stride */ 709 p.sample_rate = f2->sample_rate; 710 p.channels = f2->channels; 711 p.encoding = f2->encoding; 712 p.validbits = f2->precision; 713 p.precision = f2->stride; 714 return p; 715 } 716 717 static __inline audio_format2_t 718 params_to_format2(const struct audio_params *p) 719 { 720 audio_format2_t f2; 721 722 /* precision/stride <-> validbits/precision */ 723 f2.sample_rate = p->sample_rate; 724 f2.channels = p->channels; 725 f2.encoding = p->encoding; 726 f2.precision = p->validbits; 727 f2.stride = p->precision; 728 return f2; 729 } 730 731 /* Return true if this track is a playback track. */ 732 static __inline bool 733 audio_track_is_playback(const audio_track_t *track) 734 { 735 736 return ((track->mode & AUMODE_PLAY) != 0); 737 } 738 739 #if 0 740 /* Return true if this track is a recording track. */ 741 static __inline bool 742 audio_track_is_record(const audio_track_t *track) 743 { 744 745 return ((track->mode & AUMODE_RECORD) != 0); 746 } 747 #endif 748 749 #if 0 /* XXX Not used yet */ 750 /* 751 * Convert 0..255 volume used in userland to internal presentation 0..256. 752 */ 753 static __inline u_int 754 audio_volume_to_inner(u_int v) 755 { 756 757 return v < 127 ? v : v + 1; 758 } 759 760 /* 761 * Convert 0..256 internal presentation to 0..255 volume used in userland. 762 */ 763 static __inline u_int 764 audio_volume_to_outer(u_int v) 765 { 766 767 return v < 127 ? v : v - 1; 768 } 769 #endif /* 0 */ 770 771 static dev_type_open(audioopen); 772 /* XXXMRG use more dev_type_xxx */ 773 774 static int 775 audiounit(dev_t dev) 776 { 777 778 return AUDIOUNIT(dev); 779 } 780 781 const struct cdevsw audio_cdevsw = { 782 .d_open = audioopen, 783 .d_close = noclose, 784 .d_read = noread, 785 .d_write = nowrite, 786 .d_ioctl = noioctl, 787 .d_stop = nostop, 788 .d_tty = notty, 789 .d_poll = nopoll, 790 .d_mmap = nommap, 791 .d_kqfilter = nokqfilter, 792 .d_discard = nodiscard, 793 .d_cfdriver = &audio_cd, 794 .d_devtounit = audiounit, 795 .d_flag = D_OTHER | D_MPSAFE 796 }; 797 798 const struct fileops audio_fileops = { 799 .fo_name = "audio", 800 .fo_read = audioread, 801 .fo_write = audiowrite, 802 .fo_ioctl = audioioctl, 803 .fo_fcntl = fnullop_fcntl, 804 .fo_stat = audiostat, 805 .fo_poll = audiopoll, 806 .fo_close = audioclose, 807 .fo_mmap = audiommap, 808 .fo_kqfilter = audiokqfilter, 809 .fo_restart = fnullop_restart 810 }; 811 812 /* The default audio mode: 8 kHz mono mu-law */ 813 static const struct audio_params audio_default = { 814 .sample_rate = 8000, 815 .encoding = AUDIO_ENCODING_ULAW, 816 .precision = 8, 817 .validbits = 8, 818 .channels = 1, 819 }; 820 821 static const char *encoding_names[] = { 822 "none", 823 AudioEmulaw, 824 AudioEalaw, 825 "pcm16", 826 "pcm8", 827 AudioEadpcm, 828 AudioEslinear_le, 829 AudioEslinear_be, 830 AudioEulinear_le, 831 AudioEulinear_be, 832 AudioEslinear, 833 AudioEulinear, 834 AudioEmpeg_l1_stream, 835 AudioEmpeg_l1_packets, 836 AudioEmpeg_l1_system, 837 AudioEmpeg_l2_stream, 838 AudioEmpeg_l2_packets, 839 AudioEmpeg_l2_system, 840 AudioEac3, 841 }; 842 843 /* 844 * Returns encoding name corresponding to AUDIO_ENCODING_*. 845 * Note that it may return a local buffer because it is mainly for debugging. 846 */ 847 const char * 848 audio_encoding_name(int encoding) 849 { 850 static char buf[16]; 851 852 if (0 <= encoding && encoding < __arraycount(encoding_names)) { 853 return encoding_names[encoding]; 854 } else { 855 snprintf(buf, sizeof(buf), "enc=%d", encoding); 856 return buf; 857 } 858 } 859 860 /* 861 * Supported encodings used by AUDIO_GETENC. 862 * index and flags are set by code. 863 * XXX is there any needs for SLINEAR_OE:>=16/ULINEAR_OE:>=16 ? 864 */ 865 static const audio_encoding_t audio_encodings[] = { 866 { 0, AudioEmulaw, AUDIO_ENCODING_ULAW, 8, 0 }, 867 { 0, AudioEalaw, AUDIO_ENCODING_ALAW, 8, 0 }, 868 { 0, AudioEslinear, AUDIO_ENCODING_SLINEAR, 8, 0 }, 869 { 0, AudioEulinear, AUDIO_ENCODING_ULINEAR, 8, 0 }, 870 { 0, AudioEslinear_le, AUDIO_ENCODING_SLINEAR_LE, 16, 0 }, 871 { 0, AudioEulinear_le, AUDIO_ENCODING_ULINEAR_LE, 16, 0 }, 872 { 0, AudioEslinear_be, AUDIO_ENCODING_SLINEAR_BE, 16, 0 }, 873 { 0, AudioEulinear_be, AUDIO_ENCODING_ULINEAR_BE, 16, 0 }, 874 #if defined(AUDIO_SUPPORT_LINEAR24) 875 { 0, AudioEslinear_le, AUDIO_ENCODING_SLINEAR_LE, 24, 0 }, 876 { 0, AudioEulinear_le, AUDIO_ENCODING_ULINEAR_LE, 24, 0 }, 877 { 0, AudioEslinear_be, AUDIO_ENCODING_SLINEAR_BE, 24, 0 }, 878 { 0, AudioEulinear_be, AUDIO_ENCODING_ULINEAR_BE, 24, 0 }, 879 #endif 880 { 0, AudioEslinear_le, AUDIO_ENCODING_SLINEAR_LE, 32, 0 }, 881 { 0, AudioEulinear_le, AUDIO_ENCODING_ULINEAR_LE, 32, 0 }, 882 { 0, AudioEslinear_be, AUDIO_ENCODING_SLINEAR_BE, 32, 0 }, 883 { 0, AudioEulinear_be, AUDIO_ENCODING_ULINEAR_BE, 32, 0 }, 884 }; 885 886 static const struct portname itable[] = { 887 { AudioNmicrophone, AUDIO_MICROPHONE }, 888 { AudioNline, AUDIO_LINE_IN }, 889 { AudioNcd, AUDIO_CD }, 890 { 0, 0 } 891 }; 892 static const struct portname otable[] = { 893 { AudioNspeaker, AUDIO_SPEAKER }, 894 { AudioNheadphone, AUDIO_HEADPHONE }, 895 { AudioNline, AUDIO_LINE_OUT }, 896 { 0, 0 } 897 }; 898 899 static struct psref_class *audio_psref_class __read_mostly; 900 901 CFATTACH_DECL3_NEW(audio, sizeof(struct audio_softc), 902 audiomatch, audioattach, audiodetach, audioactivate, audiorescan, 903 audiochilddet, DVF_DETACH_SHUTDOWN); 904 905 static int 906 audiomatch(device_t parent, cfdata_t match, void *aux) 907 { 908 struct audio_attach_args *sa; 909 910 sa = aux; 911 DPRINTF(1, "%s: type=%d sa=%p hw=%p\n", 912 __func__, sa->type, sa, sa->hwif); 913 return (sa->type == AUDIODEV_TYPE_AUDIO) ? 1 : 0; 914 } 915 916 static void 917 audioattach(device_t parent, device_t self, void *aux) 918 { 919 struct audio_softc *sc; 920 struct audio_attach_args *sa; 921 const struct audio_hw_if *hw_if; 922 audio_format2_t phwfmt; 923 audio_format2_t rhwfmt; 924 audio_filter_reg_t pfil; 925 audio_filter_reg_t rfil; 926 const struct sysctlnode *node; 927 void *hdlp; 928 bool has_playback; 929 bool has_capture; 930 bool has_indep; 931 bool has_fulldup; 932 int mode; 933 int error; 934 935 sc = device_private(self); 936 sc->sc_dev = self; 937 sa = (struct audio_attach_args *)aux; 938 hw_if = sa->hwif; 939 hdlp = sa->hdl; 940 941 if (hw_if == NULL) { 942 panic("audioattach: missing hw_if method"); 943 } 944 if (hw_if->get_locks == NULL || hw_if->get_props == NULL) { 945 aprint_error(": missing mandatory method\n"); 946 return; 947 } 948 949 hw_if->get_locks(hdlp, &sc->sc_intr_lock, &sc->sc_lock); 950 sc->sc_props = hw_if->get_props(hdlp); 951 952 has_playback = (sc->sc_props & AUDIO_PROP_PLAYBACK); 953 has_capture = (sc->sc_props & AUDIO_PROP_CAPTURE); 954 has_indep = (sc->sc_props & AUDIO_PROP_INDEPENDENT); 955 has_fulldup = (sc->sc_props & AUDIO_PROP_FULLDUPLEX); 956 957 #ifdef DIAGNOSTIC 958 if (hw_if->query_format == NULL || 959 hw_if->set_format == NULL || 960 hw_if->getdev == NULL || 961 hw_if->set_port == NULL || 962 hw_if->get_port == NULL || 963 hw_if->query_devinfo == NULL) { 964 aprint_error(": missing mandatory method\n"); 965 return; 966 } 967 if (has_playback) { 968 if ((hw_if->start_output == NULL && 969 hw_if->trigger_output == NULL) || 970 hw_if->halt_output == NULL) { 971 aprint_error(": missing playback method\n"); 972 } 973 } 974 if (has_capture) { 975 if ((hw_if->start_input == NULL && 976 hw_if->trigger_input == NULL) || 977 hw_if->halt_input == NULL) { 978 aprint_error(": missing capture method\n"); 979 } 980 } 981 #endif 982 983 sc->hw_if = hw_if; 984 sc->hw_hdl = hdlp; 985 sc->hw_dev = parent; 986 987 sc->sc_exlock = 1; 988 sc->sc_blk_ms = AUDIO_BLK_MS; 989 SLIST_INIT(&sc->sc_files); 990 cv_init(&sc->sc_exlockcv, "audiolk"); 991 sc->sc_am_capacity = 0; 992 sc->sc_am_used = 0; 993 sc->sc_am = NULL; 994 995 /* MMAP is now supported by upper layer. */ 996 sc->sc_props |= AUDIO_PROP_MMAP; 997 998 KASSERT(has_playback || has_capture); 999 /* Unidirectional device must have neither FULLDUP nor INDEPENDENT. */ 1000 if (!has_playback || !has_capture) { 1001 KASSERT(!has_indep); 1002 KASSERT(!has_fulldup); 1003 } 1004 1005 mode = 0; 1006 if (has_playback) { 1007 aprint_normal(": playback"); 1008 mode |= AUMODE_PLAY; 1009 } 1010 if (has_capture) { 1011 aprint_normal("%c capture", has_playback ? ',' : ':'); 1012 mode |= AUMODE_RECORD; 1013 } 1014 if (has_playback && has_capture) { 1015 if (has_fulldup) 1016 aprint_normal(", full duplex"); 1017 else 1018 aprint_normal(", half duplex"); 1019 1020 if (has_indep) 1021 aprint_normal(", independent"); 1022 } 1023 1024 aprint_naive("\n"); 1025 aprint_normal("\n"); 1026 1027 /* probe hw params */ 1028 memset(&phwfmt, 0, sizeof(phwfmt)); 1029 memset(&rhwfmt, 0, sizeof(rhwfmt)); 1030 memset(&pfil, 0, sizeof(pfil)); 1031 memset(&rfil, 0, sizeof(rfil)); 1032 if (has_indep) { 1033 int perror, rerror; 1034 1035 /* On independent devices, probe separately. */ 1036 perror = audio_hw_probe(sc, &phwfmt, AUMODE_PLAY); 1037 rerror = audio_hw_probe(sc, &rhwfmt, AUMODE_RECORD); 1038 if (perror && rerror) { 1039 aprint_error_dev(self, 1040 "audio_hw_probe failed: perror=%d, rerror=%d\n", 1041 perror, rerror); 1042 goto bad; 1043 } 1044 if (perror) { 1045 mode &= ~AUMODE_PLAY; 1046 aprint_error_dev(self, "audio_hw_probe failed: " 1047 "errno=%d, playback disabled\n", perror); 1048 } 1049 if (rerror) { 1050 mode &= ~AUMODE_RECORD; 1051 aprint_error_dev(self, "audio_hw_probe failed: " 1052 "errno=%d, capture disabled\n", rerror); 1053 } 1054 } else { 1055 /* 1056 * On non independent devices or uni-directional devices, 1057 * probe once (simultaneously). 1058 */ 1059 audio_format2_t *fmt = has_playback ? &phwfmt : &rhwfmt; 1060 error = audio_hw_probe(sc, fmt, mode); 1061 if (error) { 1062 aprint_error_dev(self, 1063 "audio_hw_probe failed: errno=%d\n", error); 1064 goto bad; 1065 } 1066 if (has_playback && has_capture) 1067 rhwfmt = phwfmt; 1068 } 1069 1070 /* Make device id available */ 1071 if (audio_properties(sc)) 1072 aprint_error_dev(self, "audio_properties failed\n"); 1073 1074 /* Init hardware. */ 1075 /* hw_probe() also validates [pr]hwfmt. */ 1076 error = audio_hw_set_format(sc, mode, &phwfmt, &rhwfmt, &pfil, &rfil); 1077 if (error) { 1078 aprint_error_dev(self, 1079 "audio_hw_set_format failed: errno=%d\n", error); 1080 goto bad; 1081 } 1082 1083 /* 1084 * Init track mixers. If at least one direction is available on 1085 * attach time, we assume a success. 1086 */ 1087 error = audio_mixers_init(sc, mode, &phwfmt, &rhwfmt, &pfil, &rfil); 1088 if (sc->sc_pmixer == NULL && sc->sc_rmixer == NULL) { 1089 aprint_error_dev(self, 1090 "audio_mixers_init failed: errno=%d\n", error); 1091 goto bad; 1092 } 1093 1094 sc->sc_psz = pserialize_create(); 1095 psref_target_init(&sc->sc_psref, audio_psref_class); 1096 1097 selinit(&sc->sc_wsel); 1098 selinit(&sc->sc_rsel); 1099 1100 /* Initial parameter of /dev/sound */ 1101 sc->sc_sound_pparams = params_to_format2(&audio_default); 1102 sc->sc_sound_rparams = params_to_format2(&audio_default); 1103 sc->sc_sound_ppause = false; 1104 sc->sc_sound_rpause = false; 1105 1106 /* XXX TODO: consider about sc_ai */ 1107 1108 mixer_init(sc); 1109 TRACE(2, "inputs ports=0x%x, input master=%d, " 1110 "output ports=0x%x, output master=%d", 1111 sc->sc_inports.allports, sc->sc_inports.master, 1112 sc->sc_outports.allports, sc->sc_outports.master); 1113 1114 sysctl_createv(&sc->sc_log, 0, NULL, &node, 1115 0, 1116 CTLTYPE_NODE, device_xname(sc->sc_dev), 1117 SYSCTL_DESCR("audio test"), 1118 NULL, 0, 1119 NULL, 0, 1120 CTL_HW, 1121 CTL_CREATE, CTL_EOL); 1122 1123 if (node != NULL) { 1124 sysctl_createv(&sc->sc_log, 0, NULL, NULL, 1125 CTLFLAG_READWRITE, 1126 CTLTYPE_INT, "blk_ms", 1127 SYSCTL_DESCR("blocksize in msec"), 1128 audio_sysctl_blk_ms, 0, (void *)sc, 0, 1129 CTL_HW, node->sysctl_num, CTL_CREATE, CTL_EOL); 1130 1131 sysctl_createv(&sc->sc_log, 0, NULL, NULL, 1132 CTLFLAG_READWRITE, 1133 CTLTYPE_BOOL, "multiuser", 1134 SYSCTL_DESCR("allow multiple user access"), 1135 audio_sysctl_multiuser, 0, (void *)sc, 0, 1136 CTL_HW, node->sysctl_num, CTL_CREATE, CTL_EOL); 1137 1138 #if defined(AUDIO_DEBUG) 1139 sysctl_createv(&sc->sc_log, 0, NULL, NULL, 1140 CTLFLAG_READWRITE, 1141 CTLTYPE_INT, "debug", 1142 SYSCTL_DESCR("debug level (0..4)"), 1143 audio_sysctl_debug, 0, (void *)sc, 0, 1144 CTL_HW, node->sysctl_num, CTL_CREATE, CTL_EOL); 1145 #endif 1146 } 1147 1148 #ifdef AUDIO_PM_IDLE 1149 callout_init(&sc->sc_idle_counter, 0); 1150 callout_setfunc(&sc->sc_idle_counter, audio_idle, self); 1151 #endif 1152 1153 if (!pmf_device_register(self, audio_suspend, audio_resume)) 1154 aprint_error_dev(self, "couldn't establish power handler\n"); 1155 #ifdef AUDIO_PM_IDLE 1156 if (!device_active_register(self, audio_activity)) 1157 aprint_error_dev(self, "couldn't register activity handler\n"); 1158 #endif 1159 1160 if (!pmf_event_register(self, PMFE_AUDIO_VOLUME_DOWN, 1161 audio_volume_down, true)) 1162 aprint_error_dev(self, "couldn't add volume down handler\n"); 1163 if (!pmf_event_register(self, PMFE_AUDIO_VOLUME_UP, 1164 audio_volume_up, true)) 1165 aprint_error_dev(self, "couldn't add volume up handler\n"); 1166 if (!pmf_event_register(self, PMFE_AUDIO_VOLUME_TOGGLE, 1167 audio_volume_toggle, true)) 1168 aprint_error_dev(self, "couldn't add volume toggle handler\n"); 1169 1170 #ifdef AUDIO_PM_IDLE 1171 callout_schedule(&sc->sc_idle_counter, audio_idle_timeout * hz); 1172 #endif 1173 1174 #if defined(AUDIO_DEBUG) 1175 audio_mlog_init(); 1176 #endif 1177 1178 audiorescan(self, NULL, NULL); 1179 sc->sc_exlock = 0; 1180 return; 1181 1182 bad: 1183 /* Clearing hw_if means that device is attached but disabled. */ 1184 sc->hw_if = NULL; 1185 sc->sc_exlock = 0; 1186 aprint_error_dev(sc->sc_dev, "disabled\n"); 1187 return; 1188 } 1189 1190 /* 1191 * Identify audio backend device for drvctl. 1192 */ 1193 static int 1194 audio_properties(struct audio_softc *sc) 1195 { 1196 prop_dictionary_t dict = device_properties(sc->sc_dev); 1197 audio_device_t adev; 1198 int error; 1199 1200 error = sc->hw_if->getdev(sc->hw_hdl, &adev); 1201 if (error) 1202 return error; 1203 1204 prop_dictionary_set_string(dict, "name", adev.name); 1205 prop_dictionary_set_string(dict, "version", adev.version); 1206 prop_dictionary_set_string(dict, "config", adev.config); 1207 1208 return 0; 1209 } 1210 1211 /* 1212 * Initialize hardware mixer. 1213 * This function is called from audioattach(). 1214 */ 1215 static void 1216 mixer_init(struct audio_softc *sc) 1217 { 1218 mixer_devinfo_t mi; 1219 int iclass, mclass, oclass, rclass; 1220 int record_master_found, record_source_found; 1221 1222 iclass = mclass = oclass = rclass = -1; 1223 sc->sc_inports.index = -1; 1224 sc->sc_inports.master = -1; 1225 sc->sc_inports.nports = 0; 1226 sc->sc_inports.isenum = false; 1227 sc->sc_inports.allports = 0; 1228 sc->sc_inports.isdual = false; 1229 sc->sc_inports.mixerout = -1; 1230 sc->sc_inports.cur_port = -1; 1231 sc->sc_outports.index = -1; 1232 sc->sc_outports.master = -1; 1233 sc->sc_outports.nports = 0; 1234 sc->sc_outports.isenum = false; 1235 sc->sc_outports.allports = 0; 1236 sc->sc_outports.isdual = false; 1237 sc->sc_outports.mixerout = -1; 1238 sc->sc_outports.cur_port = -1; 1239 sc->sc_monitor_port = -1; 1240 /* 1241 * Read through the underlying driver's list, picking out the class 1242 * names from the mixer descriptions. We'll need them to decode the 1243 * mixer descriptions on the next pass through the loop. 1244 */ 1245 mutex_enter(sc->sc_lock); 1246 for(mi.index = 0; ; mi.index++) { 1247 if (audio_query_devinfo(sc, &mi) != 0) 1248 break; 1249 /* 1250 * The type of AUDIO_MIXER_CLASS merely introduces a class. 1251 * All the other types describe an actual mixer. 1252 */ 1253 if (mi.type == AUDIO_MIXER_CLASS) { 1254 if (strcmp(mi.label.name, AudioCinputs) == 0) 1255 iclass = mi.mixer_class; 1256 if (strcmp(mi.label.name, AudioCmonitor) == 0) 1257 mclass = mi.mixer_class; 1258 if (strcmp(mi.label.name, AudioCoutputs) == 0) 1259 oclass = mi.mixer_class; 1260 if (strcmp(mi.label.name, AudioCrecord) == 0) 1261 rclass = mi.mixer_class; 1262 } 1263 } 1264 mutex_exit(sc->sc_lock); 1265 1266 /* Allocate save area. Ensure non-zero allocation. */ 1267 sc->sc_nmixer_states = mi.index; 1268 sc->sc_mixer_state = kmem_zalloc(sizeof(sc->sc_mixer_state[0]) * 1269 (sc->sc_nmixer_states + 1), KM_SLEEP); 1270 1271 /* 1272 * This is where we assign each control in the "audio" model, to the 1273 * underlying "mixer" control. We walk through the whole list once, 1274 * assigning likely candidates as we come across them. 1275 */ 1276 record_master_found = 0; 1277 record_source_found = 0; 1278 mutex_enter(sc->sc_lock); 1279 for(mi.index = 0; ; mi.index++) { 1280 if (audio_query_devinfo(sc, &mi) != 0) 1281 break; 1282 KASSERT(mi.index < sc->sc_nmixer_states); 1283 if (mi.type == AUDIO_MIXER_CLASS) 1284 continue; 1285 if (mi.mixer_class == iclass) { 1286 /* 1287 * AudioCinputs is only a fallback, when we don't 1288 * find what we're looking for in AudioCrecord, so 1289 * check the flags before accepting one of these. 1290 */ 1291 if (strcmp(mi.label.name, AudioNmaster) == 0 1292 && record_master_found == 0) 1293 sc->sc_inports.master = mi.index; 1294 if (strcmp(mi.label.name, AudioNsource) == 0 1295 && record_source_found == 0) { 1296 if (mi.type == AUDIO_MIXER_ENUM) { 1297 int i; 1298 for(i = 0; i < mi.un.e.num_mem; i++) 1299 if (strcmp(mi.un.e.member[i].label.name, 1300 AudioNmixerout) == 0) 1301 sc->sc_inports.mixerout = 1302 mi.un.e.member[i].ord; 1303 } 1304 au_setup_ports(sc, &sc->sc_inports, &mi, 1305 itable); 1306 } 1307 if (strcmp(mi.label.name, AudioNdac) == 0 && 1308 sc->sc_outports.master == -1) 1309 sc->sc_outports.master = mi.index; 1310 } else if (mi.mixer_class == mclass) { 1311 if (strcmp(mi.label.name, AudioNmonitor) == 0) 1312 sc->sc_monitor_port = mi.index; 1313 } else if (mi.mixer_class == oclass) { 1314 if (strcmp(mi.label.name, AudioNmaster) == 0) 1315 sc->sc_outports.master = mi.index; 1316 if (strcmp(mi.label.name, AudioNselect) == 0) 1317 au_setup_ports(sc, &sc->sc_outports, &mi, 1318 otable); 1319 } else if (mi.mixer_class == rclass) { 1320 /* 1321 * These are the preferred mixers for the audio record 1322 * controls, so set the flags here, but don't check. 1323 */ 1324 if (strcmp(mi.label.name, AudioNmaster) == 0) { 1325 sc->sc_inports.master = mi.index; 1326 record_master_found = 1; 1327 } 1328 #if 1 /* Deprecated. Use AudioNmaster. */ 1329 if (strcmp(mi.label.name, AudioNrecord) == 0) { 1330 sc->sc_inports.master = mi.index; 1331 record_master_found = 1; 1332 } 1333 if (strcmp(mi.label.name, AudioNvolume) == 0) { 1334 sc->sc_inports.master = mi.index; 1335 record_master_found = 1; 1336 } 1337 #endif 1338 if (strcmp(mi.label.name, AudioNsource) == 0) { 1339 if (mi.type == AUDIO_MIXER_ENUM) { 1340 int i; 1341 for(i = 0; i < mi.un.e.num_mem; i++) 1342 if (strcmp(mi.un.e.member[i].label.name, 1343 AudioNmixerout) == 0) 1344 sc->sc_inports.mixerout = 1345 mi.un.e.member[i].ord; 1346 } 1347 au_setup_ports(sc, &sc->sc_inports, &mi, 1348 itable); 1349 record_source_found = 1; 1350 } 1351 } 1352 } 1353 mutex_exit(sc->sc_lock); 1354 } 1355 1356 static int 1357 audioactivate(device_t self, enum devact act) 1358 { 1359 struct audio_softc *sc = device_private(self); 1360 1361 switch (act) { 1362 case DVACT_DEACTIVATE: 1363 mutex_enter(sc->sc_lock); 1364 sc->sc_dying = true; 1365 cv_broadcast(&sc->sc_exlockcv); 1366 mutex_exit(sc->sc_lock); 1367 return 0; 1368 default: 1369 return EOPNOTSUPP; 1370 } 1371 } 1372 1373 static int 1374 audiodetach(device_t self, int flags) 1375 { 1376 struct audio_softc *sc; 1377 struct audio_file *file; 1378 int maj, mn; 1379 int error; 1380 1381 sc = device_private(self); 1382 TRACE(2, "flags=%d", flags); 1383 1384 /* device is not initialized */ 1385 if (sc->hw_if == NULL) 1386 return 0; 1387 1388 /* Start draining existing accessors of the device. */ 1389 error = config_detach_children(self, flags); 1390 if (error) 1391 return error; 1392 1393 /* 1394 * Prevent new opens and wait for existing opens to complete. 1395 * 1396 * At the moment there are only four bits in the minor for the 1397 * unit number, so we only revoke if the unit number could be 1398 * used in a device node. 1399 * 1400 * XXX If we want more audio units, we need to encode them 1401 * more elaborately in the minor space. 1402 */ 1403 maj = cdevsw_lookup_major(&audio_cdevsw); 1404 mn = device_unit(self); 1405 if (mn <= 0xf) { 1406 vdevgone(maj, mn|SOUND_DEVICE, mn|SOUND_DEVICE, VCHR); 1407 vdevgone(maj, mn|AUDIO_DEVICE, mn|AUDIO_DEVICE, VCHR); 1408 vdevgone(maj, mn|AUDIOCTL_DEVICE, mn|AUDIOCTL_DEVICE, VCHR); 1409 vdevgone(maj, mn|MIXER_DEVICE, mn|MIXER_DEVICE, VCHR); 1410 } 1411 1412 /* 1413 * This waits currently running sysctls to finish if exists. 1414 * After this, no more new sysctls will come. 1415 */ 1416 sysctl_teardown(&sc->sc_log); 1417 1418 mutex_enter(sc->sc_lock); 1419 sc->sc_dying = true; 1420 cv_broadcast(&sc->sc_exlockcv); 1421 if (sc->sc_pmixer) 1422 cv_broadcast(&sc->sc_pmixer->outcv); 1423 if (sc->sc_rmixer) 1424 cv_broadcast(&sc->sc_rmixer->outcv); 1425 1426 /* Prevent new users */ 1427 SLIST_FOREACH(file, &sc->sc_files, entry) { 1428 atomic_store_relaxed(&file->dying, true); 1429 } 1430 mutex_exit(sc->sc_lock); 1431 1432 /* 1433 * Wait for existing users to drain. 1434 * - pserialize_perform waits for all pserialize_read sections on 1435 * all CPUs; after this, no more new psref_acquire can happen. 1436 * - psref_target_destroy waits for all extant acquired psrefs to 1437 * be psref_released. 1438 */ 1439 pserialize_perform(sc->sc_psz); 1440 psref_target_destroy(&sc->sc_psref, audio_psref_class); 1441 1442 /* 1443 * We are now guaranteed that there are no calls to audio fileops 1444 * that hold sc, and any new calls with files that were for sc will 1445 * fail. Thus, we now have exclusive access to the softc. 1446 */ 1447 sc->sc_exlock = 1; 1448 1449 /* 1450 * Clean up all open instances. 1451 */ 1452 mutex_enter(sc->sc_lock); 1453 while ((file = SLIST_FIRST(&sc->sc_files)) != NULL) { 1454 mutex_enter(sc->sc_intr_lock); 1455 SLIST_REMOVE_HEAD(&sc->sc_files, entry); 1456 mutex_exit(sc->sc_intr_lock); 1457 if (file->ptrack || file->rtrack) { 1458 mutex_exit(sc->sc_lock); 1459 audio_unlink(sc, file); 1460 mutex_enter(sc->sc_lock); 1461 } 1462 } 1463 mutex_exit(sc->sc_lock); 1464 1465 pmf_event_deregister(self, PMFE_AUDIO_VOLUME_DOWN, 1466 audio_volume_down, true); 1467 pmf_event_deregister(self, PMFE_AUDIO_VOLUME_UP, 1468 audio_volume_up, true); 1469 pmf_event_deregister(self, PMFE_AUDIO_VOLUME_TOGGLE, 1470 audio_volume_toggle, true); 1471 1472 #ifdef AUDIO_PM_IDLE 1473 callout_halt(&sc->sc_idle_counter, sc->sc_lock); 1474 1475 device_active_deregister(self, audio_activity); 1476 #endif 1477 1478 pmf_device_deregister(self); 1479 1480 /* Free resources */ 1481 if (sc->sc_pmixer) { 1482 audio_mixer_destroy(sc, sc->sc_pmixer); 1483 kmem_free(sc->sc_pmixer, sizeof(*sc->sc_pmixer)); 1484 } 1485 if (sc->sc_rmixer) { 1486 audio_mixer_destroy(sc, sc->sc_rmixer); 1487 kmem_free(sc->sc_rmixer, sizeof(*sc->sc_rmixer)); 1488 } 1489 if (sc->sc_am) 1490 kern_free(sc->sc_am); 1491 1492 seldestroy(&sc->sc_wsel); 1493 seldestroy(&sc->sc_rsel); 1494 1495 #ifdef AUDIO_PM_IDLE 1496 callout_destroy(&sc->sc_idle_counter); 1497 #endif 1498 1499 cv_destroy(&sc->sc_exlockcv); 1500 1501 #if defined(AUDIO_DEBUG) 1502 audio_mlog_free(); 1503 #endif 1504 1505 return 0; 1506 } 1507 1508 static void 1509 audiochilddet(device_t self, device_t child) 1510 { 1511 1512 /* we hold no child references, so do nothing */ 1513 } 1514 1515 static int 1516 audiosearch(device_t parent, cfdata_t cf, const int *locs, void *aux) 1517 { 1518 1519 if (config_probe(parent, cf, aux)) 1520 config_attach(parent, cf, aux, NULL, 1521 CFARGS_NONE); 1522 1523 return 0; 1524 } 1525 1526 static int 1527 audiorescan(device_t self, const char *ifattr, const int *locators) 1528 { 1529 struct audio_softc *sc = device_private(self); 1530 1531 config_search(sc->sc_dev, NULL, 1532 CFARGS(.search = audiosearch)); 1533 1534 return 0; 1535 } 1536 1537 /* 1538 * Called from hardware driver. This is where the MI audio driver gets 1539 * probed/attached to the hardware driver. 1540 */ 1541 device_t 1542 audio_attach_mi(const struct audio_hw_if *ahwp, void *hdlp, device_t dev) 1543 { 1544 struct audio_attach_args arg; 1545 1546 #ifdef DIAGNOSTIC 1547 if (ahwp == NULL) { 1548 aprint_error("audio_attach_mi: NULL\n"); 1549 return 0; 1550 } 1551 #endif 1552 arg.type = AUDIODEV_TYPE_AUDIO; 1553 arg.hwif = ahwp; 1554 arg.hdl = hdlp; 1555 return config_found(dev, &arg, audioprint, 1556 CFARGS(.iattr = "audiobus")); 1557 } 1558 1559 /* 1560 * audio_printf() outputs fmt... with the audio device name and MD device 1561 * name prefixed. If the message is considered to be related to the MD 1562 * driver, use this one instead of device_printf(). 1563 */ 1564 static void 1565 audio_printf(struct audio_softc *sc, const char *fmt, ...) 1566 { 1567 va_list ap; 1568 1569 printf("%s(%s): ", device_xname(sc->sc_dev), device_xname(sc->hw_dev)); 1570 va_start(ap, fmt); 1571 vprintf(fmt, ap); 1572 va_end(ap); 1573 } 1574 1575 /* 1576 * Enter critical section and also keep sc_lock. 1577 * If successful, returns 0 with sc_lock held. Otherwise returns errno. 1578 * Must be called without sc_lock held. 1579 */ 1580 static int 1581 audio_exlock_mutex_enter(struct audio_softc *sc) 1582 { 1583 int error; 1584 1585 mutex_enter(sc->sc_lock); 1586 if (sc->sc_dying) { 1587 mutex_exit(sc->sc_lock); 1588 return EIO; 1589 } 1590 1591 while (__predict_false(sc->sc_exlock != 0)) { 1592 error = cv_wait_sig(&sc->sc_exlockcv, sc->sc_lock); 1593 if (sc->sc_dying) 1594 error = EIO; 1595 if (error) { 1596 mutex_exit(sc->sc_lock); 1597 return error; 1598 } 1599 } 1600 1601 /* Acquire */ 1602 sc->sc_exlock = 1; 1603 return 0; 1604 } 1605 1606 /* 1607 * Exit critical section and exit sc_lock. 1608 * Must be called with sc_lock held. 1609 */ 1610 static void 1611 audio_exlock_mutex_exit(struct audio_softc *sc) 1612 { 1613 1614 KASSERT(mutex_owned(sc->sc_lock)); 1615 1616 sc->sc_exlock = 0; 1617 cv_broadcast(&sc->sc_exlockcv); 1618 mutex_exit(sc->sc_lock); 1619 } 1620 1621 /* 1622 * Enter critical section. 1623 * If successful, it returns 0. Otherwise returns errno. 1624 * Must be called without sc_lock held. 1625 * This function returns without sc_lock held. 1626 */ 1627 static int 1628 audio_exlock_enter(struct audio_softc *sc) 1629 { 1630 int error; 1631 1632 error = audio_exlock_mutex_enter(sc); 1633 if (error) 1634 return error; 1635 mutex_exit(sc->sc_lock); 1636 return 0; 1637 } 1638 1639 /* 1640 * Exit critical section. 1641 * Must be called without sc_lock held. 1642 */ 1643 static void 1644 audio_exlock_exit(struct audio_softc *sc) 1645 { 1646 1647 mutex_enter(sc->sc_lock); 1648 audio_exlock_mutex_exit(sc); 1649 } 1650 1651 /* 1652 * Get sc from file, and increment reference counter for this sc. 1653 * This is intended to be used for methods other than open. 1654 * If successful, returns sc. Otherwise returns NULL. 1655 */ 1656 struct audio_softc * 1657 audio_sc_acquire_fromfile(audio_file_t *file, struct psref *refp) 1658 { 1659 int s; 1660 bool dying; 1661 1662 /* Block audiodetach while we acquire a reference */ 1663 s = pserialize_read_enter(); 1664 1665 /* If close or audiodetach already ran, tough -- no more audio */ 1666 dying = atomic_load_relaxed(&file->dying); 1667 if (dying) { 1668 pserialize_read_exit(s); 1669 return NULL; 1670 } 1671 1672 /* Acquire a reference */ 1673 psref_acquire(refp, &file->sc->sc_psref, audio_psref_class); 1674 1675 /* Now sc won't go away until we drop the reference count */ 1676 pserialize_read_exit(s); 1677 1678 return file->sc; 1679 } 1680 1681 /* 1682 * Decrement reference counter for this sc. 1683 */ 1684 void 1685 audio_sc_release(struct audio_softc *sc, struct psref *refp) 1686 { 1687 1688 psref_release(refp, &sc->sc_psref, audio_psref_class); 1689 } 1690 1691 /* 1692 * Wait for I/O to complete, releasing sc_lock. 1693 * Must be called with sc_lock held. 1694 */ 1695 static int 1696 audio_track_waitio(struct audio_softc *sc, audio_track_t *track, 1697 const char *mess) 1698 { 1699 int error; 1700 1701 KASSERT(track); 1702 KASSERT(mutex_owned(sc->sc_lock)); 1703 1704 /* Wait for pending I/O to complete. */ 1705 error = cv_timedwait_sig(&track->mixer->outcv, sc->sc_lock, 1706 mstohz(AUDIO_TIMEOUT)); 1707 if (sc->sc_suspending) { 1708 /* If it's about to suspend, ignore timeout error. */ 1709 if (error == EWOULDBLOCK) { 1710 TRACET(2, track, "timeout (suspending)"); 1711 return 0; 1712 } 1713 } 1714 if (sc->sc_dying) { 1715 error = EIO; 1716 } 1717 if (error) { 1718 TRACET(2, track, "cv_timedwait_sig failed %d", error); 1719 if (error == EWOULDBLOCK) { 1720 audio_ring_t *usrbuf = &track->usrbuf; 1721 audio_ring_t *outbuf = &track->outbuf; 1722 audio_printf(sc, 1723 "%s: device timeout, seq=%d, usrbuf=%d/H%d, outbuf=%d/%d\n", 1724 mess, (int)track->seq, 1725 usrbuf->used, track->usrbuf_usedhigh, 1726 outbuf->used, outbuf->capacity); 1727 } 1728 } else { 1729 TRACET(3, track, "wakeup"); 1730 } 1731 return error; 1732 } 1733 1734 /* 1735 * Try to acquire track lock. 1736 * It doesn't block if the track lock is already acquired. 1737 * Returns true if the track lock was acquired, or false if the track 1738 * lock was already acquired. 1739 */ 1740 static __inline bool 1741 audio_track_lock_tryenter(audio_track_t *track) 1742 { 1743 1744 if (atomic_swap_uint(&track->lock, 1) != 0) 1745 return false; 1746 membar_acquire(); 1747 return true; 1748 } 1749 1750 /* 1751 * Acquire track lock. 1752 */ 1753 static __inline void 1754 audio_track_lock_enter(audio_track_t *track) 1755 { 1756 1757 /* Don't sleep here. */ 1758 while (audio_track_lock_tryenter(track) == false) 1759 SPINLOCK_BACKOFF_HOOK; 1760 } 1761 1762 /* 1763 * Release track lock. 1764 */ 1765 static __inline void 1766 audio_track_lock_exit(audio_track_t *track) 1767 { 1768 1769 atomic_store_release(&track->lock, 0); 1770 } 1771 1772 1773 static int 1774 audioopen(dev_t dev, int flags, int ifmt, struct lwp *l) 1775 { 1776 struct audio_softc *sc; 1777 int error; 1778 1779 /* 1780 * Find the device. Because we wired the cdevsw to the audio 1781 * autoconf instance, the system ensures it will not go away 1782 * until after we return. 1783 */ 1784 sc = device_lookup_private(&audio_cd, AUDIOUNIT(dev)); 1785 if (sc == NULL || sc->hw_if == NULL) 1786 return ENXIO; 1787 1788 error = audio_exlock_enter(sc); 1789 if (error) 1790 return error; 1791 1792 device_active(sc->sc_dev, DVA_SYSTEM); 1793 switch (AUDIODEV(dev)) { 1794 case SOUND_DEVICE: 1795 case AUDIO_DEVICE: 1796 error = audio_open(dev, sc, flags, ifmt, l, NULL); 1797 break; 1798 case AUDIOCTL_DEVICE: 1799 error = audioctl_open(dev, sc, flags, ifmt, l); 1800 break; 1801 case MIXER_DEVICE: 1802 error = mixer_open(dev, sc, flags, ifmt, l); 1803 break; 1804 default: 1805 error = ENXIO; 1806 break; 1807 } 1808 audio_exlock_exit(sc); 1809 1810 return error; 1811 } 1812 1813 static int 1814 audioclose(struct file *fp) 1815 { 1816 struct audio_softc *sc; 1817 struct psref sc_ref; 1818 audio_file_t *file; 1819 int bound; 1820 int error; 1821 dev_t dev; 1822 1823 KASSERT(fp->f_audioctx); 1824 file = fp->f_audioctx; 1825 dev = file->dev; 1826 error = 0; 1827 1828 /* 1829 * audioclose() must 1830 * - unplug track from the trackmixer (and unplug anything from softc), 1831 * if sc exists. 1832 * - free all memory objects, regardless of sc. 1833 */ 1834 1835 bound = curlwp_bind(); 1836 sc = audio_sc_acquire_fromfile(file, &sc_ref); 1837 if (sc) { 1838 switch (AUDIODEV(dev)) { 1839 case SOUND_DEVICE: 1840 case AUDIO_DEVICE: 1841 error = audio_close(sc, file); 1842 break; 1843 case AUDIOCTL_DEVICE: 1844 mutex_enter(sc->sc_lock); 1845 mutex_enter(sc->sc_intr_lock); 1846 SLIST_REMOVE(&sc->sc_files, file, audio_file, entry); 1847 mutex_exit(sc->sc_intr_lock); 1848 mutex_exit(sc->sc_lock); 1849 error = 0; 1850 break; 1851 case MIXER_DEVICE: 1852 mutex_enter(sc->sc_lock); 1853 mutex_enter(sc->sc_intr_lock); 1854 SLIST_REMOVE(&sc->sc_files, file, audio_file, entry); 1855 mutex_exit(sc->sc_intr_lock); 1856 mutex_exit(sc->sc_lock); 1857 error = mixer_close(sc, file); 1858 break; 1859 default: 1860 error = ENXIO; 1861 break; 1862 } 1863 1864 audio_sc_release(sc, &sc_ref); 1865 } 1866 curlwp_bindx(bound); 1867 1868 /* Free memory objects anyway */ 1869 TRACEF(2, file, "free memory"); 1870 if (file->ptrack) 1871 audio_track_destroy(file->ptrack); 1872 if (file->rtrack) 1873 audio_track_destroy(file->rtrack); 1874 kmem_free(file, sizeof(*file)); 1875 fp->f_audioctx = NULL; 1876 1877 return error; 1878 } 1879 1880 static int 1881 audioread(struct file *fp, off_t *offp, struct uio *uio, kauth_cred_t cred, 1882 int ioflag) 1883 { 1884 struct audio_softc *sc; 1885 struct psref sc_ref; 1886 audio_file_t *file; 1887 int bound; 1888 int error; 1889 dev_t dev; 1890 1891 KASSERT(fp->f_audioctx); 1892 file = fp->f_audioctx; 1893 dev = file->dev; 1894 1895 bound = curlwp_bind(); 1896 sc = audio_sc_acquire_fromfile(file, &sc_ref); 1897 if (sc == NULL) { 1898 error = EIO; 1899 goto done; 1900 } 1901 1902 if (fp->f_flag & O_NONBLOCK) 1903 ioflag |= IO_NDELAY; 1904 1905 switch (AUDIODEV(dev)) { 1906 case SOUND_DEVICE: 1907 case AUDIO_DEVICE: 1908 error = audio_read(sc, uio, ioflag, file); 1909 break; 1910 case AUDIOCTL_DEVICE: 1911 case MIXER_DEVICE: 1912 error = ENODEV; 1913 break; 1914 default: 1915 error = ENXIO; 1916 break; 1917 } 1918 1919 audio_sc_release(sc, &sc_ref); 1920 done: 1921 curlwp_bindx(bound); 1922 return error; 1923 } 1924 1925 static int 1926 audiowrite(struct file *fp, off_t *offp, struct uio *uio, kauth_cred_t cred, 1927 int ioflag) 1928 { 1929 struct audio_softc *sc; 1930 struct psref sc_ref; 1931 audio_file_t *file; 1932 int bound; 1933 int error; 1934 dev_t dev; 1935 1936 KASSERT(fp->f_audioctx); 1937 file = fp->f_audioctx; 1938 dev = file->dev; 1939 1940 bound = curlwp_bind(); 1941 sc = audio_sc_acquire_fromfile(file, &sc_ref); 1942 if (sc == NULL) { 1943 error = EIO; 1944 goto done; 1945 } 1946 1947 if (fp->f_flag & O_NONBLOCK) 1948 ioflag |= IO_NDELAY; 1949 1950 switch (AUDIODEV(dev)) { 1951 case SOUND_DEVICE: 1952 case AUDIO_DEVICE: 1953 error = audio_write(sc, uio, ioflag, file); 1954 break; 1955 case AUDIOCTL_DEVICE: 1956 case MIXER_DEVICE: 1957 error = ENODEV; 1958 break; 1959 default: 1960 error = ENXIO; 1961 break; 1962 } 1963 1964 audio_sc_release(sc, &sc_ref); 1965 done: 1966 curlwp_bindx(bound); 1967 return error; 1968 } 1969 1970 static int 1971 audioioctl(struct file *fp, u_long cmd, void *addr) 1972 { 1973 struct audio_softc *sc; 1974 struct psref sc_ref; 1975 audio_file_t *file; 1976 struct lwp *l = curlwp; 1977 int bound; 1978 int error; 1979 dev_t dev; 1980 1981 KASSERT(fp->f_audioctx); 1982 file = fp->f_audioctx; 1983 dev = file->dev; 1984 1985 bound = curlwp_bind(); 1986 sc = audio_sc_acquire_fromfile(file, &sc_ref); 1987 if (sc == NULL) { 1988 error = EIO; 1989 goto done; 1990 } 1991 1992 switch (AUDIODEV(dev)) { 1993 case SOUND_DEVICE: 1994 case AUDIO_DEVICE: 1995 case AUDIOCTL_DEVICE: 1996 mutex_enter(sc->sc_lock); 1997 device_active(sc->sc_dev, DVA_SYSTEM); 1998 mutex_exit(sc->sc_lock); 1999 if (IOCGROUP(cmd) == IOCGROUP(AUDIO_MIXER_READ)) 2000 error = mixer_ioctl(sc, cmd, addr, fp->f_flag, l); 2001 else 2002 error = audio_ioctl(dev, sc, cmd, addr, fp->f_flag, l, 2003 file); 2004 break; 2005 case MIXER_DEVICE: 2006 error = mixer_ioctl(sc, cmd, addr, fp->f_flag, l); 2007 break; 2008 default: 2009 error = ENXIO; 2010 break; 2011 } 2012 2013 audio_sc_release(sc, &sc_ref); 2014 done: 2015 curlwp_bindx(bound); 2016 return error; 2017 } 2018 2019 static int 2020 audiostat(struct file *fp, struct stat *st) 2021 { 2022 struct audio_softc *sc; 2023 struct psref sc_ref; 2024 audio_file_t *file; 2025 int bound; 2026 int error; 2027 2028 KASSERT(fp->f_audioctx); 2029 file = fp->f_audioctx; 2030 2031 bound = curlwp_bind(); 2032 sc = audio_sc_acquire_fromfile(file, &sc_ref); 2033 if (sc == NULL) { 2034 error = EIO; 2035 goto done; 2036 } 2037 2038 error = 0; 2039 memset(st, 0, sizeof(*st)); 2040 2041 st->st_dev = file->dev; 2042 st->st_uid = kauth_cred_geteuid(fp->f_cred); 2043 st->st_gid = kauth_cred_getegid(fp->f_cred); 2044 st->st_mode = S_IFCHR; 2045 2046 audio_sc_release(sc, &sc_ref); 2047 done: 2048 curlwp_bindx(bound); 2049 return error; 2050 } 2051 2052 static int 2053 audiopoll(struct file *fp, int events) 2054 { 2055 struct audio_softc *sc; 2056 struct psref sc_ref; 2057 audio_file_t *file; 2058 struct lwp *l = curlwp; 2059 int bound; 2060 int revents; 2061 dev_t dev; 2062 2063 KASSERT(fp->f_audioctx); 2064 file = fp->f_audioctx; 2065 dev = file->dev; 2066 2067 bound = curlwp_bind(); 2068 sc = audio_sc_acquire_fromfile(file, &sc_ref); 2069 if (sc == NULL) { 2070 revents = POLLERR; 2071 goto done; 2072 } 2073 2074 switch (AUDIODEV(dev)) { 2075 case SOUND_DEVICE: 2076 case AUDIO_DEVICE: 2077 revents = audio_poll(sc, events, l, file); 2078 break; 2079 case AUDIOCTL_DEVICE: 2080 case MIXER_DEVICE: 2081 revents = 0; 2082 break; 2083 default: 2084 revents = POLLERR; 2085 break; 2086 } 2087 2088 audio_sc_release(sc, &sc_ref); 2089 done: 2090 curlwp_bindx(bound); 2091 return revents; 2092 } 2093 2094 static int 2095 audiokqfilter(struct file *fp, struct knote *kn) 2096 { 2097 struct audio_softc *sc; 2098 struct psref sc_ref; 2099 audio_file_t *file; 2100 dev_t dev; 2101 int bound; 2102 int error; 2103 2104 KASSERT(fp->f_audioctx); 2105 file = fp->f_audioctx; 2106 dev = file->dev; 2107 2108 bound = curlwp_bind(); 2109 sc = audio_sc_acquire_fromfile(file, &sc_ref); 2110 if (sc == NULL) { 2111 error = EIO; 2112 goto done; 2113 } 2114 2115 switch (AUDIODEV(dev)) { 2116 case SOUND_DEVICE: 2117 case AUDIO_DEVICE: 2118 error = audio_kqfilter(sc, file, kn); 2119 break; 2120 case AUDIOCTL_DEVICE: 2121 case MIXER_DEVICE: 2122 error = ENODEV; 2123 break; 2124 default: 2125 error = ENXIO; 2126 break; 2127 } 2128 2129 audio_sc_release(sc, &sc_ref); 2130 done: 2131 curlwp_bindx(bound); 2132 return error; 2133 } 2134 2135 static int 2136 audiommap(struct file *fp, off_t *offp, size_t len, int prot, int *flagsp, 2137 int *advicep, struct uvm_object **uobjp, int *maxprotp) 2138 { 2139 struct audio_softc *sc; 2140 struct psref sc_ref; 2141 audio_file_t *file; 2142 dev_t dev; 2143 int bound; 2144 int error; 2145 2146 KASSERT(len > 0); 2147 2148 KASSERT(fp->f_audioctx); 2149 file = fp->f_audioctx; 2150 dev = file->dev; 2151 2152 bound = curlwp_bind(); 2153 sc = audio_sc_acquire_fromfile(file, &sc_ref); 2154 if (sc == NULL) { 2155 error = EIO; 2156 goto done; 2157 } 2158 2159 mutex_enter(sc->sc_lock); 2160 device_active(sc->sc_dev, DVA_SYSTEM); /* XXXJDM */ 2161 mutex_exit(sc->sc_lock); 2162 2163 switch (AUDIODEV(dev)) { 2164 case SOUND_DEVICE: 2165 case AUDIO_DEVICE: 2166 error = audio_mmap(sc, offp, len, prot, flagsp, advicep, 2167 uobjp, maxprotp, file); 2168 break; 2169 case AUDIOCTL_DEVICE: 2170 case MIXER_DEVICE: 2171 default: 2172 error = ENOTSUP; 2173 break; 2174 } 2175 2176 audio_sc_release(sc, &sc_ref); 2177 done: 2178 curlwp_bindx(bound); 2179 return error; 2180 } 2181 2182 2183 /* Exported interfaces for audiobell. */ 2184 2185 /* 2186 * Open for audiobell. 2187 * It stores allocated file to *filep. 2188 * If successful returns 0, otherwise errno. 2189 */ 2190 int 2191 audiobellopen(dev_t dev, audio_file_t **filep) 2192 { 2193 device_t audiodev = NULL; 2194 struct audio_softc *sc; 2195 bool exlock = false; 2196 int error; 2197 2198 /* 2199 * Find the autoconf instance and make sure it doesn't go away 2200 * while we are opening it. 2201 */ 2202 audiodev = device_lookup_acquire(&audio_cd, AUDIOUNIT(dev)); 2203 if (audiodev == NULL) { 2204 error = ENXIO; 2205 goto out; 2206 } 2207 2208 /* If attach failed, it's hopeless -- give up. */ 2209 sc = device_private(audiodev); 2210 if (sc->hw_if == NULL) { 2211 error = ENXIO; 2212 goto out; 2213 } 2214 2215 /* Take the exclusive configuration lock. */ 2216 error = audio_exlock_enter(sc); 2217 if (error) 2218 goto out; 2219 exlock = true; 2220 2221 /* Open the audio device. */ 2222 device_active(sc->sc_dev, DVA_SYSTEM); 2223 error = audio_open(dev, sc, FWRITE, 0, curlwp, filep); 2224 2225 out: if (exlock) 2226 audio_exlock_exit(sc); 2227 if (audiodev) 2228 device_release(audiodev); 2229 return error; 2230 } 2231 2232 /* Close for audiobell */ 2233 int 2234 audiobellclose(audio_file_t *file) 2235 { 2236 struct audio_softc *sc; 2237 struct psref sc_ref; 2238 int bound; 2239 int error; 2240 2241 error = 0; 2242 /* 2243 * audiobellclose() must 2244 * - unplug track from the trackmixer if sc exist. 2245 * - free all memory objects, regardless of sc. 2246 */ 2247 bound = curlwp_bind(); 2248 sc = audio_sc_acquire_fromfile(file, &sc_ref); 2249 if (sc) { 2250 error = audio_close(sc, file); 2251 audio_sc_release(sc, &sc_ref); 2252 } 2253 curlwp_bindx(bound); 2254 2255 /* Free memory objects anyway */ 2256 KASSERT(file->ptrack); 2257 audio_track_destroy(file->ptrack); 2258 KASSERT(file->rtrack == NULL); 2259 kmem_free(file, sizeof(*file)); 2260 return error; 2261 } 2262 2263 /* Set sample rate for audiobell */ 2264 int 2265 audiobellsetrate(audio_file_t *file, u_int sample_rate) 2266 { 2267 struct audio_softc *sc; 2268 struct psref sc_ref; 2269 struct audio_info ai; 2270 int bound; 2271 int error; 2272 2273 bound = curlwp_bind(); 2274 sc = audio_sc_acquire_fromfile(file, &sc_ref); 2275 if (sc == NULL) { 2276 error = EIO; 2277 goto done1; 2278 } 2279 2280 AUDIO_INITINFO(&ai); 2281 ai.play.sample_rate = sample_rate; 2282 2283 error = audio_exlock_enter(sc); 2284 if (error) 2285 goto done2; 2286 error = audio_file_setinfo(sc, file, &ai); 2287 audio_exlock_exit(sc); 2288 2289 done2: 2290 audio_sc_release(sc, &sc_ref); 2291 done1: 2292 curlwp_bindx(bound); 2293 return error; 2294 } 2295 2296 /* Playback for audiobell */ 2297 int 2298 audiobellwrite(audio_file_t *file, struct uio *uio) 2299 { 2300 struct audio_softc *sc; 2301 struct psref sc_ref; 2302 int bound; 2303 int error; 2304 2305 bound = curlwp_bind(); 2306 sc = audio_sc_acquire_fromfile(file, &sc_ref); 2307 if (sc == NULL) { 2308 error = EIO; 2309 goto done; 2310 } 2311 2312 error = audio_write(sc, uio, 0, file); 2313 2314 audio_sc_release(sc, &sc_ref); 2315 done: 2316 curlwp_bindx(bound); 2317 return error; 2318 } 2319 2320 2321 /* 2322 * Audio driver 2323 */ 2324 2325 /* 2326 * Must be called with sc_exlock held and without sc_lock held. 2327 */ 2328 int 2329 audio_open(dev_t dev, struct audio_softc *sc, int flags, int ifmt, 2330 struct lwp *l, audio_file_t **bellfile) 2331 { 2332 struct audio_info ai; 2333 struct file *fp; 2334 audio_file_t *af; 2335 audio_ring_t *hwbuf; 2336 bool fullduplex; 2337 bool cred_held; 2338 bool hw_opened; 2339 bool rmixer_started; 2340 bool inserted; 2341 int fd; 2342 int error; 2343 2344 KASSERT(sc->sc_exlock); 2345 2346 TRACE(1, "%sdev=%s flags=0x%x po=%d ro=%d", 2347 (audiodebug >= 3) ? "start " : "", 2348 ISDEVSOUND(dev) ? "sound" : "audio", 2349 flags, sc->sc_popens, sc->sc_ropens); 2350 2351 fp = NULL; 2352 cred_held = false; 2353 hw_opened = false; 2354 rmixer_started = false; 2355 inserted = false; 2356 2357 af = kmem_zalloc(sizeof(*af), KM_SLEEP); 2358 af->sc = sc; 2359 af->dev = dev; 2360 if ((flags & FWRITE) != 0 && audio_can_playback(sc)) 2361 af->mode |= AUMODE_PLAY | AUMODE_PLAY_ALL; 2362 if ((flags & FREAD) != 0 && audio_can_capture(sc)) 2363 af->mode |= AUMODE_RECORD; 2364 if (af->mode == 0) { 2365 error = ENXIO; 2366 goto bad; 2367 } 2368 2369 fullduplex = (sc->sc_props & AUDIO_PROP_FULLDUPLEX); 2370 2371 /* 2372 * On half duplex hardware, 2373 * 1. if mode is (PLAY | REC), let mode PLAY. 2374 * 2. if mode is PLAY, let mode PLAY if no rec tracks, otherwise error. 2375 * 3. if mode is REC, let mode REC if no play tracks, otherwise error. 2376 */ 2377 if (fullduplex == false) { 2378 if ((af->mode & AUMODE_PLAY)) { 2379 if (sc->sc_ropens != 0) { 2380 TRACE(1, "record track already exists"); 2381 error = ENODEV; 2382 goto bad; 2383 } 2384 /* Play takes precedence */ 2385 af->mode &= ~AUMODE_RECORD; 2386 } 2387 if ((af->mode & AUMODE_RECORD)) { 2388 if (sc->sc_popens != 0) { 2389 TRACE(1, "play track already exists"); 2390 error = ENODEV; 2391 goto bad; 2392 } 2393 } 2394 } 2395 2396 /* Create tracks */ 2397 if ((af->mode & AUMODE_PLAY)) 2398 af->ptrack = audio_track_create(sc, sc->sc_pmixer); 2399 if ((af->mode & AUMODE_RECORD)) 2400 af->rtrack = audio_track_create(sc, sc->sc_rmixer); 2401 2402 /* Set parameters */ 2403 AUDIO_INITINFO(&ai); 2404 if (bellfile) { 2405 /* If audiobell, only sample_rate will be set later. */ 2406 ai.play.sample_rate = audio_default.sample_rate; 2407 ai.play.encoding = AUDIO_ENCODING_SLINEAR_NE; 2408 ai.play.channels = 1; 2409 ai.play.precision = 16; 2410 ai.play.pause = 0; 2411 } else if (ISDEVAUDIO(dev)) { 2412 /* If /dev/audio, initialize everytime. */ 2413 ai.play.sample_rate = audio_default.sample_rate; 2414 ai.play.encoding = audio_default.encoding; 2415 ai.play.channels = audio_default.channels; 2416 ai.play.precision = audio_default.precision; 2417 ai.play.pause = 0; 2418 ai.record.sample_rate = audio_default.sample_rate; 2419 ai.record.encoding = audio_default.encoding; 2420 ai.record.channels = audio_default.channels; 2421 ai.record.precision = audio_default.precision; 2422 ai.record.pause = 0; 2423 } else { 2424 /* If /dev/sound, take over the previous parameters. */ 2425 ai.play.sample_rate = sc->sc_sound_pparams.sample_rate; 2426 ai.play.encoding = sc->sc_sound_pparams.encoding; 2427 ai.play.channels = sc->sc_sound_pparams.channels; 2428 ai.play.precision = sc->sc_sound_pparams.precision; 2429 ai.play.pause = sc->sc_sound_ppause; 2430 ai.record.sample_rate = sc->sc_sound_rparams.sample_rate; 2431 ai.record.encoding = sc->sc_sound_rparams.encoding; 2432 ai.record.channels = sc->sc_sound_rparams.channels; 2433 ai.record.precision = sc->sc_sound_rparams.precision; 2434 ai.record.pause = sc->sc_sound_rpause; 2435 } 2436 error = audio_file_setinfo(sc, af, &ai); 2437 if (error) 2438 goto bad; 2439 2440 if (sc->sc_popens + sc->sc_ropens == 0) { 2441 /* First open */ 2442 2443 sc->sc_cred = kauth_cred_get(); 2444 kauth_cred_hold(sc->sc_cred); 2445 cred_held = true; 2446 2447 if (sc->hw_if->open) { 2448 int hwflags; 2449 2450 /* 2451 * Call hw_if->open() only at first open of 2452 * combination of playback and recording. 2453 * On full duplex hardware, the flags passed to 2454 * hw_if->open() is always (FREAD | FWRITE) 2455 * regardless of this open()'s flags. 2456 * see also dev/isa/aria.c 2457 * On half duplex hardware, the flags passed to 2458 * hw_if->open() is either FREAD or FWRITE. 2459 * see also arch/evbarm/mini2440/audio_mini2440.c 2460 */ 2461 if (fullduplex) { 2462 hwflags = FREAD | FWRITE; 2463 } else { 2464 /* Construct hwflags from af->mode. */ 2465 hwflags = 0; 2466 if ((af->mode & AUMODE_PLAY) != 0) 2467 hwflags |= FWRITE; 2468 if ((af->mode & AUMODE_RECORD) != 0) 2469 hwflags |= FREAD; 2470 } 2471 2472 mutex_enter(sc->sc_lock); 2473 mutex_enter(sc->sc_intr_lock); 2474 error = sc->hw_if->open(sc->hw_hdl, hwflags); 2475 mutex_exit(sc->sc_intr_lock); 2476 mutex_exit(sc->sc_lock); 2477 if (error) 2478 goto bad; 2479 } 2480 /* 2481 * Regardless of whether we called hw_if->open (whether 2482 * hw_if->open exists) or not, we move to the Opened phase 2483 * here. Therefore from this point, we have to call 2484 * hw_if->close (if exists) whenever abort. 2485 * Note that both of hw_if->{open,close} are optional. 2486 */ 2487 hw_opened = true; 2488 2489 /* 2490 * Set speaker mode when a half duplex. 2491 * XXX I'm not sure this is correct. 2492 */ 2493 if (1/*XXX*/) { 2494 if (sc->hw_if->speaker_ctl) { 2495 int on; 2496 if (af->ptrack) { 2497 on = 1; 2498 } else { 2499 on = 0; 2500 } 2501 mutex_enter(sc->sc_lock); 2502 mutex_enter(sc->sc_intr_lock); 2503 error = sc->hw_if->speaker_ctl(sc->hw_hdl, on); 2504 mutex_exit(sc->sc_intr_lock); 2505 mutex_exit(sc->sc_lock); 2506 if (error) 2507 goto bad; 2508 } 2509 } 2510 } else if (sc->sc_multiuser == false) { 2511 uid_t euid = kauth_cred_geteuid(kauth_cred_get()); 2512 if (euid != 0 && euid != kauth_cred_geteuid(sc->sc_cred)) { 2513 error = EPERM; 2514 goto bad; 2515 } 2516 } 2517 2518 /* Call init_output if this is the first playback open. */ 2519 if (af->ptrack && sc->sc_popens == 0) { 2520 if (sc->hw_if->init_output) { 2521 hwbuf = &sc->sc_pmixer->hwbuf; 2522 mutex_enter(sc->sc_lock); 2523 mutex_enter(sc->sc_intr_lock); 2524 error = sc->hw_if->init_output(sc->hw_hdl, 2525 hwbuf->mem, 2526 hwbuf->capacity * 2527 hwbuf->fmt.channels * hwbuf->fmt.stride / NBBY); 2528 mutex_exit(sc->sc_intr_lock); 2529 mutex_exit(sc->sc_lock); 2530 if (error) 2531 goto bad; 2532 } 2533 } 2534 /* 2535 * Call init_input and start rmixer, if this is the first recording 2536 * open. See pause consideration notes. 2537 */ 2538 if (af->rtrack && sc->sc_ropens == 0) { 2539 if (sc->hw_if->init_input) { 2540 hwbuf = &sc->sc_rmixer->hwbuf; 2541 mutex_enter(sc->sc_lock); 2542 mutex_enter(sc->sc_intr_lock); 2543 error = sc->hw_if->init_input(sc->hw_hdl, 2544 hwbuf->mem, 2545 hwbuf->capacity * 2546 hwbuf->fmt.channels * hwbuf->fmt.stride / NBBY); 2547 mutex_exit(sc->sc_intr_lock); 2548 mutex_exit(sc->sc_lock); 2549 if (error) 2550 goto bad; 2551 } 2552 2553 mutex_enter(sc->sc_lock); 2554 audio_rmixer_start(sc); 2555 mutex_exit(sc->sc_lock); 2556 rmixer_started = true; 2557 } 2558 2559 /* 2560 * This is the last sc_lock section in the function, so we have to 2561 * examine sc_dying again before starting the rest tasks. Because 2562 * audiodeatch() may have been invoked (and it would set sc_dying) 2563 * from the time audioopen() was executed until now. If it happens, 2564 * audiodetach() may already have set file->dying for all sc_files 2565 * that exist at that point, so that audioopen() must abort without 2566 * inserting af to sc_files, in order to keep consistency. 2567 */ 2568 mutex_enter(sc->sc_lock); 2569 if (sc->sc_dying) { 2570 mutex_exit(sc->sc_lock); 2571 error = ENXIO; 2572 goto bad; 2573 } 2574 2575 /* Count up finally */ 2576 if (af->ptrack) 2577 sc->sc_popens++; 2578 if (af->rtrack) 2579 sc->sc_ropens++; 2580 mutex_enter(sc->sc_intr_lock); 2581 SLIST_INSERT_HEAD(&sc->sc_files, af, entry); 2582 mutex_exit(sc->sc_intr_lock); 2583 mutex_exit(sc->sc_lock); 2584 inserted = true; 2585 2586 if (bellfile) { 2587 *bellfile = af; 2588 } else { 2589 error = fd_allocfile(&fp, &fd); 2590 if (error) 2591 goto bad; 2592 2593 error = fd_clone(fp, fd, flags, &audio_fileops, af); 2594 KASSERTMSG(error == EMOVEFD, "error=%d", error); 2595 } 2596 2597 /* Be nothing else after fd_clone */ 2598 2599 TRACEF(3, af, "done"); 2600 return error; 2601 2602 bad: 2603 if (inserted) { 2604 mutex_enter(sc->sc_lock); 2605 mutex_enter(sc->sc_intr_lock); 2606 SLIST_REMOVE(&sc->sc_files, af, audio_file, entry); 2607 mutex_exit(sc->sc_intr_lock); 2608 if (af->ptrack) 2609 sc->sc_popens--; 2610 if (af->rtrack) 2611 sc->sc_ropens--; 2612 mutex_exit(sc->sc_lock); 2613 } 2614 2615 if (rmixer_started) { 2616 mutex_enter(sc->sc_lock); 2617 audio_rmixer_halt(sc); 2618 mutex_exit(sc->sc_lock); 2619 } 2620 2621 if (hw_opened) { 2622 if (sc->hw_if->close) { 2623 mutex_enter(sc->sc_lock); 2624 mutex_enter(sc->sc_intr_lock); 2625 sc->hw_if->close(sc->hw_hdl); 2626 mutex_exit(sc->sc_intr_lock); 2627 mutex_exit(sc->sc_lock); 2628 } 2629 } 2630 if (cred_held) { 2631 kauth_cred_free(sc->sc_cred); 2632 } 2633 2634 /* 2635 * Since track here is not yet linked to sc_files, 2636 * you can call track_destroy() without sc_intr_lock. 2637 */ 2638 if (af->rtrack) { 2639 audio_track_destroy(af->rtrack); 2640 af->rtrack = NULL; 2641 } 2642 if (af->ptrack) { 2643 audio_track_destroy(af->ptrack); 2644 af->ptrack = NULL; 2645 } 2646 2647 kmem_free(af, sizeof(*af)); 2648 return error; 2649 } 2650 2651 /* 2652 * Must be called without sc_lock nor sc_exlock held. 2653 */ 2654 int 2655 audio_close(struct audio_softc *sc, audio_file_t *file) 2656 { 2657 int error; 2658 2659 /* 2660 * Drain first. 2661 * It must be done before unlinking(acquiring exlock). 2662 */ 2663 if (file->ptrack) { 2664 mutex_enter(sc->sc_lock); 2665 audio_track_drain(sc, file->ptrack); 2666 mutex_exit(sc->sc_lock); 2667 } 2668 2669 mutex_enter(sc->sc_lock); 2670 mutex_enter(sc->sc_intr_lock); 2671 SLIST_REMOVE(&sc->sc_files, file, audio_file, entry); 2672 mutex_exit(sc->sc_intr_lock); 2673 mutex_exit(sc->sc_lock); 2674 2675 error = audio_exlock_enter(sc); 2676 if (error) { 2677 /* 2678 * If EIO, this sc is about to detach. In this case, even if 2679 * we don't do subsequent _unlink(), audiodetach() will do it. 2680 */ 2681 if (error == EIO) 2682 return error; 2683 2684 /* XXX This should not happen but what should I do ? */ 2685 panic("%s: can't acquire exlock: errno=%d", __func__, error); 2686 } 2687 audio_unlink(sc, file); 2688 audio_exlock_exit(sc); 2689 2690 return 0; 2691 } 2692 2693 /* 2694 * Unlink this file, but not freeing memory here. 2695 * Must be called with sc_exlock held and without sc_lock held. 2696 */ 2697 static void 2698 audio_unlink(struct audio_softc *sc, audio_file_t *file) 2699 { 2700 kauth_cred_t cred = NULL; 2701 int error; 2702 2703 mutex_enter(sc->sc_lock); 2704 2705 TRACEF(1, file, "%spid=%d.%d po=%d ro=%d", 2706 (audiodebug >= 3) ? "start " : "", 2707 (int)curproc->p_pid, (int)curlwp->l_lid, 2708 sc->sc_popens, sc->sc_ropens); 2709 KASSERTMSG(sc->sc_popens + sc->sc_ropens > 0, 2710 "sc->sc_popens=%d, sc->sc_ropens=%d", 2711 sc->sc_popens, sc->sc_ropens); 2712 2713 device_active(sc->sc_dev, DVA_SYSTEM); 2714 2715 if (file->ptrack) { 2716 TRACET(3, file->ptrack, "dropframes=%" PRIu64, 2717 file->ptrack->dropframes); 2718 2719 KASSERT(sc->sc_popens > 0); 2720 sc->sc_popens--; 2721 2722 /* Call hw halt_output if this is the last playback track. */ 2723 if (sc->sc_popens == 0 && sc->sc_pbusy) { 2724 error = audio_pmixer_halt(sc); 2725 if (error) { 2726 audio_printf(sc, 2727 "halt_output failed: errno=%d (ignored)\n", 2728 error); 2729 } 2730 } 2731 2732 /* Restore mixing volume if all tracks are gone. */ 2733 if (sc->sc_popens == 0) { 2734 /* intr_lock is not necessary, but just manners. */ 2735 mutex_enter(sc->sc_intr_lock); 2736 sc->sc_pmixer->volume = 256; 2737 sc->sc_pmixer->voltimer = 0; 2738 mutex_exit(sc->sc_intr_lock); 2739 } 2740 } 2741 if (file->rtrack) { 2742 TRACET(3, file->rtrack, "dropframes=%" PRIu64, 2743 file->rtrack->dropframes); 2744 2745 KASSERT(sc->sc_ropens > 0); 2746 sc->sc_ropens--; 2747 2748 /* Call hw halt_input if this is the last recording track. */ 2749 if (sc->sc_ropens == 0 && sc->sc_rbusy) { 2750 error = audio_rmixer_halt(sc); 2751 if (error) { 2752 audio_printf(sc, 2753 "halt_input failed: errno=%d (ignored)\n", 2754 error); 2755 } 2756 } 2757 2758 } 2759 2760 /* Call hw close if this is the last track. */ 2761 if (sc->sc_popens + sc->sc_ropens == 0) { 2762 if (sc->hw_if->close) { 2763 TRACE(2, "hw_if close"); 2764 mutex_enter(sc->sc_intr_lock); 2765 sc->hw_if->close(sc->hw_hdl); 2766 mutex_exit(sc->sc_intr_lock); 2767 } 2768 cred = sc->sc_cred; 2769 sc->sc_cred = NULL; 2770 } 2771 2772 mutex_exit(sc->sc_lock); 2773 if (cred) 2774 kauth_cred_free(cred); 2775 2776 TRACE(3, "done"); 2777 } 2778 2779 /* 2780 * Must be called without sc_lock nor sc_exlock held. 2781 */ 2782 int 2783 audio_read(struct audio_softc *sc, struct uio *uio, int ioflag, 2784 audio_file_t *file) 2785 { 2786 audio_track_t *track; 2787 audio_ring_t *usrbuf; 2788 audio_ring_t *input; 2789 int error; 2790 2791 /* 2792 * On half-duplex hardware, O_RDWR is treated as O_WRONLY. 2793 * However read() system call itself can be called because it's 2794 * opened with O_RDWR. So in this case, deny this read(). 2795 */ 2796 track = file->rtrack; 2797 if (track == NULL) { 2798 return EBADF; 2799 } 2800 2801 /* I think it's better than EINVAL. */ 2802 if (track->mmapped) 2803 return EPERM; 2804 2805 TRACET(2, track, "resid=%zd ioflag=0x%x", uio->uio_resid, ioflag); 2806 2807 #ifdef AUDIO_PM_IDLE 2808 error = audio_exlock_mutex_enter(sc); 2809 if (error) 2810 return error; 2811 2812 if (device_is_active(&sc->sc_dev) || sc->sc_idle) 2813 device_active(&sc->sc_dev, DVA_SYSTEM); 2814 2815 /* In recording, unlike playback, read() never operates rmixer. */ 2816 2817 audio_exlock_mutex_exit(sc); 2818 #endif 2819 2820 usrbuf = &track->usrbuf; 2821 input = track->input; 2822 error = 0; 2823 2824 while (uio->uio_resid > 0 && error == 0) { 2825 int bytes; 2826 2827 TRACET(3, track, 2828 "while resid=%zd input=%d/%d/%d usrbuf=%d/%d/C%d", 2829 uio->uio_resid, 2830 input->head, input->used, input->capacity, 2831 usrbuf->head, usrbuf->used, usrbuf->capacity); 2832 2833 /* Wait when buffers are empty. */ 2834 mutex_enter(sc->sc_lock); 2835 for (;;) { 2836 bool empty; 2837 audio_track_lock_enter(track); 2838 empty = (input->used == 0 && usrbuf->used == 0); 2839 audio_track_lock_exit(track); 2840 if (!empty) 2841 break; 2842 2843 if ((ioflag & IO_NDELAY)) { 2844 mutex_exit(sc->sc_lock); 2845 return EWOULDBLOCK; 2846 } 2847 2848 TRACET(3, track, "sleep"); 2849 error = audio_track_waitio(sc, track, "audio_read"); 2850 if (error) { 2851 mutex_exit(sc->sc_lock); 2852 return error; 2853 } 2854 } 2855 mutex_exit(sc->sc_lock); 2856 2857 audio_track_lock_enter(track); 2858 /* Convert one block if possible. */ 2859 if (usrbuf->used == 0 && input->used > 0) { 2860 audio_track_record(track); 2861 } 2862 2863 /* uiomove from usrbuf as many bytes as possible. */ 2864 bytes = uimin(usrbuf->used, uio->uio_resid); 2865 error = uiomove((uint8_t *)usrbuf->mem + usrbuf->head, bytes, 2866 uio); 2867 if (error) { 2868 audio_track_lock_exit(track); 2869 device_printf(sc->sc_dev, 2870 "%s: uiomove(%d) failed: errno=%d\n", 2871 __func__, bytes, error); 2872 goto abort; 2873 } 2874 auring_take(usrbuf, bytes); 2875 TRACET(3, track, "uiomove(len=%d) usrbuf=%d/%d/C%d", 2876 bytes, 2877 usrbuf->head, usrbuf->used, usrbuf->capacity); 2878 2879 audio_track_lock_exit(track); 2880 } 2881 2882 abort: 2883 return error; 2884 } 2885 2886 2887 /* 2888 * Clear file's playback and/or record track buffer immediately. 2889 */ 2890 static void 2891 audio_file_clear(struct audio_softc *sc, audio_file_t *file) 2892 { 2893 2894 if (file->ptrack) 2895 audio_track_clear(sc, file->ptrack); 2896 if (file->rtrack) 2897 audio_track_clear(sc, file->rtrack); 2898 } 2899 2900 /* 2901 * Must be called without sc_lock nor sc_exlock held. 2902 */ 2903 int 2904 audio_write(struct audio_softc *sc, struct uio *uio, int ioflag, 2905 audio_file_t *file) 2906 { 2907 audio_track_t *track; 2908 audio_ring_t *usrbuf; 2909 audio_ring_t *outbuf; 2910 int error; 2911 2912 track = file->ptrack; 2913 if (track == NULL) 2914 return EPERM; 2915 2916 /* I think it's better than EINVAL. */ 2917 if (track->mmapped) 2918 return EPERM; 2919 2920 TRACET(2, track, "%sresid=%zd pid=%d.%d ioflag=0x%x", 2921 audiodebug >= 3 ? "begin " : "", 2922 uio->uio_resid, (int)curproc->p_pid, (int)curlwp->l_lid, ioflag); 2923 2924 if (uio->uio_resid == 0) { 2925 track->eofcounter++; 2926 return 0; 2927 } 2928 2929 error = audio_exlock_mutex_enter(sc); 2930 if (error) 2931 return error; 2932 2933 #ifdef AUDIO_PM_IDLE 2934 if (device_is_active(&sc->sc_dev) || sc->sc_idle) 2935 device_active(&sc->sc_dev, DVA_SYSTEM); 2936 #endif 2937 2938 /* 2939 * The first write starts pmixer. 2940 */ 2941 if (sc->sc_pbusy == false) 2942 audio_pmixer_start(sc, false); 2943 audio_exlock_mutex_exit(sc); 2944 2945 usrbuf = &track->usrbuf; 2946 outbuf = &track->outbuf; 2947 track->pstate = AUDIO_STATE_RUNNING; 2948 error = 0; 2949 2950 while (uio->uio_resid > 0 && error == 0) { 2951 int bytes; 2952 2953 TRACET(3, track, "while resid=%zd usrbuf=%d/%d/H%d", 2954 uio->uio_resid, 2955 usrbuf->head, usrbuf->used, track->usrbuf_usedhigh); 2956 2957 /* Wait when buffers are full. */ 2958 mutex_enter(sc->sc_lock); 2959 for (;;) { 2960 bool full; 2961 audio_track_lock_enter(track); 2962 full = (usrbuf->used >= track->usrbuf_usedhigh && 2963 outbuf->used >= outbuf->capacity); 2964 audio_track_lock_exit(track); 2965 if (!full) 2966 break; 2967 2968 if ((ioflag & IO_NDELAY)) { 2969 error = EWOULDBLOCK; 2970 mutex_exit(sc->sc_lock); 2971 goto abort; 2972 } 2973 2974 TRACET(3, track, "sleep usrbuf=%d/H%d", 2975 usrbuf->used, track->usrbuf_usedhigh); 2976 error = audio_track_waitio(sc, track, "audio_write"); 2977 if (error) { 2978 mutex_exit(sc->sc_lock); 2979 goto abort; 2980 } 2981 } 2982 mutex_exit(sc->sc_lock); 2983 2984 audio_track_lock_enter(track); 2985 2986 /* uiomove to usrbuf as many bytes as possible. */ 2987 bytes = uimin(track->usrbuf_usedhigh - usrbuf->used, 2988 uio->uio_resid); 2989 while (bytes > 0) { 2990 int tail = auring_tail(usrbuf); 2991 int len = uimin(bytes, usrbuf->capacity - tail); 2992 error = uiomove((uint8_t *)usrbuf->mem + tail, len, 2993 uio); 2994 if (error) { 2995 audio_track_lock_exit(track); 2996 device_printf(sc->sc_dev, 2997 "%s: uiomove(%d) failed: errno=%d\n", 2998 __func__, len, error); 2999 goto abort; 3000 } 3001 auring_push(usrbuf, len); 3002 TRACET(3, track, "uiomove(len=%d) usrbuf=%d/%d/C%d", 3003 len, 3004 usrbuf->head, usrbuf->used, usrbuf->capacity); 3005 bytes -= len; 3006 } 3007 3008 /* Convert them as many blocks as possible. */ 3009 while (usrbuf->used >= track->usrbuf_blksize && 3010 outbuf->used < outbuf->capacity) { 3011 audio_track_play(track); 3012 } 3013 3014 audio_track_lock_exit(track); 3015 } 3016 3017 abort: 3018 TRACET(3, track, "done error=%d", error); 3019 return error; 3020 } 3021 3022 /* 3023 * Must be called without sc_lock nor sc_exlock held. 3024 */ 3025 int 3026 audio_ioctl(dev_t dev, struct audio_softc *sc, u_long cmd, void *addr, int flag, 3027 struct lwp *l, audio_file_t *file) 3028 { 3029 struct audio_offset *ao; 3030 struct audio_info ai; 3031 audio_track_t *track; 3032 audio_encoding_t *ae; 3033 audio_format_query_t *query; 3034 u_int stamp; 3035 u_int offset; 3036 int val; 3037 int index; 3038 int error; 3039 3040 #if defined(AUDIO_DEBUG) 3041 const char *ioctlnames[] = { 3042 "AUDIO_GETINFO", /* 21 */ 3043 "AUDIO_SETINFO", /* 22 */ 3044 "AUDIO_DRAIN", /* 23 */ 3045 "AUDIO_FLUSH", /* 24 */ 3046 "AUDIO_WSEEK", /* 25 */ 3047 "AUDIO_RERROR", /* 26 */ 3048 "AUDIO_GETDEV", /* 27 */ 3049 "AUDIO_GETENC", /* 28 */ 3050 "AUDIO_GETFD", /* 29 */ 3051 "AUDIO_SETFD", /* 30 */ 3052 "AUDIO_PERROR", /* 31 */ 3053 "AUDIO_GETIOFFS", /* 32 */ 3054 "AUDIO_GETOOFFS", /* 33 */ 3055 "AUDIO_GETPROPS", /* 34 */ 3056 "AUDIO_GETBUFINFO", /* 35 */ 3057 "AUDIO_SETCHAN", /* 36 */ 3058 "AUDIO_GETCHAN", /* 37 */ 3059 "AUDIO_QUERYFORMAT", /* 38 */ 3060 "AUDIO_GETFORMAT", /* 39 */ 3061 "AUDIO_SETFORMAT", /* 40 */ 3062 }; 3063 char pre[64]; 3064 int nameidx = (cmd & 0xff); 3065 if (21 <= nameidx && nameidx <= 21 + __arraycount(ioctlnames)) { 3066 snprintf(pre, sizeof(pre), "pid=%d.%d %s", 3067 (int)curproc->p_pid, (int)l->l_lid, 3068 ioctlnames[nameidx - 21]); 3069 } else { 3070 snprintf(pre, sizeof(pre), "pid=%d.%d (%lu,'%c',%u)", 3071 (int)curproc->p_pid, (int)l->l_lid, 3072 IOCPARM_LEN(cmd), (char)IOCGROUP(cmd), nameidx); 3073 } 3074 #endif 3075 3076 error = 0; 3077 switch (cmd) { 3078 case FIONBIO: 3079 /* All handled in the upper FS layer. */ 3080 break; 3081 3082 case FIONREAD: 3083 /* Get the number of bytes that can be read. */ 3084 track = file->rtrack; 3085 if (track) { 3086 val = audio_track_readablebytes(track); 3087 *(int *)addr = val; 3088 TRACET(2, track, "pid=%d.%d FIONREAD bytes=%d", 3089 (int)curproc->p_pid, (int)l->l_lid, val); 3090 } else { 3091 TRACEF(2, file, "pid=%d.%d FIONREAD no track", 3092 (int)curproc->p_pid, (int)l->l_lid); 3093 } 3094 break; 3095 3096 case FIOASYNC: 3097 /* Set/Clear ASYNC I/O. */ 3098 if (*(int *)addr) { 3099 file->async_audio = curproc->p_pid; 3100 } else { 3101 file->async_audio = 0; 3102 } 3103 TRACEF(2, file, "pid=%d.%d FIOASYNC %s", 3104 (int)curproc->p_pid, (int)l->l_lid, 3105 file->async_audio ? "on" : "off"); 3106 break; 3107 3108 case AUDIO_FLUSH: 3109 /* XXX TODO: clear errors and restart? */ 3110 TRACEF(2, file, "%s", pre); 3111 audio_file_clear(sc, file); 3112 break; 3113 3114 case AUDIO_PERROR: 3115 case AUDIO_RERROR: 3116 /* 3117 * Number of dropped bytes during playback/record. We don't 3118 * know where or when they were dropped (including conversion 3119 * stage). Therefore, the number of accurate bytes or samples 3120 * is also unknown. 3121 */ 3122 track = (cmd == AUDIO_PERROR) ? file->ptrack : file->rtrack; 3123 if (track) { 3124 val = frametobyte(&track->usrbuf.fmt, 3125 track->dropframes); 3126 *(int *)addr = val; 3127 TRACET(2, track, "%s bytes=%d", pre, val); 3128 } else { 3129 TRACEF(2, file, "%s no track", pre); 3130 } 3131 break; 3132 3133 case AUDIO_GETIOFFS: 3134 ao = (struct audio_offset *)addr; 3135 track = file->rtrack; 3136 if (track == NULL) { 3137 ao->samples = 0; 3138 ao->deltablks = 0; 3139 ao->offset = 0; 3140 TRACEF(2, file, "%s no rtrack", pre); 3141 break; 3142 } 3143 mutex_enter(sc->sc_lock); 3144 mutex_enter(sc->sc_intr_lock); 3145 /* figure out where next transfer will start */ 3146 stamp = track->stamp; 3147 offset = auring_tail(track->input); 3148 mutex_exit(sc->sc_intr_lock); 3149 mutex_exit(sc->sc_lock); 3150 3151 /* samples will overflow soon but is as per spec. */ 3152 ao->samples = stamp * track->usrbuf_blksize; 3153 ao->deltablks = stamp - track->last_stamp; 3154 ao->offset = audio_track_inputblk_as_usrbyte(track, offset); 3155 TRACET(2, track, "%s samples=%u deltablks=%u offset=%u", 3156 pre, ao->samples, ao->deltablks, ao->offset); 3157 3158 track->last_stamp = stamp; 3159 break; 3160 3161 case AUDIO_GETOOFFS: 3162 ao = (struct audio_offset *)addr; 3163 track = file->ptrack; 3164 if (track == NULL) { 3165 ao->samples = 0; 3166 ao->deltablks = 0; 3167 ao->offset = 0; 3168 TRACEF(2, file, "%s no ptrack", pre); 3169 break; 3170 } 3171 mutex_enter(sc->sc_lock); 3172 mutex_enter(sc->sc_intr_lock); 3173 /* figure out where next transfer will start */ 3174 stamp = track->stamp; 3175 offset = track->usrbuf.head; 3176 mutex_exit(sc->sc_intr_lock); 3177 mutex_exit(sc->sc_lock); 3178 3179 /* samples will overflow soon but is as per spec. */ 3180 ao->samples = stamp * track->usrbuf_blksize; 3181 ao->deltablks = stamp - track->last_stamp; 3182 ao->offset = offset; 3183 TRACET(2, track, "%s samples=%u deltablks=%u offset=%u", 3184 pre, ao->samples, ao->deltablks, ao->offset); 3185 3186 track->last_stamp = stamp; 3187 break; 3188 3189 case AUDIO_WSEEK: 3190 track = file->ptrack; 3191 if (track) { 3192 val = track->usrbuf.used; 3193 *(u_long *)addr = val; 3194 TRACET(2, track, "%s bytes=%d", pre, val); 3195 } else { 3196 TRACEF(2, file, "%s no ptrack", pre); 3197 } 3198 break; 3199 3200 case AUDIO_SETINFO: 3201 TRACEF(2, file, "%s", pre); 3202 error = audio_exlock_enter(sc); 3203 if (error) 3204 break; 3205 error = audio_file_setinfo(sc, file, (struct audio_info *)addr); 3206 if (error) { 3207 audio_exlock_exit(sc); 3208 break; 3209 } 3210 if (ISDEVSOUND(dev)) 3211 error = audiogetinfo(sc, &sc->sc_ai, 0, file); 3212 audio_exlock_exit(sc); 3213 break; 3214 3215 case AUDIO_GETINFO: 3216 TRACEF(2, file, "%s", pre); 3217 error = audio_exlock_enter(sc); 3218 if (error) 3219 break; 3220 error = audiogetinfo(sc, (struct audio_info *)addr, 1, file); 3221 audio_exlock_exit(sc); 3222 break; 3223 3224 case AUDIO_GETBUFINFO: 3225 TRACEF(2, file, "%s", pre); 3226 error = audio_exlock_enter(sc); 3227 if (error) 3228 break; 3229 error = audiogetinfo(sc, (struct audio_info *)addr, 0, file); 3230 audio_exlock_exit(sc); 3231 break; 3232 3233 case AUDIO_DRAIN: 3234 track = file->ptrack; 3235 if (track) { 3236 TRACET(2, track, "%s", pre); 3237 mutex_enter(sc->sc_lock); 3238 error = audio_track_drain(sc, track); 3239 mutex_exit(sc->sc_lock); 3240 } else { 3241 TRACEF(2, file, "%s no ptrack", pre); 3242 } 3243 break; 3244 3245 case AUDIO_GETDEV: 3246 TRACEF(2, file, "%s", pre); 3247 error = sc->hw_if->getdev(sc->hw_hdl, (audio_device_t *)addr); 3248 break; 3249 3250 case AUDIO_GETENC: 3251 ae = (audio_encoding_t *)addr; 3252 index = ae->index; 3253 TRACEF(2, file, "%s index=%d", pre, index); 3254 if (index < 0 || index >= __arraycount(audio_encodings)) { 3255 error = EINVAL; 3256 break; 3257 } 3258 *ae = audio_encodings[index]; 3259 ae->index = index; 3260 /* 3261 * EMULATED always. 3262 * EMULATED flag at that time used to mean that it could 3263 * not be passed directly to the hardware as-is. But 3264 * currently, all formats including hardware native is not 3265 * passed directly to the hardware. So I set EMULATED 3266 * flag for all formats. 3267 */ 3268 ae->flags = AUDIO_ENCODINGFLAG_EMULATED; 3269 break; 3270 3271 case AUDIO_GETFD: 3272 /* 3273 * Returns the current setting of full duplex mode. 3274 * If HW has full duplex mode and there are two mixers, 3275 * it is full duplex. Otherwise half duplex. 3276 */ 3277 error = audio_exlock_enter(sc); 3278 if (error) 3279 break; 3280 val = (sc->sc_props & AUDIO_PROP_FULLDUPLEX) 3281 && (sc->sc_pmixer && sc->sc_rmixer); 3282 audio_exlock_exit(sc); 3283 *(int *)addr = val; 3284 TRACEF(2, file, "%s fulldup=%d", pre, val); 3285 break; 3286 3287 case AUDIO_GETPROPS: 3288 val = sc->sc_props; 3289 *(int *)addr = val; 3290 #if defined(AUDIO_DEBUG) 3291 char pbuf[64]; 3292 snprintb(pbuf, sizeof(pbuf), "\x10" 3293 "\6CAPTURE" "\5PLAY" "\3INDEP" "\2MMAP" "\1FULLDUP", val); 3294 TRACEF(2, file, "%s %s", pre, pbuf); 3295 #endif 3296 break; 3297 3298 case AUDIO_QUERYFORMAT: 3299 query = (audio_format_query_t *)addr; 3300 TRACEF(2, file, "%s index=%u", pre, query->index); 3301 mutex_enter(sc->sc_lock); 3302 error = sc->hw_if->query_format(sc->hw_hdl, query); 3303 mutex_exit(sc->sc_lock); 3304 /* Hide internal information */ 3305 query->fmt.driver_data = NULL; 3306 break; 3307 3308 case AUDIO_GETFORMAT: 3309 TRACEF(2, file, "%s", pre); 3310 error = audio_exlock_enter(sc); 3311 if (error) 3312 break; 3313 audio_mixers_get_format(sc, (struct audio_info *)addr); 3314 audio_exlock_exit(sc); 3315 break; 3316 3317 case AUDIO_SETFORMAT: 3318 TRACEF(2, file, "%s", pre); 3319 error = audio_exlock_enter(sc); 3320 audio_mixers_get_format(sc, &ai); 3321 error = audio_mixers_set_format(sc, (struct audio_info *)addr); 3322 if (error) { 3323 /* Rollback */ 3324 audio_mixers_set_format(sc, &ai); 3325 } 3326 audio_exlock_exit(sc); 3327 break; 3328 3329 case AUDIO_SETFD: 3330 case AUDIO_SETCHAN: 3331 case AUDIO_GETCHAN: 3332 /* Obsoleted */ 3333 TRACEF(2, file, "%s", pre); 3334 break; 3335 3336 default: 3337 TRACEF(2, file, "%s", pre); 3338 if (sc->hw_if->dev_ioctl) { 3339 mutex_enter(sc->sc_lock); 3340 error = sc->hw_if->dev_ioctl(sc->hw_hdl, 3341 cmd, addr, flag, l); 3342 mutex_exit(sc->sc_lock); 3343 } else { 3344 error = EINVAL; 3345 } 3346 break; 3347 } 3348 3349 if (error) 3350 TRACEF(2, file, "%s error=%d", pre, error); 3351 return error; 3352 } 3353 3354 /* 3355 * Convert n [frames] of the input buffer to bytes in the usrbuf format. 3356 * n is in frames but should be a multiple of frame/block. Note that the 3357 * usrbuf's frame/block and the input buffer's frame/block may be different 3358 * (i.e., if frequencies are different). 3359 * 3360 * This function is for recording track only. 3361 */ 3362 static int 3363 audio_track_inputblk_as_usrbyte(const audio_track_t *track, int n) 3364 { 3365 int input_fpb; 3366 3367 /* 3368 * In the input buffer on recording track, these are the same. 3369 * input_fpb = frame_per_block(track->mixer, &track->input->fmt); 3370 */ 3371 input_fpb = track->mixer->frames_per_block; 3372 3373 return (n / input_fpb) * track->usrbuf_blksize; 3374 } 3375 3376 /* 3377 * Returns the number of bytes that can be read on recording buffer. 3378 */ 3379 static int 3380 audio_track_readablebytes(const audio_track_t *track) 3381 { 3382 int bytes; 3383 3384 KASSERT(track); 3385 KASSERT(track->mode == AUMODE_RECORD); 3386 3387 /* 3388 * For recording, track->input is the main block-unit buffer and 3389 * track->usrbuf holds less than one block of byte data ("fragment"). 3390 * Note that the input buffer is in frames and the usrbuf is in bytes. 3391 * 3392 * Actual total capacity of these two buffers is 3393 * input->capacity [frames] + usrbuf.capacity [bytes], 3394 * but only input->capacity is reported to userland as buffer_size. 3395 * So, even if the total used bytes exceed input->capacity, report it 3396 * as input->capacity for consistency. 3397 */ 3398 bytes = audio_track_inputblk_as_usrbyte(track, track->input->used); 3399 if (track->input->used < track->input->capacity) { 3400 bytes += track->usrbuf.used; 3401 } 3402 return bytes; 3403 } 3404 3405 /* 3406 * Must be called without sc_lock nor sc_exlock held. 3407 */ 3408 int 3409 audio_poll(struct audio_softc *sc, int events, struct lwp *l, 3410 audio_file_t *file) 3411 { 3412 audio_track_t *track; 3413 int revents; 3414 bool in_is_valid; 3415 bool out_is_valid; 3416 3417 #if defined(AUDIO_DEBUG) 3418 #define POLLEV_BITMAP "\177\020" \ 3419 "b\10WRBAND\0" \ 3420 "b\7RDBAND\0" "b\6RDNORM\0" "b\5NVAL\0" "b\4HUP\0" \ 3421 "b\3ERR\0" "b\2OUT\0" "b\1PRI\0" "b\0IN\0" 3422 char evbuf[64]; 3423 snprintb(evbuf, sizeof(evbuf), POLLEV_BITMAP, events); 3424 TRACEF(2, file, "pid=%d.%d events=%s", 3425 (int)curproc->p_pid, (int)l->l_lid, evbuf); 3426 #endif 3427 3428 revents = 0; 3429 in_is_valid = false; 3430 out_is_valid = false; 3431 if (events & (POLLIN | POLLRDNORM)) { 3432 track = file->rtrack; 3433 if (track) { 3434 int used; 3435 in_is_valid = true; 3436 used = audio_track_readablebytes(track); 3437 if (used > 0) 3438 revents |= events & (POLLIN | POLLRDNORM); 3439 } 3440 } 3441 if (events & (POLLOUT | POLLWRNORM)) { 3442 track = file->ptrack; 3443 if (track) { 3444 out_is_valid = true; 3445 if (track->usrbuf.used <= track->usrbuf_usedlow) 3446 revents |= events & (POLLOUT | POLLWRNORM); 3447 } 3448 } 3449 3450 if (revents == 0) { 3451 mutex_enter(sc->sc_lock); 3452 if (in_is_valid) { 3453 TRACEF(3, file, "selrecord rsel"); 3454 selrecord(l, &sc->sc_rsel); 3455 } 3456 if (out_is_valid) { 3457 TRACEF(3, file, "selrecord wsel"); 3458 selrecord(l, &sc->sc_wsel); 3459 } 3460 mutex_exit(sc->sc_lock); 3461 } 3462 3463 #if defined(AUDIO_DEBUG) 3464 snprintb(evbuf, sizeof(evbuf), POLLEV_BITMAP, revents); 3465 TRACEF(2, file, "revents=%s", evbuf); 3466 #endif 3467 return revents; 3468 } 3469 3470 static const struct filterops audioread_filtops = { 3471 .f_flags = FILTEROP_ISFD, 3472 .f_attach = NULL, 3473 .f_detach = filt_audioread_detach, 3474 .f_event = filt_audioread_event, 3475 }; 3476 3477 static void 3478 filt_audioread_detach(struct knote *kn) 3479 { 3480 struct audio_softc *sc; 3481 audio_file_t *file; 3482 3483 file = kn->kn_hook; 3484 sc = file->sc; 3485 TRACEF(3, file, "called"); 3486 3487 mutex_enter(sc->sc_lock); 3488 selremove_knote(&sc->sc_rsel, kn); 3489 mutex_exit(sc->sc_lock); 3490 } 3491 3492 static int 3493 filt_audioread_event(struct knote *kn, long hint) 3494 { 3495 audio_file_t *file; 3496 audio_track_t *track; 3497 3498 file = kn->kn_hook; 3499 track = file->rtrack; 3500 3501 /* 3502 * kn_data must contain the number of bytes can be read. 3503 * The return value indicates whether the event occurs or not. 3504 */ 3505 3506 if (track == NULL) { 3507 /* can not read with this descriptor. */ 3508 kn->kn_data = 0; 3509 return 0; 3510 } 3511 3512 kn->kn_data = audio_track_readablebytes(track); 3513 TRACEF(3, file, "data=%" PRId64, kn->kn_data); 3514 return kn->kn_data > 0; 3515 } 3516 3517 static const struct filterops audiowrite_filtops = { 3518 .f_flags = FILTEROP_ISFD, 3519 .f_attach = NULL, 3520 .f_detach = filt_audiowrite_detach, 3521 .f_event = filt_audiowrite_event, 3522 }; 3523 3524 static void 3525 filt_audiowrite_detach(struct knote *kn) 3526 { 3527 struct audio_softc *sc; 3528 audio_file_t *file; 3529 3530 file = kn->kn_hook; 3531 sc = file->sc; 3532 TRACEF(3, file, "called"); 3533 3534 mutex_enter(sc->sc_lock); 3535 selremove_knote(&sc->sc_wsel, kn); 3536 mutex_exit(sc->sc_lock); 3537 } 3538 3539 static int 3540 filt_audiowrite_event(struct knote *kn, long hint) 3541 { 3542 audio_file_t *file; 3543 audio_track_t *track; 3544 3545 file = kn->kn_hook; 3546 track = file->ptrack; 3547 3548 /* 3549 * kn_data must contain the number of bytes can be write. 3550 * The return value indicates whether the event occurs or not. 3551 */ 3552 3553 if (track == NULL) { 3554 /* can not write with this descriptor. */ 3555 kn->kn_data = 0; 3556 return 0; 3557 } 3558 3559 kn->kn_data = track->usrbuf_usedhigh - track->usrbuf.used; 3560 TRACEF(3, file, "data=%" PRId64, kn->kn_data); 3561 return (track->usrbuf.used < track->usrbuf_usedlow); 3562 } 3563 3564 /* 3565 * Must be called without sc_lock nor sc_exlock held. 3566 */ 3567 int 3568 audio_kqfilter(struct audio_softc *sc, audio_file_t *file, struct knote *kn) 3569 { 3570 struct selinfo *sip; 3571 3572 TRACEF(3, file, "kn=%p kn_filter=%x", kn, (int)kn->kn_filter); 3573 3574 switch (kn->kn_filter) { 3575 case EVFILT_READ: 3576 sip = &sc->sc_rsel; 3577 kn->kn_fop = &audioread_filtops; 3578 break; 3579 3580 case EVFILT_WRITE: 3581 sip = &sc->sc_wsel; 3582 kn->kn_fop = &audiowrite_filtops; 3583 break; 3584 3585 default: 3586 return EINVAL; 3587 } 3588 3589 kn->kn_hook = file; 3590 3591 mutex_enter(sc->sc_lock); 3592 selrecord_knote(sip, kn); 3593 mutex_exit(sc->sc_lock); 3594 3595 return 0; 3596 } 3597 3598 /* 3599 * Must be called without sc_lock nor sc_exlock held. 3600 */ 3601 int 3602 audio_mmap(struct audio_softc *sc, off_t *offp, size_t len, int prot, 3603 int *flagsp, int *advicep, struct uvm_object **uobjp, int *maxprotp, 3604 audio_file_t *file) 3605 { 3606 audio_track_t *track; 3607 struct uvm_object *uobj; 3608 vaddr_t vstart; 3609 vsize_t vsize; 3610 int error; 3611 3612 TRACEF(1, file, "off=%jd, len=%ju, prot=%d", 3613 (intmax_t)(*offp), (uintmax_t)len, prot); 3614 3615 KASSERT(len > 0); 3616 3617 if (*offp < 0) 3618 return EINVAL; 3619 3620 #if 0 3621 /* XXX 3622 * The idea here was to use the protection to determine if 3623 * we are mapping the read or write buffer, but it fails. 3624 * The VM system is broken in (at least) two ways. 3625 * 1) If you map memory VM_PROT_WRITE you SIGSEGV 3626 * when writing to it, so VM_PROT_READ|VM_PROT_WRITE 3627 * has to be used for mmapping the play buffer. 3628 * 2) Even if calling mmap() with VM_PROT_READ|VM_PROT_WRITE 3629 * audio_mmap will get called at some point with VM_PROT_READ 3630 * only. 3631 * So, alas, we always map the play buffer for now. 3632 */ 3633 if (prot == (VM_PROT_READ|VM_PROT_WRITE) || 3634 prot == VM_PROT_WRITE) 3635 track = file->ptrack; 3636 else if (prot == VM_PROT_READ) 3637 track = file->rtrack; 3638 else 3639 return EINVAL; 3640 #else 3641 track = file->ptrack; 3642 #endif 3643 if (track == NULL) 3644 return EACCES; 3645 3646 /* XXX TODO: what happens when mmap twice. */ 3647 if (track->mmapped) 3648 return EIO; 3649 3650 /* Create a uvm anonymous object */ 3651 vsize = roundup2(MAX(track->usrbuf.capacity, PAGE_SIZE), PAGE_SIZE); 3652 if (*offp + len > vsize) 3653 return EOVERFLOW; 3654 uobj = uao_create(vsize, 0); 3655 3656 /* Map it into the kernel virtual address space */ 3657 vstart = 0; 3658 error = uvm_map(kernel_map, &vstart, vsize, uobj, 0, 0, 3659 UVM_MAPFLAG(UVM_PROT_RW, UVM_PROT_RW, UVM_INH_NONE, 3660 UVM_ADV_RANDOM, 0)); 3661 if (error) { 3662 device_printf(sc->sc_dev, "uvm_map failed: errno=%d\n", error); 3663 uao_detach(uobj); /* release reference */ 3664 return error; 3665 } 3666 3667 error = uvm_map_pageable(kernel_map, vstart, vstart + vsize, 3668 false, 0); 3669 if (error) { 3670 device_printf(sc->sc_dev, "uvm_map_pageable failed: errno=%d\n", 3671 error); 3672 goto abort; 3673 } 3674 3675 error = audio_exlock_mutex_enter(sc); 3676 if (error) 3677 goto abort; 3678 3679 /* 3680 * mmap() will start playing immediately. XXX Maybe we lack API... 3681 * If no one has played yet, start pmixer here. 3682 */ 3683 if (sc->sc_pbusy == false) 3684 audio_pmixer_start(sc, true); 3685 audio_exlock_mutex_exit(sc); 3686 3687 /* Finally, replace the usrbuf from kmem to uvm. */ 3688 audio_track_lock_enter(track); 3689 kmem_free(track->usrbuf.mem, track->usrbuf_allocsize); 3690 track->usrbuf.mem = (void *)vstart; 3691 track->usrbuf_allocsize = vsize; 3692 memset(track->usrbuf.mem, 0, vsize); 3693 track->mmapped = true; 3694 audio_track_lock_exit(track); 3695 3696 /* Acquire a reference for the mmap. munmap will release. */ 3697 uao_reference(uobj); 3698 *uobjp = uobj; 3699 *maxprotp = prot; 3700 *advicep = UVM_ADV_RANDOM; 3701 *flagsp = MAP_SHARED; 3702 3703 return 0; 3704 3705 abort: 3706 uvm_unmap(kernel_map, vstart, vstart + vsize); 3707 /* uvm_unmap also detach uobj */ 3708 return error; 3709 } 3710 3711 /* 3712 * /dev/audioctl has to be able to open at any time without interference 3713 * with any /dev/audio or /dev/sound. 3714 * Must be called with sc_exlock held and without sc_lock held. 3715 */ 3716 static int 3717 audioctl_open(dev_t dev, struct audio_softc *sc, int flags, int ifmt, 3718 struct lwp *l) 3719 { 3720 struct file *fp; 3721 audio_file_t *af; 3722 int fd; 3723 int error; 3724 3725 KASSERT(sc->sc_exlock); 3726 3727 TRACE(1, "called"); 3728 3729 error = fd_allocfile(&fp, &fd); 3730 if (error) 3731 return error; 3732 3733 af = kmem_zalloc(sizeof(*af), KM_SLEEP); 3734 af->sc = sc; 3735 af->dev = dev; 3736 3737 mutex_enter(sc->sc_lock); 3738 if (sc->sc_dying) { 3739 mutex_exit(sc->sc_lock); 3740 kmem_free(af, sizeof(*af)); 3741 fd_abort(curproc, fp, fd); 3742 return ENXIO; 3743 } 3744 mutex_enter(sc->sc_intr_lock); 3745 SLIST_INSERT_HEAD(&sc->sc_files, af, entry); 3746 mutex_exit(sc->sc_intr_lock); 3747 mutex_exit(sc->sc_lock); 3748 3749 error = fd_clone(fp, fd, flags, &audio_fileops, af); 3750 KASSERTMSG(error == EMOVEFD, "error=%d", error); 3751 3752 return error; 3753 } 3754 3755 /* 3756 * Free 'mem' if available, and initialize the pointer. 3757 * For this reason, this is implemented as macro. 3758 */ 3759 #define audio_free(mem) do { \ 3760 if (mem != NULL) { \ 3761 kern_free(mem); \ 3762 mem = NULL; \ 3763 } \ 3764 } while (0) 3765 3766 /* 3767 * (Re)allocate 'memblock' with specified 'bytes'. 3768 * bytes must not be 0. 3769 * This function never returns NULL. 3770 */ 3771 static void * 3772 audio_realloc(void *memblock, size_t bytes) 3773 { 3774 3775 KASSERT(bytes != 0); 3776 if (memblock) 3777 kern_free(memblock); 3778 return kern_malloc(bytes, M_WAITOK); 3779 } 3780 3781 /* 3782 * Free usrbuf (if available). 3783 */ 3784 static void 3785 audio_free_usrbuf(audio_track_t *track) 3786 { 3787 vaddr_t vstart; 3788 vsize_t vsize; 3789 3790 if (track->usrbuf_allocsize != 0) { 3791 if (track->mmapped) { 3792 /* 3793 * Unmap the kernel mapping. uvm_unmap releases the 3794 * reference to the uvm object, and this should be the 3795 * last virtual mapping of the uvm object, so no need 3796 * to explicitly release (`detach') the object. 3797 */ 3798 vstart = (vaddr_t)track->usrbuf.mem; 3799 vsize = track->usrbuf_allocsize; 3800 uvm_unmap(kernel_map, vstart, vstart + vsize); 3801 track->mmapped = false; 3802 } else { 3803 kmem_free(track->usrbuf.mem, track->usrbuf_allocsize); 3804 } 3805 } 3806 track->usrbuf.mem = NULL; 3807 track->usrbuf.capacity = 0; 3808 track->usrbuf_allocsize = 0; 3809 } 3810 3811 /* 3812 * This filter changes the volume for each channel. 3813 * arg->context points track->ch_volume[]. 3814 */ 3815 static void 3816 audio_track_chvol(audio_filter_arg_t *arg) 3817 { 3818 int16_t *ch_volume; 3819 const aint_t *s; 3820 aint_t *d; 3821 u_int i; 3822 u_int ch; 3823 u_int channels; 3824 3825 DIAGNOSTIC_filter_arg(arg); 3826 KASSERTMSG(arg->srcfmt->channels == arg->dstfmt->channels, 3827 "arg->srcfmt->channels=%d, arg->dstfmt->channels=%d", 3828 arg->srcfmt->channels, arg->dstfmt->channels); 3829 KASSERT(arg->context != NULL); 3830 KASSERTMSG(arg->srcfmt->channels <= AUDIO_MAX_CHANNELS, 3831 "arg->srcfmt->channels=%d", arg->srcfmt->channels); 3832 3833 s = arg->src; 3834 d = arg->dst; 3835 ch_volume = arg->context; 3836 3837 channels = arg->srcfmt->channels; 3838 for (i = 0; i < arg->count; i++) { 3839 for (ch = 0; ch < channels; ch++) { 3840 aint2_t val; 3841 val = *s++; 3842 val = AUDIO_SCALEDOWN(val * ch_volume[ch], 8); 3843 *d++ = (aint_t)val; 3844 } 3845 } 3846 } 3847 3848 /* 3849 * This filter performs conversion from stereo (or more channels) to mono. 3850 */ 3851 static void 3852 audio_track_chmix_mixLR(audio_filter_arg_t *arg) 3853 { 3854 const aint_t *s; 3855 aint_t *d; 3856 u_int i; 3857 3858 DIAGNOSTIC_filter_arg(arg); 3859 3860 s = arg->src; 3861 d = arg->dst; 3862 3863 for (i = 0; i < arg->count; i++) { 3864 *d++ = AUDIO_SCALEDOWN(s[0], 1) + AUDIO_SCALEDOWN(s[1], 1); 3865 s += arg->srcfmt->channels; 3866 } 3867 } 3868 3869 /* 3870 * This filter performs conversion from mono to stereo (or more channels). 3871 */ 3872 static void 3873 audio_track_chmix_dupLR(audio_filter_arg_t *arg) 3874 { 3875 const aint_t *s; 3876 aint_t *d; 3877 u_int i; 3878 u_int ch; 3879 u_int dstchannels; 3880 3881 DIAGNOSTIC_filter_arg(arg); 3882 3883 s = arg->src; 3884 d = arg->dst; 3885 dstchannels = arg->dstfmt->channels; 3886 3887 for (i = 0; i < arg->count; i++) { 3888 d[0] = s[0]; 3889 d[1] = s[0]; 3890 s++; 3891 d += dstchannels; 3892 } 3893 if (dstchannels > 2) { 3894 d = arg->dst; 3895 for (i = 0; i < arg->count; i++) { 3896 for (ch = 2; ch < dstchannels; ch++) { 3897 d[ch] = 0; 3898 } 3899 d += dstchannels; 3900 } 3901 } 3902 } 3903 3904 /* 3905 * This filter shrinks M channels into N channels. 3906 * Extra channels are discarded. 3907 */ 3908 static void 3909 audio_track_chmix_shrink(audio_filter_arg_t *arg) 3910 { 3911 const aint_t *s; 3912 aint_t *d; 3913 u_int i; 3914 u_int ch; 3915 3916 DIAGNOSTIC_filter_arg(arg); 3917 3918 s = arg->src; 3919 d = arg->dst; 3920 3921 for (i = 0; i < arg->count; i++) { 3922 for (ch = 0; ch < arg->dstfmt->channels; ch++) { 3923 *d++ = s[ch]; 3924 } 3925 s += arg->srcfmt->channels; 3926 } 3927 } 3928 3929 /* 3930 * This filter expands M channels into N channels. 3931 * Silence is inserted for missing channels. 3932 */ 3933 static void 3934 audio_track_chmix_expand(audio_filter_arg_t *arg) 3935 { 3936 const aint_t *s; 3937 aint_t *d; 3938 u_int i; 3939 u_int ch; 3940 u_int srcchannels; 3941 u_int dstchannels; 3942 3943 DIAGNOSTIC_filter_arg(arg); 3944 3945 s = arg->src; 3946 d = arg->dst; 3947 3948 srcchannels = arg->srcfmt->channels; 3949 dstchannels = arg->dstfmt->channels; 3950 for (i = 0; i < arg->count; i++) { 3951 for (ch = 0; ch < srcchannels; ch++) { 3952 *d++ = *s++; 3953 } 3954 for (; ch < dstchannels; ch++) { 3955 *d++ = 0; 3956 } 3957 } 3958 } 3959 3960 /* 3961 * This filter performs frequency conversion (up sampling). 3962 * It uses linear interpolation. 3963 */ 3964 static void 3965 audio_track_freq_up(audio_filter_arg_t *arg) 3966 { 3967 audio_track_t *track; 3968 audio_ring_t *src; 3969 audio_ring_t *dst; 3970 const aint_t *s; 3971 aint_t *d; 3972 aint_t prev[AUDIO_MAX_CHANNELS]; 3973 aint_t curr[AUDIO_MAX_CHANNELS]; 3974 aint_t grad[AUDIO_MAX_CHANNELS]; 3975 u_int i; 3976 u_int t; 3977 u_int step; 3978 u_int channels; 3979 u_int ch; 3980 int srcused; 3981 3982 track = arg->context; 3983 KASSERT(track); 3984 src = &track->freq.srcbuf; 3985 dst = track->freq.dst; 3986 DIAGNOSTIC_ring(dst); 3987 DIAGNOSTIC_ring(src); 3988 KASSERT(src->used > 0); 3989 KASSERTMSG(src->fmt.channels == dst->fmt.channels, 3990 "src->fmt.channels=%d dst->fmt.channels=%d", 3991 src->fmt.channels, dst->fmt.channels); 3992 KASSERTMSG(src->head % track->mixer->frames_per_block == 0, 3993 "src->head=%d track->mixer->frames_per_block=%d", 3994 src->head, track->mixer->frames_per_block); 3995 3996 s = arg->src; 3997 d = arg->dst; 3998 3999 /* 4000 * In order to facilitate interpolation for each block, slide (delay) 4001 * input by one sample. As a result, strictly speaking, the output 4002 * phase is delayed by 1/dstfreq. However, I believe there is no 4003 * observable impact. 4004 * 4005 * Example) 4006 * srcfreq:dstfreq = 1:3 4007 * 4008 * A - - 4009 * | 4010 * | 4011 * | B - - 4012 * +-----+-----> input timeframe 4013 * 0 1 4014 * 4015 * 0 1 4016 * +-----+-----> input timeframe 4017 * | A 4018 * | x x 4019 * | x x 4020 * x (B) 4021 * +-+-+-+-+-+-> output timeframe 4022 * 0 1 2 3 4 5 4023 */ 4024 4025 /* Last samples in previous block */ 4026 channels = src->fmt.channels; 4027 for (ch = 0; ch < channels; ch++) { 4028 prev[ch] = track->freq_prev[ch]; 4029 curr[ch] = track->freq_curr[ch]; 4030 grad[ch] = curr[ch] - prev[ch]; 4031 } 4032 4033 step = track->freq_step; 4034 t = track->freq_current; 4035 //#define FREQ_DEBUG 4036 #if defined(FREQ_DEBUG) 4037 #define PRINTF(fmt...) printf(fmt) 4038 #else 4039 #define PRINTF(fmt...) do { } while (0) 4040 #endif 4041 srcused = src->used; 4042 PRINTF("upstart step=%d leap=%d", step, track->freq_leap); 4043 PRINTF(" srcused=%d arg->count=%u", src->used, arg->count); 4044 PRINTF(" prev=%d curr=%d grad=%d", prev[0], curr[0], grad[0]); 4045 PRINTF(" t=%d\n", t); 4046 4047 for (i = 0; i < arg->count; i++) { 4048 PRINTF("i=%d t=%5d", i, t); 4049 if (t >= 65536) { 4050 for (ch = 0; ch < channels; ch++) { 4051 prev[ch] = curr[ch]; 4052 curr[ch] = *s++; 4053 grad[ch] = curr[ch] - prev[ch]; 4054 } 4055 PRINTF(" prev=%d s[%d]=%d", 4056 prev[0], src->used - srcused, curr[0]); 4057 4058 /* Update */ 4059 t -= 65536; 4060 srcused--; 4061 if (srcused < 0) { 4062 PRINTF(" break\n"); 4063 break; 4064 } 4065 } 4066 4067 for (ch = 0; ch < channels; ch++) { 4068 *d++ = prev[ch] + (aint2_t)grad[ch] * t / 65536; 4069 #if defined(FREQ_DEBUG) 4070 if (ch == 0) 4071 printf(" t=%5d *d=%d", t, d[-1]); 4072 #endif 4073 } 4074 t += step; 4075 4076 PRINTF("\n"); 4077 } 4078 PRINTF("end prev=%d curr=%d\n", prev[0], curr[0]); 4079 4080 auring_take(src, src->used); 4081 auring_push(dst, i); 4082 4083 /* Adjust */ 4084 t += track->freq_leap; 4085 4086 track->freq_current = t; 4087 for (ch = 0; ch < channels; ch++) { 4088 track->freq_prev[ch] = prev[ch]; 4089 track->freq_curr[ch] = curr[ch]; 4090 } 4091 } 4092 4093 /* 4094 * This filter performs frequency conversion (down sampling). 4095 * It uses simple thinning. 4096 */ 4097 static void 4098 audio_track_freq_down(audio_filter_arg_t *arg) 4099 { 4100 audio_track_t *track; 4101 audio_ring_t *src; 4102 audio_ring_t *dst; 4103 const aint_t *s0; 4104 aint_t *d; 4105 u_int i; 4106 u_int t; 4107 u_int step; 4108 u_int ch; 4109 u_int channels; 4110 4111 track = arg->context; 4112 KASSERT(track); 4113 src = &track->freq.srcbuf; 4114 dst = track->freq.dst; 4115 4116 DIAGNOSTIC_ring(dst); 4117 DIAGNOSTIC_ring(src); 4118 KASSERT(src->used > 0); 4119 KASSERTMSG(src->fmt.channels == dst->fmt.channels, 4120 "src->fmt.channels=%d dst->fmt.channels=%d", 4121 src->fmt.channels, dst->fmt.channels); 4122 KASSERTMSG(src->head % track->mixer->frames_per_block == 0, 4123 "src->head=%d track->mixer->frames_per_block=%d", 4124 src->head, track->mixer->frames_per_block); 4125 4126 s0 = arg->src; 4127 d = arg->dst; 4128 t = track->freq_current; 4129 step = track->freq_step; 4130 channels = dst->fmt.channels; 4131 PRINTF("downstart step=%d leap=%d", step, track->freq_leap); 4132 PRINTF(" srcused=%d arg->count=%u", src->used, arg->count); 4133 PRINTF(" t=%d\n", t); 4134 4135 for (i = 0; i < arg->count && t / 65536 < src->used; i++) { 4136 const aint_t *s; 4137 PRINTF("i=%4d t=%10d", i, t); 4138 s = s0 + (t / 65536) * channels; 4139 PRINTF(" s=%5ld", (s - s0) / channels); 4140 for (ch = 0; ch < channels; ch++) { 4141 if (ch == 0) PRINTF(" *s=%d", s[ch]); 4142 *d++ = s[ch]; 4143 } 4144 PRINTF("\n"); 4145 t += step; 4146 } 4147 t += track->freq_leap; 4148 PRINTF("end t=%d\n", t); 4149 auring_take(src, src->used); 4150 auring_push(dst, i); 4151 track->freq_current = t % 65536; 4152 } 4153 4154 /* 4155 * Creates track and returns it. 4156 * Must be called without sc_lock held. 4157 */ 4158 audio_track_t * 4159 audio_track_create(struct audio_softc *sc, audio_trackmixer_t *mixer) 4160 { 4161 audio_track_t *track; 4162 static int newid = 0; 4163 4164 track = kmem_zalloc(sizeof(*track), KM_SLEEP); 4165 4166 track->id = newid++; 4167 track->mixer = mixer; 4168 track->mode = mixer->mode; 4169 4170 /* Do TRACE after id is assigned. */ 4171 TRACET(3, track, "for %s", 4172 mixer->mode == AUMODE_PLAY ? "playback" : "recording"); 4173 4174 #if defined(AUDIO_SUPPORT_TRACK_VOLUME) 4175 track->volume = 256; 4176 #endif 4177 for (int i = 0; i < AUDIO_MAX_CHANNELS; i++) { 4178 track->ch_volume[i] = 256; 4179 } 4180 4181 return track; 4182 } 4183 4184 /* 4185 * Release all resources of the track and track itself. 4186 * track must not be NULL. Don't specify the track within the file 4187 * structure linked from sc->sc_files. 4188 */ 4189 static void 4190 audio_track_destroy(audio_track_t *track) 4191 { 4192 4193 KASSERT(track); 4194 4195 audio_free_usrbuf(track); 4196 audio_free(track->codec.srcbuf.mem); 4197 audio_free(track->chvol.srcbuf.mem); 4198 audio_free(track->chmix.srcbuf.mem); 4199 audio_free(track->freq.srcbuf.mem); 4200 audio_free(track->outbuf.mem); 4201 4202 kmem_free(track, sizeof(*track)); 4203 } 4204 4205 /* 4206 * It returns encoding conversion filter according to src and dst format. 4207 * If it is not a convertible pair, it returns NULL. Either src or dst 4208 * must be internal format. 4209 */ 4210 static audio_filter_t 4211 audio_track_get_codec(audio_track_t *track, const audio_format2_t *src, 4212 const audio_format2_t *dst) 4213 { 4214 4215 if (audio_format2_is_internal(src)) { 4216 if (dst->encoding == AUDIO_ENCODING_ULAW) { 4217 return audio_internal_to_mulaw; 4218 } else if (dst->encoding == AUDIO_ENCODING_ALAW) { 4219 return audio_internal_to_alaw; 4220 } else if (audio_format2_is_linear(dst)) { 4221 switch (dst->stride) { 4222 case 8: 4223 return audio_internal_to_linear8; 4224 case 16: 4225 return audio_internal_to_linear16; 4226 #if defined(AUDIO_SUPPORT_LINEAR24) 4227 case 24: 4228 return audio_internal_to_linear24; 4229 #endif 4230 case 32: 4231 return audio_internal_to_linear32; 4232 default: 4233 TRACET(1, track, "unsupported %s stride %d", 4234 "dst", dst->stride); 4235 goto abort; 4236 } 4237 } 4238 } else if (audio_format2_is_internal(dst)) { 4239 if (src->encoding == AUDIO_ENCODING_ULAW) { 4240 return audio_mulaw_to_internal; 4241 } else if (src->encoding == AUDIO_ENCODING_ALAW) { 4242 return audio_alaw_to_internal; 4243 } else if (audio_format2_is_linear(src)) { 4244 switch (src->stride) { 4245 case 8: 4246 return audio_linear8_to_internal; 4247 case 16: 4248 return audio_linear16_to_internal; 4249 #if defined(AUDIO_SUPPORT_LINEAR24) 4250 case 24: 4251 return audio_linear24_to_internal; 4252 #endif 4253 case 32: 4254 return audio_linear32_to_internal; 4255 default: 4256 TRACET(1, track, "unsupported %s stride %d", 4257 "src", src->stride); 4258 goto abort; 4259 } 4260 } 4261 } 4262 4263 TRACET(1, track, "unsupported encoding"); 4264 abort: 4265 #if defined(AUDIO_DEBUG) 4266 if (audiodebug >= 2) { 4267 char buf[100]; 4268 audio_format2_tostr(buf, sizeof(buf), src); 4269 TRACET(2, track, "src %s", buf); 4270 audio_format2_tostr(buf, sizeof(buf), dst); 4271 TRACET(2, track, "dst %s", buf); 4272 } 4273 #endif 4274 return NULL; 4275 } 4276 4277 /* 4278 * Initialize the codec stage of this track as necessary. 4279 * If successful, it initializes the codec stage as necessary, stores updated 4280 * last_dst in *last_dstp in any case, and returns 0. 4281 * Otherwise, it returns errno without modifying *last_dstp. 4282 */ 4283 static int 4284 audio_track_init_codec(audio_track_t *track, audio_ring_t **last_dstp) 4285 { 4286 audio_ring_t *last_dst; 4287 audio_ring_t *srcbuf; 4288 audio_format2_t *srcfmt; 4289 audio_format2_t *dstfmt; 4290 audio_filter_arg_t *arg; 4291 u_int len; 4292 int error; 4293 4294 KASSERT(track); 4295 4296 last_dst = *last_dstp; 4297 dstfmt = &last_dst->fmt; 4298 srcfmt = &track->inputfmt; 4299 srcbuf = &track->codec.srcbuf; 4300 error = 0; 4301 4302 if (srcfmt->encoding != dstfmt->encoding 4303 || srcfmt->precision != dstfmt->precision 4304 || srcfmt->stride != dstfmt->stride) { 4305 track->codec.dst = last_dst; 4306 4307 srcbuf->fmt = *dstfmt; 4308 srcbuf->fmt.encoding = srcfmt->encoding; 4309 srcbuf->fmt.precision = srcfmt->precision; 4310 srcbuf->fmt.stride = srcfmt->stride; 4311 4312 track->codec.filter = audio_track_get_codec(track, 4313 &srcbuf->fmt, dstfmt); 4314 if (track->codec.filter == NULL) { 4315 error = EINVAL; 4316 goto abort; 4317 } 4318 4319 srcbuf->head = 0; 4320 srcbuf->used = 0; 4321 srcbuf->capacity = frame_per_block(track->mixer, &srcbuf->fmt); 4322 len = auring_bytelen(srcbuf); 4323 srcbuf->mem = audio_realloc(srcbuf->mem, len); 4324 4325 arg = &track->codec.arg; 4326 arg->srcfmt = &srcbuf->fmt; 4327 arg->dstfmt = dstfmt; 4328 arg->context = NULL; 4329 4330 *last_dstp = srcbuf; 4331 return 0; 4332 } 4333 4334 abort: 4335 track->codec.filter = NULL; 4336 audio_free(srcbuf->mem); 4337 return error; 4338 } 4339 4340 /* 4341 * Initialize the chvol stage of this track as necessary. 4342 * If successful, it initializes the chvol stage as necessary, stores updated 4343 * last_dst in *last_dstp in any case, and returns 0. 4344 * Otherwise, it returns errno without modifying *last_dstp. 4345 */ 4346 static int 4347 audio_track_init_chvol(audio_track_t *track, audio_ring_t **last_dstp) 4348 { 4349 audio_ring_t *last_dst; 4350 audio_ring_t *srcbuf; 4351 audio_format2_t *srcfmt; 4352 audio_format2_t *dstfmt; 4353 audio_filter_arg_t *arg; 4354 u_int len; 4355 int error; 4356 4357 KASSERT(track); 4358 4359 last_dst = *last_dstp; 4360 dstfmt = &last_dst->fmt; 4361 srcfmt = &track->inputfmt; 4362 srcbuf = &track->chvol.srcbuf; 4363 error = 0; 4364 4365 /* Check whether channel volume conversion is necessary. */ 4366 bool use_chvol = false; 4367 for (int ch = 0; ch < srcfmt->channels; ch++) { 4368 if (track->ch_volume[ch] != 256) { 4369 use_chvol = true; 4370 break; 4371 } 4372 } 4373 4374 if (use_chvol == true) { 4375 track->chvol.dst = last_dst; 4376 track->chvol.filter = audio_track_chvol; 4377 4378 srcbuf->fmt = *dstfmt; 4379 /* no format conversion occurs */ 4380 4381 srcbuf->head = 0; 4382 srcbuf->used = 0; 4383 srcbuf->capacity = frame_per_block(track->mixer, &srcbuf->fmt); 4384 len = auring_bytelen(srcbuf); 4385 srcbuf->mem = audio_realloc(srcbuf->mem, len); 4386 4387 arg = &track->chvol.arg; 4388 arg->srcfmt = &srcbuf->fmt; 4389 arg->dstfmt = dstfmt; 4390 arg->context = track->ch_volume; 4391 4392 *last_dstp = srcbuf; 4393 return 0; 4394 } 4395 4396 track->chvol.filter = NULL; 4397 audio_free(srcbuf->mem); 4398 return error; 4399 } 4400 4401 /* 4402 * Initialize the chmix stage of this track as necessary. 4403 * If successful, it initializes the chmix stage as necessary, stores updated 4404 * last_dst in *last_dstp in any case, and returns 0. 4405 * Otherwise, it returns errno without modifying *last_dstp. 4406 */ 4407 static int 4408 audio_track_init_chmix(audio_track_t *track, audio_ring_t **last_dstp) 4409 { 4410 audio_ring_t *last_dst; 4411 audio_ring_t *srcbuf; 4412 audio_format2_t *srcfmt; 4413 audio_format2_t *dstfmt; 4414 audio_filter_arg_t *arg; 4415 u_int srcch; 4416 u_int dstch; 4417 u_int len; 4418 int error; 4419 4420 KASSERT(track); 4421 4422 last_dst = *last_dstp; 4423 dstfmt = &last_dst->fmt; 4424 srcfmt = &track->inputfmt; 4425 srcbuf = &track->chmix.srcbuf; 4426 error = 0; 4427 4428 srcch = srcfmt->channels; 4429 dstch = dstfmt->channels; 4430 if (srcch != dstch) { 4431 track->chmix.dst = last_dst; 4432 4433 if (srcch >= 2 && dstch == 1) { 4434 track->chmix.filter = audio_track_chmix_mixLR; 4435 } else if (srcch == 1 && dstch >= 2) { 4436 track->chmix.filter = audio_track_chmix_dupLR; 4437 } else if (srcch > dstch) { 4438 track->chmix.filter = audio_track_chmix_shrink; 4439 } else { 4440 track->chmix.filter = audio_track_chmix_expand; 4441 } 4442 4443 srcbuf->fmt = *dstfmt; 4444 srcbuf->fmt.channels = srcch; 4445 4446 srcbuf->head = 0; 4447 srcbuf->used = 0; 4448 /* XXX The buffer size should be able to calculate. */ 4449 srcbuf->capacity = frame_per_block(track->mixer, &srcbuf->fmt); 4450 len = auring_bytelen(srcbuf); 4451 srcbuf->mem = audio_realloc(srcbuf->mem, len); 4452 4453 arg = &track->chmix.arg; 4454 arg->srcfmt = &srcbuf->fmt; 4455 arg->dstfmt = dstfmt; 4456 arg->context = NULL; 4457 4458 *last_dstp = srcbuf; 4459 return 0; 4460 } 4461 4462 track->chmix.filter = NULL; 4463 audio_free(srcbuf->mem); 4464 return error; 4465 } 4466 4467 /* 4468 * Initialize the freq stage of this track as necessary. 4469 * If successful, it initializes the freq stage as necessary, stores updated 4470 * last_dst in *last_dstp in any case, and returns 0. 4471 * Otherwise, it returns errno without modifying *last_dstp. 4472 */ 4473 static int 4474 audio_track_init_freq(audio_track_t *track, audio_ring_t **last_dstp) 4475 { 4476 audio_ring_t *last_dst; 4477 audio_ring_t *srcbuf; 4478 audio_format2_t *srcfmt; 4479 audio_format2_t *dstfmt; 4480 audio_filter_arg_t *arg; 4481 uint32_t srcfreq; 4482 uint32_t dstfreq; 4483 u_int dst_capacity; 4484 u_int mod; 4485 u_int len; 4486 int error; 4487 4488 KASSERT(track); 4489 4490 last_dst = *last_dstp; 4491 dstfmt = &last_dst->fmt; 4492 srcfmt = &track->inputfmt; 4493 srcbuf = &track->freq.srcbuf; 4494 error = 0; 4495 4496 srcfreq = srcfmt->sample_rate; 4497 dstfreq = dstfmt->sample_rate; 4498 if (srcfreq != dstfreq) { 4499 track->freq.dst = last_dst; 4500 4501 memset(track->freq_prev, 0, sizeof(track->freq_prev)); 4502 memset(track->freq_curr, 0, sizeof(track->freq_curr)); 4503 4504 /* freq_step is the ratio of src/dst when let dst 65536. */ 4505 track->freq_step = (uint64_t)srcfreq * 65536 / dstfreq; 4506 4507 dst_capacity = frame_per_block(track->mixer, dstfmt); 4508 mod = (uint64_t)srcfreq * 65536 % dstfreq; 4509 track->freq_leap = (mod * dst_capacity + dstfreq / 2) / dstfreq; 4510 4511 if (track->freq_step < 65536) { 4512 track->freq.filter = audio_track_freq_up; 4513 /* In order to carry at the first time. */ 4514 track->freq_current = 65536; 4515 } else { 4516 track->freq.filter = audio_track_freq_down; 4517 track->freq_current = 0; 4518 } 4519 4520 srcbuf->fmt = *dstfmt; 4521 srcbuf->fmt.sample_rate = srcfreq; 4522 4523 srcbuf->head = 0; 4524 srcbuf->used = 0; 4525 srcbuf->capacity = frame_per_block(track->mixer, &srcbuf->fmt); 4526 len = auring_bytelen(srcbuf); 4527 srcbuf->mem = audio_realloc(srcbuf->mem, len); 4528 4529 arg = &track->freq.arg; 4530 arg->srcfmt = &srcbuf->fmt; 4531 arg->dstfmt = dstfmt; 4532 arg->context = track; 4533 4534 *last_dstp = srcbuf; 4535 return 0; 4536 } 4537 4538 track->freq.filter = NULL; 4539 audio_free(srcbuf->mem); 4540 return error; 4541 } 4542 4543 /* 4544 * There are two unit of buffers; A block buffer and a byte buffer. Both use 4545 * audio_ring_t. Internally, audio data is always handled in block unit. 4546 * Converting format, sythesizing tracks, transferring from/to the hardware, 4547 * and etc. Only one exception is usrbuf. To transfer with userland, usrbuf 4548 * is buffered in byte unit. 4549 * For playing back, write(2) writes arbitrary length of data to usrbuf. 4550 * When one block is filled, it is sent to the next stage (converting and/or 4551 * synthesizing). 4552 * For recording, the rmixer writes one block length of data to input buffer 4553 * (the bottom stage buffer) each time. read(2) (converts one block if usrbuf 4554 * is empty and then) reads arbitrary length of data from usrbuf. 4555 * 4556 * The following charts show the data flow and buffer types for playback and 4557 * recording track. In this example, both have two conversion stages, codec 4558 * and freq. Every [**] represents a buffer described below. 4559 * 4560 * On playback track: 4561 * 4562 * write(2) 4563 * | 4564 * | uiomove 4565 * v 4566 * usrbuf [BB|BB ... BB|BB] .. Byte ring buffer 4567 * | 4568 * | memcpy one block 4569 * v 4570 * codec.srcbuf [FF] .. 1 block (ring) buffer 4571 * .dst ----+ 4572 * | 4573 * | convert 4574 * v 4575 * freq.srcbuf [FF] .. 1 block (ring) buffer 4576 * .dst ----+ 4577 * | 4578 * | convert 4579 * v 4580 * outbuf [FF|FF|FF|FF] .. NBLKOUT blocks ring buffer 4581 * | 4582 * v 4583 * pmixer 4584 * 4585 * There are three different types of buffers: 4586 * 4587 * [BB|BB ... BB|BB] usrbuf. Is the buffer closest to userland. Mandatory. 4588 * This is a byte buffer and its length is basically less 4589 * than or equal to 64KB or at least AUMINNOBLK blocks. 4590 * 4591 * [FF] Interim conversion stage's srcbuf if necessary. 4592 * This is one block (ring) buffer counted in frames. 4593 * 4594 * [FF|FF|FF|FF] outbuf. Is the buffer closest to pmixer. Mandatory. 4595 * This is NBLKOUT blocks ring buffer counted in frames. 4596 * 4597 * 4598 * On recording track: 4599 * 4600 * read(2) 4601 * ^ 4602 * | uiomove 4603 * | 4604 * usrbuf [BB] .. Byte (ring) buffer 4605 * ^ 4606 * | memcpy one block 4607 * | 4608 * outbuf [FF] .. 1 block (ring) buffer 4609 * ^ 4610 * | convert 4611 * | 4612 * codec.dst ----+ 4613 * .srcbuf [FF] .. 1 block (ring) buffer 4614 * ^ 4615 * | convert 4616 * | 4617 * freq.dst ----+ 4618 * .srcbuf [FF|FF ... FF|FF] .. NBLKIN blocks ring buffer 4619 * ^ 4620 * | 4621 * rmixer 4622 * 4623 * There are also three different types of buffers. 4624 * 4625 * [BB] usrbuf. Is the buffer closest to userland. Mandatory. 4626 * This is a byte buffer and its length is one block. 4627 * This buffer holds only "fragment". 4628 * 4629 * [FF] Interim conversion stage's srcbuf (or outbuf). 4630 * This is one block (ring) buffer counted in frames. 4631 * 4632 * [FF|FF ... FF|FF] The bottom conversion stage's srcbuf (or outbuf). 4633 * This is the buffer closest to rmixer, and mandatory. 4634 * This is NBLKIN blocks ring buffer counted in frames. 4635 * Also pointed by *input. 4636 */ 4637 4638 /* 4639 * Set the userland format of this track. 4640 * usrfmt argument should have been previously verified by 4641 * audio_track_setinfo_check(). 4642 * This function may release and reallocate all internal conversion buffers. 4643 * It returns 0 if successful. Otherwise it returns errno with clearing all 4644 * internal buffers. 4645 * It must be called without sc_intr_lock since uvm_* routines require non 4646 * intr_lock state. 4647 * It must be called with track lock held since it may release and reallocate 4648 * outbuf. 4649 */ 4650 static int 4651 audio_track_set_format(audio_track_t *track, audio_format2_t *usrfmt) 4652 { 4653 audio_ring_t *last_dst; 4654 int is_playback; 4655 u_int newbufsize; 4656 u_int newvsize; 4657 u_int len; 4658 int error; 4659 4660 KASSERT(track); 4661 4662 is_playback = audio_track_is_playback(track); 4663 4664 /* Once mmap is called, the track format cannot be changed. */ 4665 if (track->mmapped) 4666 return EIO; 4667 4668 /* usrbuf is the closest buffer to the userland. */ 4669 track->usrbuf.fmt = *usrfmt; 4670 4671 /* 4672 * Usrbuf. 4673 * On the playback track, its capacity is less than or equal to 64KB 4674 * (for historical reason) and must be a multiple of a block 4675 * (constraint in this implementation). But at least AUMINNOBLK 4676 * blocks. 4677 * On the recording track, its capacity is one block. 4678 */ 4679 /* 4680 * For references, one block size (in 40msec) is: 4681 * 320 bytes = 204 blocks/64KB for mulaw/8kHz/1ch 4682 * 7680 bytes = 8 blocks/64KB for s16/48kHz/2ch 4683 * 30720 bytes = 90 KB/3blocks for s16/48kHz/8ch 4684 * 61440 bytes = 180 KB/3blocks for s16/96kHz/8ch 4685 * 245760 bytes = 720 KB/3blocks for s32/192kHz/8ch 4686 * 4687 * For example, 4688 * 1) If usrbuf_blksize = 7056 (s16/44.1k/2ch) and PAGE_SIZE = 8192, 4689 * newbufsize = rounddown(65536 / 7056) = 63504 4690 * newvsize = roundup2(63504, PAGE_SIZE) = 65536 4691 * Therefore it maps 8 * 8K pages and usrbuf->capacity = 63504. 4692 * 4693 * 2) If usrbuf_blksize = 7680 (s16/48k/2ch) and PAGE_SIZE = 4096, 4694 * newbufsize = rounddown(65536 / 7680) = 61440 4695 * newvsize = roundup2(61440, PAGE_SIZE) = 61440 (= 15 pages) 4696 * Therefore it maps 15 * 4K pages and usrbuf->capacity = 61440. 4697 */ 4698 track->usrbuf_blksize = frametobyte(&track->usrbuf.fmt, 4699 frame_per_block(track->mixer, &track->usrbuf.fmt)); 4700 track->usrbuf.head = 0; 4701 track->usrbuf.used = 0; 4702 if (is_playback) { 4703 newbufsize = track->usrbuf_blksize * AUMINNOBLK; 4704 if (newbufsize < 65536) 4705 newbufsize = rounddown(65536, track->usrbuf_blksize); 4706 newvsize = roundup2(newbufsize, PAGE_SIZE); 4707 } else { 4708 newbufsize = track->usrbuf_blksize; 4709 newvsize = track->usrbuf_blksize; 4710 } 4711 /* 4712 * Reallocate only if the number of pages changes. 4713 * This is because we expect kmem to allocate memory on per page 4714 * basis if the request size is about 64KB. 4715 */ 4716 if (newvsize != track->usrbuf_allocsize) { 4717 if (track->usrbuf_allocsize != 0) { 4718 kmem_free(track->usrbuf.mem, track->usrbuf_allocsize); 4719 } 4720 TRACET(2, track, "usrbuf_allocsize %d -> %d", 4721 track->usrbuf_allocsize, newvsize); 4722 track->usrbuf.mem = kmem_alloc(newvsize, KM_SLEEP); 4723 track->usrbuf_allocsize = newvsize; 4724 } 4725 track->usrbuf.capacity = newbufsize; 4726 4727 /* Recalc water mark. */ 4728 if (is_playback) { 4729 /* Set high at 100%, low at 75%. */ 4730 track->usrbuf_usedhigh = track->usrbuf.capacity; 4731 track->usrbuf_usedlow = track->usrbuf.capacity * 3 / 4; 4732 } else { 4733 /* Set high at 100%, low at 0%. (But not used) */ 4734 track->usrbuf_usedhigh = track->usrbuf.capacity; 4735 track->usrbuf_usedlow = 0; 4736 } 4737 4738 /* Stage buffer */ 4739 last_dst = &track->outbuf; 4740 if (is_playback) { 4741 /* On playback, initialize from the mixer side in order. */ 4742 track->inputfmt = *usrfmt; 4743 track->outbuf.fmt = track->mixer->track_fmt; 4744 4745 if ((error = audio_track_init_freq(track, &last_dst)) != 0) 4746 goto error; 4747 if ((error = audio_track_init_chmix(track, &last_dst)) != 0) 4748 goto error; 4749 if ((error = audio_track_init_chvol(track, &last_dst)) != 0) 4750 goto error; 4751 if ((error = audio_track_init_codec(track, &last_dst)) != 0) 4752 goto error; 4753 } else { 4754 /* On recording, initialize from userland side in order. */ 4755 track->inputfmt = track->mixer->track_fmt; 4756 track->outbuf.fmt = *usrfmt; 4757 4758 if ((error = audio_track_init_codec(track, &last_dst)) != 0) 4759 goto error; 4760 if ((error = audio_track_init_chvol(track, &last_dst)) != 0) 4761 goto error; 4762 if ((error = audio_track_init_chmix(track, &last_dst)) != 0) 4763 goto error; 4764 if ((error = audio_track_init_freq(track, &last_dst)) != 0) 4765 goto error; 4766 } 4767 #if 0 4768 /* debug */ 4769 if (track->freq.filter) { 4770 audio_print_format2("freq src", &track->freq.srcbuf.fmt); 4771 audio_print_format2("freq dst", &track->freq.dst->fmt); 4772 } 4773 if (track->chmix.filter) { 4774 audio_print_format2("chmix src", &track->chmix.srcbuf.fmt); 4775 audio_print_format2("chmix dst", &track->chmix.dst->fmt); 4776 } 4777 if (track->chvol.filter) { 4778 audio_print_format2("chvol src", &track->chvol.srcbuf.fmt); 4779 audio_print_format2("chvol dst", &track->chvol.dst->fmt); 4780 } 4781 if (track->codec.filter) { 4782 audio_print_format2("codec src", &track->codec.srcbuf.fmt); 4783 audio_print_format2("codec dst", &track->codec.dst->fmt); 4784 } 4785 #endif 4786 4787 /* Stage input buffer */ 4788 track->input = last_dst; 4789 4790 /* 4791 * Output buffer. 4792 * On the playback track, its capacity is NBLKOUT blocks. 4793 * On the recording track, its capacity is 1 block. 4794 */ 4795 track->outbuf.head = 0; 4796 track->outbuf.used = 0; 4797 track->outbuf.capacity = frame_per_block(track->mixer, 4798 &track->outbuf.fmt); 4799 if (is_playback) 4800 track->outbuf.capacity *= NBLKOUT; 4801 len = auring_bytelen(&track->outbuf); 4802 track->outbuf.mem = audio_realloc(track->outbuf.mem, len); 4803 4804 /* 4805 * On the recording track, expand the input stage buffer, which is 4806 * the closest buffer to rmixer, to NBLKIN blocks. 4807 * Note that input buffer may point to outbuf. 4808 */ 4809 if (!is_playback) { 4810 int input_fpb; 4811 4812 input_fpb = frame_per_block(track->mixer, &track->input->fmt); 4813 track->input->capacity = input_fpb * NBLKIN; 4814 len = auring_bytelen(track->input); 4815 track->input->mem = audio_realloc(track->input->mem, len); 4816 } 4817 4818 #if defined(AUDIO_DEBUG) 4819 if (audiodebug >= 3) { 4820 struct audio_track_debugbuf m; 4821 4822 memset(&m, 0, sizeof(m)); 4823 snprintf(m.outbuf, sizeof(m.outbuf), " out=%d", 4824 track->outbuf.capacity * frametobyte(&track->outbuf.fmt,1)); 4825 if (track->freq.filter) 4826 snprintf(m.freq, sizeof(m.freq), " freq=%d", 4827 track->freq.srcbuf.capacity * 4828 frametobyte(&track->freq.srcbuf.fmt, 1)); 4829 if (track->chmix.filter) 4830 snprintf(m.chmix, sizeof(m.chmix), " chmix=%d", 4831 track->chmix.srcbuf.capacity * 4832 frametobyte(&track->chmix.srcbuf.fmt, 1)); 4833 if (track->chvol.filter) 4834 snprintf(m.chvol, sizeof(m.chvol), " chvol=%d", 4835 track->chvol.srcbuf.capacity * 4836 frametobyte(&track->chvol.srcbuf.fmt, 1)); 4837 if (track->codec.filter) 4838 snprintf(m.codec, sizeof(m.codec), " codec=%d", 4839 track->codec.srcbuf.capacity * 4840 frametobyte(&track->codec.srcbuf.fmt, 1)); 4841 snprintf(m.usrbuf, sizeof(m.usrbuf), 4842 " usr=%d", track->usrbuf.capacity); 4843 4844 if (is_playback) { 4845 TRACET(0, track, "bufsize%s%s%s%s%s%s", 4846 m.outbuf, m.freq, m.chmix, 4847 m.chvol, m.codec, m.usrbuf); 4848 } else { 4849 TRACET(0, track, "bufsize%s%s%s%s%s%s", 4850 m.freq, m.chmix, m.chvol, 4851 m.codec, m.outbuf, m.usrbuf); 4852 } 4853 } 4854 #endif 4855 return 0; 4856 4857 error: 4858 audio_free_usrbuf(track); 4859 audio_free(track->codec.srcbuf.mem); 4860 audio_free(track->chvol.srcbuf.mem); 4861 audio_free(track->chmix.srcbuf.mem); 4862 audio_free(track->freq.srcbuf.mem); 4863 audio_free(track->outbuf.mem); 4864 return error; 4865 } 4866 4867 /* 4868 * Fill silence frames (as the internal format) up to 1 block 4869 * if the ring is not empty and less than 1 block. 4870 * It returns the number of appended frames. 4871 */ 4872 static int 4873 audio_append_silence(audio_track_t *track, audio_ring_t *ring) 4874 { 4875 int fpb; 4876 int n; 4877 4878 KASSERT(track); 4879 KASSERT(audio_format2_is_internal(&ring->fmt)); 4880 4881 /* XXX is n correct? */ 4882 /* XXX memset uses frametobyte()? */ 4883 4884 if (ring->used == 0) 4885 return 0; 4886 4887 fpb = frame_per_block(track->mixer, &ring->fmt); 4888 if (ring->used >= fpb) 4889 return 0; 4890 4891 n = (ring->capacity - ring->used) % fpb; 4892 4893 KASSERTMSG(auring_get_contig_free(ring) >= n, 4894 "auring_get_contig_free(ring)=%d n=%d", 4895 auring_get_contig_free(ring), n); 4896 4897 memset(auring_tailptr_aint(ring), 0, 4898 n * ring->fmt.channels * sizeof(aint_t)); 4899 auring_push(ring, n); 4900 return n; 4901 } 4902 4903 /* 4904 * Execute the conversion stage. 4905 * It prepares arg from this stage and executes stage->filter. 4906 * It must be called only if stage->filter is not NULL. 4907 * 4908 * For stages other than frequency conversion, the function increments 4909 * src and dst counters here. For frequency conversion stage, on the 4910 * other hand, the function does not touch src and dst counters and 4911 * filter side has to increment them. 4912 */ 4913 static void 4914 audio_apply_stage(audio_track_t *track, audio_stage_t *stage, bool isfreq) 4915 { 4916 audio_filter_arg_t *arg; 4917 int srccount; 4918 int dstcount; 4919 int count; 4920 4921 KASSERT(track); 4922 KASSERT(stage->filter); 4923 4924 srccount = auring_get_contig_used(&stage->srcbuf); 4925 dstcount = auring_get_contig_free(stage->dst); 4926 4927 if (isfreq) { 4928 KASSERTMSG(srccount > 0, "freq but srccount=%d", srccount); 4929 count = uimin(dstcount, track->mixer->frames_per_block); 4930 } else { 4931 count = uimin(srccount, dstcount); 4932 } 4933 4934 if (count > 0) { 4935 arg = &stage->arg; 4936 arg->src = auring_headptr(&stage->srcbuf); 4937 arg->dst = auring_tailptr(stage->dst); 4938 arg->count = count; 4939 4940 stage->filter(arg); 4941 4942 if (!isfreq) { 4943 auring_take(&stage->srcbuf, count); 4944 auring_push(stage->dst, count); 4945 } 4946 } 4947 } 4948 4949 /* 4950 * Produce output buffer for playback from user input buffer. 4951 * It must be called only if usrbuf is not empty and outbuf is 4952 * available at least one free block. 4953 */ 4954 static void 4955 audio_track_play(audio_track_t *track) 4956 { 4957 audio_ring_t *usrbuf; 4958 audio_ring_t *input; 4959 int count; 4960 int framesize; 4961 int bytes; 4962 4963 KASSERT(track); 4964 KASSERT(track->lock); 4965 TRACET(4, track, "start pstate=%d", track->pstate); 4966 4967 /* At this point usrbuf must not be empty. */ 4968 KASSERT(track->usrbuf.used > 0); 4969 /* Also, outbuf must be available at least one block. */ 4970 count = auring_get_contig_free(&track->outbuf); 4971 KASSERTMSG(count >= frame_per_block(track->mixer, &track->outbuf.fmt), 4972 "count=%d fpb=%d", 4973 count, frame_per_block(track->mixer, &track->outbuf.fmt)); 4974 4975 usrbuf = &track->usrbuf; 4976 input = track->input; 4977 4978 /* 4979 * framesize is always 1 byte or more since all formats supported as 4980 * usrfmt(=input) have 8bit or more stride. 4981 */ 4982 framesize = frametobyte(&input->fmt, 1); 4983 KASSERT(framesize >= 1); 4984 4985 /* The next stage of usrbuf (=input) must be available. */ 4986 KASSERT(auring_get_contig_free(input) > 0); 4987 4988 /* 4989 * Copy usrbuf up to 1block to input buffer. 4990 * count is the number of frames to copy from usrbuf. 4991 * bytes is the number of bytes to copy from usrbuf. However it is 4992 * not copied less than one frame. 4993 */ 4994 count = uimin(usrbuf->used, track->usrbuf_blksize) / framesize; 4995 bytes = count * framesize; 4996 4997 if (usrbuf->head + bytes < usrbuf->capacity) { 4998 memcpy((uint8_t *)input->mem + auring_tail(input) * framesize, 4999 (uint8_t *)usrbuf->mem + usrbuf->head, 5000 bytes); 5001 auring_push(input, count); 5002 auring_take(usrbuf, bytes); 5003 } else { 5004 int bytes1; 5005 int bytes2; 5006 5007 bytes1 = auring_get_contig_used(usrbuf); 5008 KASSERTMSG(bytes1 % framesize == 0, 5009 "bytes1=%d framesize=%d", bytes1, framesize); 5010 memcpy((uint8_t *)input->mem + auring_tail(input) * framesize, 5011 (uint8_t *)usrbuf->mem + usrbuf->head, 5012 bytes1); 5013 auring_push(input, bytes1 / framesize); 5014 auring_take(usrbuf, bytes1); 5015 5016 bytes2 = bytes - bytes1; 5017 memcpy((uint8_t *)input->mem + auring_tail(input) * framesize, 5018 (uint8_t *)usrbuf->mem + usrbuf->head, 5019 bytes2); 5020 auring_push(input, bytes2 / framesize); 5021 auring_take(usrbuf, bytes2); 5022 } 5023 5024 /* Encoding conversion */ 5025 if (track->codec.filter) 5026 audio_apply_stage(track, &track->codec, false); 5027 5028 /* Channel volume */ 5029 if (track->chvol.filter) 5030 audio_apply_stage(track, &track->chvol, false); 5031 5032 /* Channel mix */ 5033 if (track->chmix.filter) 5034 audio_apply_stage(track, &track->chmix, false); 5035 5036 /* Frequency conversion */ 5037 /* 5038 * Since the frequency conversion needs correction for each block, 5039 * it rounds up to 1 block. 5040 */ 5041 if (track->freq.filter) { 5042 int n; 5043 n = audio_append_silence(track, &track->freq.srcbuf); 5044 if (n > 0) { 5045 TRACET(4, track, 5046 "freq.srcbuf add silence %d -> %d/%d/%d", 5047 n, 5048 track->freq.srcbuf.head, 5049 track->freq.srcbuf.used, 5050 track->freq.srcbuf.capacity); 5051 } 5052 if (track->freq.srcbuf.used > 0) { 5053 audio_apply_stage(track, &track->freq, true); 5054 } 5055 } 5056 5057 if (bytes < track->usrbuf_blksize) { 5058 /* 5059 * Clear all conversion buffer pointer if the conversion was 5060 * not exactly one block. These conversion stage buffers are 5061 * certainly circular buffers because of symmetry with the 5062 * previous and next stage buffer. However, since they are 5063 * treated as simple contiguous buffers in operation, so head 5064 * always should point 0. This may happen during drain-age. 5065 */ 5066 TRACET(4, track, "reset stage"); 5067 if (track->codec.filter) { 5068 KASSERT(track->codec.srcbuf.used == 0); 5069 track->codec.srcbuf.head = 0; 5070 } 5071 if (track->chvol.filter) { 5072 KASSERT(track->chvol.srcbuf.used == 0); 5073 track->chvol.srcbuf.head = 0; 5074 } 5075 if (track->chmix.filter) { 5076 KASSERT(track->chmix.srcbuf.used == 0); 5077 track->chmix.srcbuf.head = 0; 5078 } 5079 if (track->freq.filter) { 5080 KASSERT(track->freq.srcbuf.used == 0); 5081 track->freq.srcbuf.head = 0; 5082 } 5083 } 5084 5085 track->stamp++; 5086 5087 #if defined(AUDIO_DEBUG) 5088 if (audiodebug >= 3) { 5089 struct audio_track_debugbuf m; 5090 audio_track_bufstat(track, &m); 5091 TRACET(0, track, "end%s%s%s%s%s%s", 5092 m.outbuf, m.freq, m.chvol, m.chmix, m.codec, m.usrbuf); 5093 } 5094 #endif 5095 } 5096 5097 /* 5098 * Produce user output buffer for recording from input buffer. 5099 */ 5100 static void 5101 audio_track_record(audio_track_t *track) 5102 { 5103 audio_ring_t *outbuf; 5104 audio_ring_t *usrbuf; 5105 int count; 5106 int bytes; 5107 int framesize; 5108 5109 KASSERT(track); 5110 KASSERT(track->lock); 5111 5112 if (auring_get_contig_used(track->input) == 0) { 5113 TRACET(4, track, "input->used == 0"); 5114 return; 5115 } 5116 5117 /* Frequency conversion */ 5118 if (track->freq.filter) { 5119 if (track->freq.srcbuf.used > 0) { 5120 audio_apply_stage(track, &track->freq, true); 5121 /* XXX should input of freq be from beginning of buf? */ 5122 } 5123 } 5124 5125 /* Channel mix */ 5126 if (track->chmix.filter) 5127 audio_apply_stage(track, &track->chmix, false); 5128 5129 /* Channel volume */ 5130 if (track->chvol.filter) 5131 audio_apply_stage(track, &track->chvol, false); 5132 5133 /* Encoding conversion */ 5134 if (track->codec.filter) 5135 audio_apply_stage(track, &track->codec, false); 5136 5137 /* Copy outbuf to usrbuf */ 5138 outbuf = &track->outbuf; 5139 usrbuf = &track->usrbuf; 5140 /* usrbuf should be empty. */ 5141 KASSERT(usrbuf->used == 0); 5142 /* 5143 * framesize is always 1 byte or more since all formats supported 5144 * as usrfmt(=output) have 8bit or more stride. 5145 */ 5146 framesize = frametobyte(&outbuf->fmt, 1); 5147 KASSERT(framesize >= 1); 5148 /* 5149 * count is the number of frames to copy to usrbuf. 5150 * bytes is the number of bytes to copy to usrbuf. 5151 */ 5152 count = outbuf->used; 5153 count = uimin(count, track->usrbuf_blksize / framesize); 5154 bytes = count * framesize; 5155 if (auring_tail(usrbuf) + bytes < usrbuf->capacity) { 5156 memcpy((uint8_t *)usrbuf->mem + auring_tail(usrbuf), 5157 (uint8_t *)outbuf->mem + outbuf->head * framesize, 5158 bytes); 5159 auring_push(usrbuf, bytes); 5160 auring_take(outbuf, count); 5161 } else { 5162 int bytes1; 5163 int bytes2; 5164 5165 bytes1 = auring_get_contig_free(usrbuf); 5166 KASSERTMSG(bytes1 % framesize == 0, 5167 "bytes1=%d framesize=%d", bytes1, framesize); 5168 memcpy((uint8_t *)usrbuf->mem + auring_tail(usrbuf), 5169 (uint8_t *)outbuf->mem + outbuf->head * framesize, 5170 bytes1); 5171 auring_push(usrbuf, bytes1); 5172 auring_take(outbuf, bytes1 / framesize); 5173 5174 bytes2 = bytes - bytes1; 5175 memcpy((uint8_t *)usrbuf->mem + auring_tail(usrbuf), 5176 (uint8_t *)outbuf->mem + outbuf->head * framesize, 5177 bytes2); 5178 auring_push(usrbuf, bytes2); 5179 auring_take(outbuf, bytes2 / framesize); 5180 } 5181 5182 #if defined(AUDIO_DEBUG) 5183 if (audiodebug >= 3) { 5184 struct audio_track_debugbuf m; 5185 audio_track_bufstat(track, &m); 5186 TRACET(0, track, "end%s%s%s%s%s%s", 5187 m.freq, m.chvol, m.chmix, m.codec, m.outbuf, m.usrbuf); 5188 } 5189 #endif 5190 } 5191 5192 /* 5193 * Calculate blktime [msec] from mixer(.hwbuf.fmt). 5194 * Must be called with sc_exlock held. 5195 */ 5196 static u_int 5197 audio_mixer_calc_blktime(struct audio_softc *sc, audio_trackmixer_t *mixer) 5198 { 5199 audio_format2_t *fmt; 5200 u_int blktime; 5201 u_int frames_per_block; 5202 5203 KASSERT(sc->sc_exlock); 5204 5205 fmt = &mixer->hwbuf.fmt; 5206 blktime = sc->sc_blk_ms; 5207 5208 /* 5209 * If stride is not multiples of 8, special treatment is necessary. 5210 * For now, it is only x68k's vs(4), 4 bit/sample ADPCM. 5211 */ 5212 if (fmt->stride == 4) { 5213 frames_per_block = fmt->sample_rate * blktime / 1000; 5214 if ((frames_per_block & 1) != 0) 5215 blktime *= 2; 5216 } 5217 #ifdef DIAGNOSTIC 5218 else if (fmt->stride % NBBY != 0) { 5219 panic("unsupported HW stride %d", fmt->stride); 5220 } 5221 #endif 5222 5223 return blktime; 5224 } 5225 5226 /* 5227 * Initialize the mixer corresponding to the mode. 5228 * Set AUMODE_PLAY to the 'mode' for playback or AUMODE_RECORD for recording. 5229 * sc->sc_[pr]mixer (corresponding to the 'mode') must be zero-filled. 5230 * This function returns 0 on successful. Otherwise returns errno. 5231 * Must be called with sc_exlock held and without sc_lock held. 5232 */ 5233 static int 5234 audio_mixer_init(struct audio_softc *sc, int mode, 5235 const audio_format2_t *hwfmt, const audio_filter_reg_t *reg) 5236 { 5237 char codecbuf[64]; 5238 char blkdmsbuf[8]; 5239 audio_trackmixer_t *mixer; 5240 void (*softint_handler)(void *); 5241 int len; 5242 int blksize; 5243 int capacity; 5244 size_t bufsize; 5245 int hwblks; 5246 int blkms; 5247 int blkdms; 5248 int error; 5249 5250 KASSERT(hwfmt != NULL); 5251 KASSERT(reg != NULL); 5252 KASSERT(sc->sc_exlock); 5253 5254 error = 0; 5255 if (mode == AUMODE_PLAY) 5256 mixer = sc->sc_pmixer; 5257 else 5258 mixer = sc->sc_rmixer; 5259 5260 mixer->sc = sc; 5261 mixer->mode = mode; 5262 5263 mixer->hwbuf.fmt = *hwfmt; 5264 mixer->volume = 256; 5265 mixer->blktime_d = 1000; 5266 mixer->blktime_n = audio_mixer_calc_blktime(sc, mixer); 5267 sc->sc_blk_ms = mixer->blktime_n; 5268 hwblks = NBLKHW; 5269 5270 mixer->frames_per_block = frame_per_block(mixer, &mixer->hwbuf.fmt); 5271 blksize = frametobyte(&mixer->hwbuf.fmt, mixer->frames_per_block); 5272 if (sc->hw_if->round_blocksize) { 5273 int rounded; 5274 audio_params_t p = format2_to_params(&mixer->hwbuf.fmt); 5275 mutex_enter(sc->sc_lock); 5276 rounded = sc->hw_if->round_blocksize(sc->hw_hdl, blksize, 5277 mode, &p); 5278 mutex_exit(sc->sc_lock); 5279 TRACE(1, "round_blocksize %d -> %d", blksize, rounded); 5280 if (rounded != blksize) { 5281 if ((rounded * NBBY) % (mixer->hwbuf.fmt.stride * 5282 mixer->hwbuf.fmt.channels) != 0) { 5283 audio_printf(sc, 5284 "round_blocksize returned blocksize " 5285 "indivisible by framesize: " 5286 "blksize=%d rounded=%d " 5287 "stride=%ubit channels=%u\n", 5288 blksize, rounded, 5289 mixer->hwbuf.fmt.stride, 5290 mixer->hwbuf.fmt.channels); 5291 return EINVAL; 5292 } 5293 /* Recalculation */ 5294 blksize = rounded; 5295 mixer->frames_per_block = blksize * NBBY / 5296 (mixer->hwbuf.fmt.stride * 5297 mixer->hwbuf.fmt.channels); 5298 } 5299 } 5300 mixer->blktime_n = mixer->frames_per_block; 5301 mixer->blktime_d = mixer->hwbuf.fmt.sample_rate; 5302 5303 capacity = mixer->frames_per_block * hwblks; 5304 bufsize = frametobyte(&mixer->hwbuf.fmt, capacity); 5305 if (sc->hw_if->round_buffersize) { 5306 size_t rounded; 5307 mutex_enter(sc->sc_lock); 5308 rounded = sc->hw_if->round_buffersize(sc->hw_hdl, mode, 5309 bufsize); 5310 mutex_exit(sc->sc_lock); 5311 TRACE(1, "round_buffersize %zd -> %zd", bufsize, rounded); 5312 if (rounded < bufsize) { 5313 /* buffersize needs NBLKHW blocks at least. */ 5314 audio_printf(sc, 5315 "round_buffersize returned too small buffersize: " 5316 "buffersize=%zd blksize=%d\n", 5317 rounded, blksize); 5318 return EINVAL; 5319 } 5320 if (rounded % blksize != 0) { 5321 /* buffersize/blksize constraint mismatch? */ 5322 audio_printf(sc, 5323 "round_buffersize returned buffersize indivisible " 5324 "by blksize: buffersize=%zu blksize=%d\n", 5325 rounded, blksize); 5326 return EINVAL; 5327 } 5328 if (rounded != bufsize) { 5329 /* Recalculation */ 5330 bufsize = rounded; 5331 hwblks = bufsize / blksize; 5332 capacity = mixer->frames_per_block * hwblks; 5333 } 5334 } 5335 TRACE(1, "buffersize for %s = %zu", 5336 (mode == AUMODE_PLAY) ? "playback" : "recording", 5337 bufsize); 5338 mixer->hwbuf.capacity = capacity; 5339 5340 if (sc->hw_if->allocm) { 5341 /* sc_lock is not necessary for allocm */ 5342 mixer->hwbuf.mem = sc->hw_if->allocm(sc->hw_hdl, mode, bufsize); 5343 if (mixer->hwbuf.mem == NULL) { 5344 audio_printf(sc, "allocm(%zu) failed\n", bufsize); 5345 return ENOMEM; 5346 } 5347 } else { 5348 mixer->hwbuf.mem = kmem_alloc(bufsize, KM_SLEEP); 5349 } 5350 5351 /* From here, audio_mixer_destroy is necessary to exit. */ 5352 if (mode == AUMODE_PLAY) { 5353 cv_init(&mixer->outcv, "audiowr"); 5354 } else { 5355 cv_init(&mixer->outcv, "audiord"); 5356 } 5357 5358 if (mode == AUMODE_PLAY) { 5359 softint_handler = audio_softintr_wr; 5360 } else { 5361 softint_handler = audio_softintr_rd; 5362 } 5363 mixer->sih = softint_establish(SOFTINT_SERIAL | SOFTINT_MPSAFE, 5364 softint_handler, sc); 5365 if (mixer->sih == NULL) { 5366 device_printf(sc->sc_dev, "softint_establish failed\n"); 5367 goto abort; 5368 } 5369 5370 mixer->track_fmt.encoding = AUDIO_ENCODING_SLINEAR_NE; 5371 mixer->track_fmt.precision = AUDIO_INTERNAL_BITS; 5372 mixer->track_fmt.stride = AUDIO_INTERNAL_BITS; 5373 mixer->track_fmt.channels = mixer->hwbuf.fmt.channels; 5374 mixer->track_fmt.sample_rate = mixer->hwbuf.fmt.sample_rate; 5375 5376 if (mixer->hwbuf.fmt.encoding == AUDIO_ENCODING_SLINEAR_OE && 5377 mixer->hwbuf.fmt.precision == AUDIO_INTERNAL_BITS) { 5378 mixer->swap_endian = true; 5379 TRACE(1, "swap_endian"); 5380 } 5381 5382 if (mode == AUMODE_PLAY) { 5383 /* Mixing buffer */ 5384 mixer->mixfmt = mixer->track_fmt; 5385 mixer->mixfmt.precision *= 2; 5386 mixer->mixfmt.stride *= 2; 5387 /* XXX TODO: use some macros? */ 5388 len = mixer->frames_per_block * mixer->mixfmt.channels * 5389 mixer->mixfmt.stride / NBBY; 5390 mixer->mixsample = audio_realloc(mixer->mixsample, len); 5391 } else if (reg->codec == NULL) { 5392 /* 5393 * Recording requires an input conversion buffer 5394 * unless the hardware provides a codec itself 5395 */ 5396 mixer->mixfmt = mixer->track_fmt; 5397 len = mixer->frames_per_block * mixer->mixfmt.channels * 5398 mixer->mixfmt.stride / NBBY; 5399 mixer->mixsample = audio_realloc(mixer->mixsample, len); 5400 } 5401 5402 if (reg->codec) { 5403 mixer->codec = reg->codec; 5404 mixer->codecarg.context = reg->context; 5405 if (mode == AUMODE_PLAY) { 5406 mixer->codecarg.srcfmt = &mixer->track_fmt; 5407 mixer->codecarg.dstfmt = &mixer->hwbuf.fmt; 5408 } else { 5409 mixer->codecarg.srcfmt = &mixer->hwbuf.fmt; 5410 mixer->codecarg.dstfmt = &mixer->track_fmt; 5411 } 5412 mixer->codecbuf.fmt = mixer->track_fmt; 5413 mixer->codecbuf.capacity = mixer->frames_per_block; 5414 len = auring_bytelen(&mixer->codecbuf); 5415 mixer->codecbuf.mem = audio_realloc(mixer->codecbuf.mem, len); 5416 } 5417 5418 /* Succeeded so display it. */ 5419 codecbuf[0] = '\0'; 5420 if (mixer->codec || mixer->swap_endian) { 5421 snprintf(codecbuf, sizeof(codecbuf), " %s %s:%d", 5422 (mode == AUMODE_PLAY) ? "->" : "<-", 5423 audio_encoding_name(mixer->hwbuf.fmt.encoding), 5424 mixer->hwbuf.fmt.precision); 5425 } 5426 blkms = mixer->blktime_n * 1000 / mixer->blktime_d; 5427 blkdms = (mixer->blktime_n * 10000 / mixer->blktime_d) % 10; 5428 blkdmsbuf[0] = '\0'; 5429 if (blkdms != 0) { 5430 snprintf(blkdmsbuf, sizeof(blkdmsbuf), ".%1d", blkdms); 5431 } 5432 aprint_normal_dev(sc->sc_dev, 5433 "%s:%d%s %dch %dHz, blk %d bytes (%d%sms) for %s\n", 5434 audio_encoding_name(mixer->track_fmt.encoding), 5435 mixer->track_fmt.precision, 5436 codecbuf, 5437 mixer->track_fmt.channels, 5438 mixer->track_fmt.sample_rate, 5439 blksize, 5440 blkms, blkdmsbuf, 5441 (mode == AUMODE_PLAY) ? "playback" : "recording"); 5442 5443 return 0; 5444 5445 abort: 5446 audio_mixer_destroy(sc, mixer); 5447 return error; 5448 } 5449 5450 /* 5451 * Releases all resources of 'mixer'. 5452 * Note that it does not release the memory area of 'mixer' itself. 5453 * Must be called with sc_exlock held and without sc_lock held. 5454 */ 5455 static void 5456 audio_mixer_destroy(struct audio_softc *sc, audio_trackmixer_t *mixer) 5457 { 5458 int bufsize; 5459 5460 KASSERT(sc->sc_exlock == 1); 5461 5462 bufsize = frametobyte(&mixer->hwbuf.fmt, mixer->hwbuf.capacity); 5463 5464 if (mixer->hwbuf.mem != NULL) { 5465 if (sc->hw_if->freem) { 5466 /* sc_lock is not necessary for freem */ 5467 sc->hw_if->freem(sc->hw_hdl, mixer->hwbuf.mem, bufsize); 5468 } else { 5469 kmem_free(mixer->hwbuf.mem, bufsize); 5470 } 5471 mixer->hwbuf.mem = NULL; 5472 } 5473 5474 audio_free(mixer->codecbuf.mem); 5475 audio_free(mixer->mixsample); 5476 5477 cv_destroy(&mixer->outcv); 5478 5479 if (mixer->sih) { 5480 softint_disestablish(mixer->sih); 5481 mixer->sih = NULL; 5482 } 5483 } 5484 5485 /* 5486 * Starts playback mixer. 5487 * Must be called only if sc_pbusy is false. 5488 * Must be called with sc_lock && sc_exlock held. 5489 * Must not be called from the interrupt context. 5490 */ 5491 static void 5492 audio_pmixer_start(struct audio_softc *sc, bool force) 5493 { 5494 audio_trackmixer_t *mixer; 5495 int minimum; 5496 5497 KASSERT(mutex_owned(sc->sc_lock)); 5498 KASSERT(sc->sc_exlock); 5499 KASSERT(sc->sc_pbusy == false); 5500 5501 mutex_enter(sc->sc_intr_lock); 5502 5503 mixer = sc->sc_pmixer; 5504 TRACE(2, "%smixseq=%d hwseq=%d hwbuf=%d/%d/%d%s", 5505 (audiodebug >= 3) ? "begin " : "", 5506 (int)mixer->mixseq, (int)mixer->hwseq, 5507 mixer->hwbuf.head, mixer->hwbuf.used, mixer->hwbuf.capacity, 5508 force ? " force" : ""); 5509 5510 /* Need two blocks to start normally. */ 5511 minimum = (force) ? 1 : 2; 5512 while (mixer->hwbuf.used < mixer->frames_per_block * minimum) { 5513 audio_pmixer_process(sc); 5514 } 5515 5516 /* Start output */ 5517 audio_pmixer_output(sc); 5518 sc->sc_pbusy = true; 5519 5520 TRACE(3, "end mixseq=%d hwseq=%d hwbuf=%d/%d/%d", 5521 (int)mixer->mixseq, (int)mixer->hwseq, 5522 mixer->hwbuf.head, mixer->hwbuf.used, mixer->hwbuf.capacity); 5523 5524 mutex_exit(sc->sc_intr_lock); 5525 } 5526 5527 /* 5528 * When playing back with MD filter: 5529 * 5530 * track track ... 5531 * v v 5532 * + mix (with aint2_t) 5533 * | master volume (with aint2_t) 5534 * v 5535 * mixsample [::::] wide-int 1 block (ring) buffer 5536 * | 5537 * | convert aint2_t -> aint_t 5538 * v 5539 * codecbuf [....] 1 block (ring) buffer 5540 * | 5541 * | convert to hw format 5542 * v 5543 * hwbuf [............] NBLKHW blocks ring buffer 5544 * 5545 * When playing back without MD filter: 5546 * 5547 * mixsample [::::] wide-int 1 block (ring) buffer 5548 * | 5549 * | convert aint2_t -> aint_t 5550 * | (with byte swap if necessary) 5551 * v 5552 * hwbuf [............] NBLKHW blocks ring buffer 5553 * 5554 * mixsample: slinear_NE, wide internal precision, HW ch, HW freq. 5555 * codecbuf: slinear_NE, internal precision, HW ch, HW freq. 5556 * hwbuf: HW encoding, HW precision, HW ch, HW freq. 5557 */ 5558 5559 /* 5560 * Performs track mixing and converts it to hwbuf. 5561 * Note that this function doesn't transfer hwbuf to hardware. 5562 * Must be called with sc_intr_lock held. 5563 */ 5564 static void 5565 audio_pmixer_process(struct audio_softc *sc) 5566 { 5567 audio_trackmixer_t *mixer; 5568 audio_file_t *f; 5569 int frame_count; 5570 int sample_count; 5571 int mixed; 5572 int i; 5573 aint2_t *m; 5574 aint_t *h; 5575 5576 mixer = sc->sc_pmixer; 5577 5578 frame_count = mixer->frames_per_block; 5579 KASSERTMSG(auring_get_contig_free(&mixer->hwbuf) >= frame_count, 5580 "auring_get_contig_free()=%d frame_count=%d", 5581 auring_get_contig_free(&mixer->hwbuf), frame_count); 5582 sample_count = frame_count * mixer->mixfmt.channels; 5583 5584 mixer->mixseq++; 5585 5586 /* Mix all tracks */ 5587 mixed = 0; 5588 SLIST_FOREACH(f, &sc->sc_files, entry) { 5589 audio_track_t *track = f->ptrack; 5590 5591 if (track == NULL) 5592 continue; 5593 5594 if (track->is_pause) { 5595 TRACET(4, track, "skip; paused"); 5596 continue; 5597 } 5598 5599 /* Skip if the track is used by process context. */ 5600 if (audio_track_lock_tryenter(track) == false) { 5601 TRACET(4, track, "skip; in use"); 5602 continue; 5603 } 5604 5605 /* Emulate mmap'ped track */ 5606 if (track->mmapped) { 5607 auring_push(&track->usrbuf, track->usrbuf_blksize); 5608 TRACET(4, track, "mmap; usr=%d/%d/C%d", 5609 track->usrbuf.head, 5610 track->usrbuf.used, 5611 track->usrbuf.capacity); 5612 } 5613 5614 if (track->outbuf.used < mixer->frames_per_block && 5615 track->usrbuf.used > 0) { 5616 TRACET(4, track, "process"); 5617 audio_track_play(track); 5618 } 5619 5620 if (track->outbuf.used > 0) { 5621 mixed = audio_pmixer_mix_track(mixer, track, mixed); 5622 } else { 5623 TRACET(4, track, "skip; empty"); 5624 } 5625 5626 audio_track_lock_exit(track); 5627 } 5628 5629 if (mixed == 0) { 5630 /* Silence */ 5631 memset(mixer->mixsample, 0, 5632 frametobyte(&mixer->mixfmt, frame_count)); 5633 } else { 5634 if (mixed > 1) { 5635 /* If there are multiple tracks, do auto gain control */ 5636 audio_pmixer_agc(mixer, sample_count); 5637 } 5638 5639 /* Apply master volume */ 5640 if (mixer->volume < 256) { 5641 m = mixer->mixsample; 5642 for (i = 0; i < sample_count; i++) { 5643 *m = AUDIO_SCALEDOWN(*m * mixer->volume, 8); 5644 m++; 5645 } 5646 5647 /* 5648 * Recover the volume gradually at the pace of 5649 * several times per second. If it's too fast, you 5650 * can recognize that the volume changes up and down 5651 * quickly and it's not so comfortable. 5652 */ 5653 mixer->voltimer += mixer->blktime_n; 5654 if (mixer->voltimer * 4 >= mixer->blktime_d) { 5655 mixer->volume++; 5656 mixer->voltimer = 0; 5657 #if defined(AUDIO_DEBUG_AGC) 5658 TRACE(1, "volume recover: %d", mixer->volume); 5659 #endif 5660 } 5661 } 5662 } 5663 5664 /* 5665 * The rest is the hardware part. 5666 */ 5667 5668 m = mixer->mixsample; 5669 5670 if (mixer->codec) { 5671 TRACE(4, "codec count=%d", frame_count); 5672 5673 h = auring_tailptr_aint(&mixer->codecbuf); 5674 for (i=0; i<sample_count; ++i) 5675 *h++ = *m++; 5676 5677 /* Hardware driver's codec */ 5678 auring_push(&mixer->codecbuf, frame_count); 5679 mixer->codecarg.src = auring_headptr(&mixer->codecbuf); 5680 mixer->codecarg.dst = auring_tailptr(&mixer->hwbuf); 5681 mixer->codecarg.count = frame_count; 5682 mixer->codec(&mixer->codecarg); 5683 auring_take(&mixer->codecbuf, mixer->codecarg.count); 5684 } else { 5685 TRACE(4, "direct count=%d", frame_count); 5686 5687 /* Direct conversion to linear output */ 5688 mixer->codecarg.src = m; 5689 mixer->codecarg.dst = auring_tailptr(&mixer->hwbuf); 5690 mixer->codecarg.count = frame_count; 5691 mixer->codecarg.srcfmt = &mixer->mixfmt; 5692 mixer->codecarg.dstfmt = &mixer->hwbuf.fmt; 5693 audio_mixsample_to_linear(&mixer->codecarg); 5694 } 5695 5696 auring_push(&mixer->hwbuf, frame_count); 5697 5698 TRACE(4, "done mixseq=%d hwbuf=%d/%d/%d%s", 5699 (int)mixer->mixseq, 5700 mixer->hwbuf.head, mixer->hwbuf.used, mixer->hwbuf.capacity, 5701 (mixed == 0) ? " silent" : ""); 5702 } 5703 5704 /* 5705 * Do auto gain control. 5706 * Must be called sc_intr_lock held. 5707 */ 5708 static void 5709 audio_pmixer_agc(audio_trackmixer_t *mixer, int sample_count) 5710 { 5711 struct audio_softc *sc __unused; 5712 aint2_t val; 5713 aint2_t maxval; 5714 aint2_t minval; 5715 aint2_t over_plus; 5716 aint2_t over_minus; 5717 aint2_t *m; 5718 int newvol; 5719 int i; 5720 5721 sc = mixer->sc; 5722 5723 /* Overflow detection */ 5724 maxval = AINT_T_MAX; 5725 minval = AINT_T_MIN; 5726 m = mixer->mixsample; 5727 for (i = 0; i < sample_count; i++) { 5728 val = *m++; 5729 if (val > maxval) 5730 maxval = val; 5731 else if (val < minval) 5732 minval = val; 5733 } 5734 5735 /* Absolute value of overflowed amount */ 5736 over_plus = maxval - AINT_T_MAX; 5737 over_minus = AINT_T_MIN - minval; 5738 5739 if (over_plus > 0 || over_minus > 0) { 5740 if (over_plus > over_minus) { 5741 newvol = (int)((aint2_t)AINT_T_MAX * 256 / maxval); 5742 } else { 5743 newvol = (int)((aint2_t)AINT_T_MIN * 256 / minval); 5744 } 5745 5746 /* 5747 * Change the volume only if new one is smaller. 5748 * Reset the timer even if the volume isn't changed. 5749 */ 5750 if (newvol <= mixer->volume) { 5751 mixer->volume = newvol; 5752 mixer->voltimer = 0; 5753 #if defined(AUDIO_DEBUG_AGC) 5754 TRACE(1, "auto volume adjust: %d", mixer->volume); 5755 #endif 5756 } 5757 } 5758 } 5759 5760 /* 5761 * Mix one track. 5762 * 'mixed' specifies the number of tracks mixed so far. 5763 * It returns the number of tracks mixed. In other words, it returns 5764 * mixed + 1 if this track is mixed. 5765 */ 5766 static int 5767 audio_pmixer_mix_track(audio_trackmixer_t *mixer, audio_track_t *track, 5768 int mixed) 5769 { 5770 int count; 5771 int sample_count; 5772 int remain; 5773 int i; 5774 const aint_t *s; 5775 aint2_t *d; 5776 5777 /* XXX TODO: Is this necessary for now? */ 5778 if (mixer->mixseq < track->seq) 5779 return mixed; 5780 5781 count = auring_get_contig_used(&track->outbuf); 5782 count = uimin(count, mixer->frames_per_block); 5783 5784 s = auring_headptr_aint(&track->outbuf); 5785 d = mixer->mixsample; 5786 5787 /* 5788 * Apply track volume with double-sized integer and perform 5789 * additive synthesis. 5790 * 5791 * XXX If you limit the track volume to 1.0 or less (<= 256), 5792 * it would be better to do this in the track conversion stage 5793 * rather than here. However, if you accept the volume to 5794 * be greater than 1.0 (> 256), it's better to do it here. 5795 * Because the operation here is done by double-sized integer. 5796 */ 5797 sample_count = count * mixer->mixfmt.channels; 5798 if (mixed == 0) { 5799 /* If this is the first track, assignment can be used. */ 5800 #if defined(AUDIO_SUPPORT_TRACK_VOLUME) 5801 if (track->volume != 256) { 5802 for (i = 0; i < sample_count; i++) { 5803 aint2_t v; 5804 v = *s++; 5805 *d++ = AUDIO_SCALEDOWN(v * track->volume, 8) 5806 } 5807 } else 5808 #endif 5809 { 5810 for (i = 0; i < sample_count; i++) { 5811 *d++ = ((aint2_t)*s++); 5812 } 5813 } 5814 /* Fill silence if the first track is not filled. */ 5815 for (; i < mixer->frames_per_block * mixer->mixfmt.channels; i++) 5816 *d++ = 0; 5817 } else { 5818 /* If this is the second or later, add it. */ 5819 #if defined(AUDIO_SUPPORT_TRACK_VOLUME) 5820 if (track->volume != 256) { 5821 for (i = 0; i < sample_count; i++) { 5822 aint2_t v; 5823 v = *s++; 5824 *d++ += AUDIO_SCALEDOWN(v * track->volume, 8); 5825 } 5826 } else 5827 #endif 5828 { 5829 for (i = 0; i < sample_count; i++) { 5830 *d++ += ((aint2_t)*s++); 5831 } 5832 } 5833 } 5834 5835 auring_take(&track->outbuf, count); 5836 /* 5837 * The counters have to align block even if outbuf is less than 5838 * one block. XXX Is this still necessary? 5839 */ 5840 remain = mixer->frames_per_block - count; 5841 if (__predict_false(remain != 0)) { 5842 auring_push(&track->outbuf, remain); 5843 auring_take(&track->outbuf, remain); 5844 } 5845 5846 /* 5847 * Update track sequence. 5848 * mixseq has previous value yet at this point. 5849 */ 5850 track->seq = mixer->mixseq + 1; 5851 5852 return mixed + 1; 5853 } 5854 5855 /* 5856 * Output one block from hwbuf to HW. 5857 * Must be called with sc_intr_lock held. 5858 */ 5859 static void 5860 audio_pmixer_output(struct audio_softc *sc) 5861 { 5862 audio_trackmixer_t *mixer; 5863 audio_params_t params; 5864 void *start; 5865 void *end; 5866 int blksize; 5867 int error; 5868 5869 mixer = sc->sc_pmixer; 5870 TRACE(4, "pbusy=%d hwbuf=%d/%d/%d", 5871 sc->sc_pbusy, 5872 mixer->hwbuf.head, mixer->hwbuf.used, mixer->hwbuf.capacity); 5873 KASSERTMSG(mixer->hwbuf.used >= mixer->frames_per_block, 5874 "mixer->hwbuf.used=%d mixer->frames_per_block=%d", 5875 mixer->hwbuf.used, mixer->frames_per_block); 5876 5877 blksize = frametobyte(&mixer->hwbuf.fmt, mixer->frames_per_block); 5878 5879 if (sc->hw_if->trigger_output) { 5880 /* trigger (at once) */ 5881 if (!sc->sc_pbusy) { 5882 start = mixer->hwbuf.mem; 5883 end = (uint8_t *)start + auring_bytelen(&mixer->hwbuf); 5884 params = format2_to_params(&mixer->hwbuf.fmt); 5885 5886 error = sc->hw_if->trigger_output(sc->hw_hdl, 5887 start, end, blksize, audio_pintr, sc, ¶ms); 5888 if (error) { 5889 audio_printf(sc, 5890 "trigger_output failed: errno=%d\n", 5891 error); 5892 return; 5893 } 5894 } 5895 } else { 5896 /* start (everytime) */ 5897 start = auring_headptr(&mixer->hwbuf); 5898 5899 error = sc->hw_if->start_output(sc->hw_hdl, 5900 start, blksize, audio_pintr, sc); 5901 if (error) { 5902 audio_printf(sc, 5903 "start_output failed: errno=%d\n", error); 5904 return; 5905 } 5906 } 5907 } 5908 5909 /* 5910 * This is an interrupt handler for playback. 5911 * It is called with sc_intr_lock held. 5912 * 5913 * It is usually called from hardware interrupt. However, note that 5914 * for some drivers (e.g. uaudio) it is called from software interrupt. 5915 */ 5916 static void 5917 audio_pintr(void *arg) 5918 { 5919 struct audio_softc *sc; 5920 audio_trackmixer_t *mixer; 5921 5922 sc = arg; 5923 KASSERT(mutex_owned(sc->sc_intr_lock)); 5924 5925 if (sc->sc_dying) 5926 return; 5927 if (sc->sc_pbusy == false) { 5928 #if defined(DIAGNOSTIC) 5929 audio_printf(sc, "DIAGNOSTIC: %s raised stray interrupt\n", 5930 device_xname(sc->hw_dev)); 5931 #endif 5932 return; 5933 } 5934 5935 mixer = sc->sc_pmixer; 5936 mixer->hw_complete_counter += mixer->frames_per_block; 5937 mixer->hwseq++; 5938 5939 auring_take(&mixer->hwbuf, mixer->frames_per_block); 5940 5941 TRACE(4, 5942 "HW_INT ++hwseq=%" PRIu64 " cmplcnt=%" PRIu64 " hwbuf=%d/%d/%d", 5943 mixer->hwseq, mixer->hw_complete_counter, 5944 mixer->hwbuf.head, mixer->hwbuf.used, mixer->hwbuf.capacity); 5945 5946 #if defined(AUDIO_HW_SINGLE_BUFFER) 5947 /* 5948 * Create a new block here and output it immediately. 5949 * It makes a latency lower but needs machine power. 5950 */ 5951 audio_pmixer_process(sc); 5952 audio_pmixer_output(sc); 5953 #else 5954 /* 5955 * It is called when block N output is done. 5956 * Output immediately block N+1 created by the last interrupt. 5957 * And then create block N+2 for the next interrupt. 5958 * This method makes playback robust even on slower machines. 5959 * Instead the latency is increased by one block. 5960 */ 5961 5962 /* At first, output ready block. */ 5963 if (mixer->hwbuf.used >= mixer->frames_per_block) { 5964 audio_pmixer_output(sc); 5965 } 5966 5967 bool later = false; 5968 5969 if (mixer->hwbuf.used < mixer->frames_per_block) { 5970 later = true; 5971 } 5972 5973 /* Then, process next block. */ 5974 audio_pmixer_process(sc); 5975 5976 if (later) { 5977 audio_pmixer_output(sc); 5978 } 5979 #endif 5980 5981 /* 5982 * When this interrupt is the real hardware interrupt, disabling 5983 * preemption here is not necessary. But some drivers (e.g. uaudio) 5984 * emulate it by software interrupt, so kpreempt_disable is necessary. 5985 */ 5986 kpreempt_disable(); 5987 softint_schedule(mixer->sih); 5988 kpreempt_enable(); 5989 } 5990 5991 /* 5992 * Starts record mixer. 5993 * Must be called only if sc_rbusy is false. 5994 * Must be called with sc_lock && sc_exlock held. 5995 * Must not be called from the interrupt context. 5996 */ 5997 static void 5998 audio_rmixer_start(struct audio_softc *sc) 5999 { 6000 6001 KASSERT(mutex_owned(sc->sc_lock)); 6002 KASSERT(sc->sc_exlock); 6003 KASSERT(sc->sc_rbusy == false); 6004 6005 mutex_enter(sc->sc_intr_lock); 6006 6007 TRACE(2, "%s", (audiodebug >= 3) ? "begin" : ""); 6008 audio_rmixer_input(sc); 6009 sc->sc_rbusy = true; 6010 TRACE(3, "end"); 6011 6012 mutex_exit(sc->sc_intr_lock); 6013 } 6014 6015 /* 6016 * When recording with MD filter: 6017 * 6018 * hwbuf [............] NBLKHW blocks ring buffer 6019 * | 6020 * | convert from hw format 6021 * v 6022 * codecbuf [....] 1 block (ring) buffer 6023 * | | 6024 * v v 6025 * track track ... 6026 * 6027 * When recording without MD filter: 6028 * 6029 * hwbuf [............] NBLKHW blocks ring buffer 6030 * | | 6031 * v v 6032 * track track ... 6033 * 6034 * hwbuf: HW encoding, HW precision, HW ch, HW freq. 6035 * codecbuf: slinear_NE, internal precision, HW ch, HW freq. 6036 */ 6037 6038 /* 6039 * Distribute a recorded block to all recording tracks. 6040 */ 6041 static void 6042 audio_rmixer_process(struct audio_softc *sc) 6043 { 6044 audio_trackmixer_t *mixer; 6045 audio_ring_t *mixersrc; 6046 audio_ring_t tmpsrc; 6047 audio_filter_t codec; 6048 audio_filter_arg_t codecarg; 6049 audio_file_t *f; 6050 int count; 6051 int bytes; 6052 6053 mixer = sc->sc_rmixer; 6054 6055 /* 6056 * count is the number of frames to be retrieved this time. 6057 * count should be one block. 6058 */ 6059 count = auring_get_contig_used(&mixer->hwbuf); 6060 count = uimin(count, mixer->frames_per_block); 6061 if (count <= 0) { 6062 TRACE(4, "count %d: too short", count); 6063 return; 6064 } 6065 bytes = frametobyte(&mixer->track_fmt, count); 6066 6067 /* Hardware driver's codec */ 6068 if (mixer->codec) { 6069 TRACE(4, "codec count=%d", count); 6070 mixer->codecarg.src = auring_headptr(&mixer->hwbuf); 6071 mixer->codecarg.dst = auring_tailptr(&mixer->codecbuf); 6072 mixer->codecarg.count = count; 6073 mixer->codec(&mixer->codecarg); 6074 mixersrc = &mixer->codecbuf; 6075 } else { 6076 TRACE(4, "direct count=%d", count); 6077 /* temporary ring using mixsample buffer */ 6078 tmpsrc.fmt = mixer->mixfmt; 6079 tmpsrc.capacity = mixer->frames_per_block; 6080 tmpsrc.mem = mixer->mixsample; 6081 tmpsrc.head = 0; 6082 tmpsrc.used = 0; 6083 6084 /* ad-hoc codec */ 6085 codecarg.srcfmt = &mixer->hwbuf.fmt; 6086 codecarg.dstfmt = &mixer->mixfmt; 6087 codec = NULL; 6088 if (audio_format2_is_linear(codecarg.srcfmt)) { 6089 switch (codecarg.srcfmt->stride) { 6090 case 8: 6091 codec = audio_linear8_to_internal; 6092 break; 6093 case 16: 6094 codec = audio_linear16_to_internal; 6095 break; 6096 #if defined(AUDIO_SUPPORT_LINEAR24) 6097 case 24: 6098 codec = audio_linear24_to_internal; 6099 break; 6100 #endif 6101 case 32: 6102 codec = audio_linear32_to_internal; 6103 break; 6104 } 6105 } 6106 if (codec == NULL) { 6107 TRACE(4, "unsupported hw format"); 6108 return; 6109 } 6110 6111 codecarg.src = auring_headptr(&mixer->hwbuf); 6112 codecarg.dst = auring_tailptr(&tmpsrc); 6113 codecarg.count = count; 6114 codec(&codecarg); 6115 mixersrc = &tmpsrc; 6116 } 6117 6118 auring_take(&mixer->hwbuf, count); 6119 auring_push(mixersrc, count); 6120 6121 TRACE(4, "distribute"); 6122 6123 /* Distribute to all tracks. */ 6124 SLIST_FOREACH(f, &sc->sc_files, entry) { 6125 audio_track_t *track = f->rtrack; 6126 audio_ring_t *input; 6127 6128 if (track == NULL) 6129 continue; 6130 6131 if (track->is_pause) { 6132 TRACET(4, track, "skip; paused"); 6133 continue; 6134 } 6135 6136 if (audio_track_lock_tryenter(track) == false) { 6137 TRACET(4, track, "skip; in use"); 6138 continue; 6139 } 6140 6141 /* 6142 * If the track buffer has less than one block of free space, 6143 * make one block free. 6144 */ 6145 input = track->input; 6146 if (input->capacity - input->used < mixer->frames_per_block) { 6147 int drops = mixer->frames_per_block - 6148 (input->capacity - input->used); 6149 track->dropframes += drops; 6150 TRACET(4, track, "drop %d frames: inp=%d/%d/%d", 6151 drops, 6152 input->head, input->used, input->capacity); 6153 auring_take(input, drops); 6154 } 6155 6156 KASSERTMSG(auring_tail(input) % mixer->frames_per_block == 0, 6157 "inputtail=%d mixer->frames_per_block=%d", 6158 auring_tail(input), mixer->frames_per_block); 6159 memcpy(auring_tailptr_aint(input), 6160 auring_headptr_aint(mixersrc), 6161 bytes); 6162 auring_push(input, count); 6163 6164 track->stamp++; 6165 6166 audio_track_lock_exit(track); 6167 } 6168 6169 auring_take(mixersrc, count); 6170 } 6171 6172 /* 6173 * Input one block from HW to hwbuf. 6174 * Must be called with sc_intr_lock held. 6175 */ 6176 static void 6177 audio_rmixer_input(struct audio_softc *sc) 6178 { 6179 audio_trackmixer_t *mixer; 6180 audio_params_t params; 6181 void *start; 6182 void *end; 6183 int blksize; 6184 int error; 6185 6186 mixer = sc->sc_rmixer; 6187 blksize = frametobyte(&mixer->hwbuf.fmt, mixer->frames_per_block); 6188 6189 if (sc->hw_if->trigger_input) { 6190 /* trigger (at once) */ 6191 if (!sc->sc_rbusy) { 6192 start = mixer->hwbuf.mem; 6193 end = (uint8_t *)start + auring_bytelen(&mixer->hwbuf); 6194 params = format2_to_params(&mixer->hwbuf.fmt); 6195 6196 error = sc->hw_if->trigger_input(sc->hw_hdl, 6197 start, end, blksize, audio_rintr, sc, ¶ms); 6198 if (error) { 6199 audio_printf(sc, 6200 "trigger_input failed: errno=%d\n", 6201 error); 6202 return; 6203 } 6204 } 6205 } else { 6206 /* start (everytime) */ 6207 start = auring_tailptr(&mixer->hwbuf); 6208 6209 error = sc->hw_if->start_input(sc->hw_hdl, 6210 start, blksize, audio_rintr, sc); 6211 if (error) { 6212 audio_printf(sc, 6213 "start_input failed: errno=%d\n", error); 6214 return; 6215 } 6216 } 6217 } 6218 6219 /* 6220 * This is an interrupt handler for recording. 6221 * It is called with sc_intr_lock. 6222 * 6223 * It is usually called from hardware interrupt. However, note that 6224 * for some drivers (e.g. uaudio) it is called from software interrupt. 6225 */ 6226 static void 6227 audio_rintr(void *arg) 6228 { 6229 struct audio_softc *sc; 6230 audio_trackmixer_t *mixer; 6231 6232 sc = arg; 6233 KASSERT(mutex_owned(sc->sc_intr_lock)); 6234 6235 if (sc->sc_dying) 6236 return; 6237 if (sc->sc_rbusy == false) { 6238 #if defined(DIAGNOSTIC) 6239 audio_printf(sc, "DIAGNOSTIC: %s raised stray interrupt\n", 6240 device_xname(sc->hw_dev)); 6241 #endif 6242 return; 6243 } 6244 6245 mixer = sc->sc_rmixer; 6246 mixer->hw_complete_counter += mixer->frames_per_block; 6247 mixer->hwseq++; 6248 6249 auring_push(&mixer->hwbuf, mixer->frames_per_block); 6250 6251 TRACE(4, 6252 "HW_INT ++hwseq=%" PRIu64 " cmplcnt=%" PRIu64 " hwbuf=%d/%d/%d", 6253 mixer->hwseq, mixer->hw_complete_counter, 6254 mixer->hwbuf.head, mixer->hwbuf.used, mixer->hwbuf.capacity); 6255 6256 /* Distrubute recorded block */ 6257 audio_rmixer_process(sc); 6258 6259 /* Request next block */ 6260 audio_rmixer_input(sc); 6261 6262 /* 6263 * When this interrupt is the real hardware interrupt, disabling 6264 * preemption here is not necessary. But some drivers (e.g. uaudio) 6265 * emulate it by software interrupt, so kpreempt_disable is necessary. 6266 */ 6267 kpreempt_disable(); 6268 softint_schedule(mixer->sih); 6269 kpreempt_enable(); 6270 } 6271 6272 /* 6273 * Halts playback mixer. 6274 * This function also clears related parameters, so call this function 6275 * instead of calling halt_output directly. 6276 * Must be called only if sc_pbusy is true. 6277 * Must be called with sc_lock && sc_exlock held. 6278 */ 6279 static int 6280 audio_pmixer_halt(struct audio_softc *sc) 6281 { 6282 int error; 6283 6284 TRACE(2, "called"); 6285 KASSERT(mutex_owned(sc->sc_lock)); 6286 KASSERT(sc->sc_exlock); 6287 6288 mutex_enter(sc->sc_intr_lock); 6289 error = sc->hw_if->halt_output(sc->hw_hdl); 6290 6291 /* Halts anyway even if some error has occurred. */ 6292 sc->sc_pbusy = false; 6293 sc->sc_pmixer->hwbuf.head = 0; 6294 sc->sc_pmixer->hwbuf.used = 0; 6295 sc->sc_pmixer->mixseq = 0; 6296 sc->sc_pmixer->hwseq = 0; 6297 mutex_exit(sc->sc_intr_lock); 6298 6299 return error; 6300 } 6301 6302 /* 6303 * Halts recording mixer. 6304 * This function also clears related parameters, so call this function 6305 * instead of calling halt_input directly. 6306 * Must be called only if sc_rbusy is true. 6307 * Must be called with sc_lock && sc_exlock held. 6308 */ 6309 static int 6310 audio_rmixer_halt(struct audio_softc *sc) 6311 { 6312 int error; 6313 6314 TRACE(2, "called"); 6315 KASSERT(mutex_owned(sc->sc_lock)); 6316 KASSERT(sc->sc_exlock); 6317 6318 mutex_enter(sc->sc_intr_lock); 6319 error = sc->hw_if->halt_input(sc->hw_hdl); 6320 6321 /* Halts anyway even if some error has occurred. */ 6322 sc->sc_rbusy = false; 6323 sc->sc_rmixer->hwbuf.head = 0; 6324 sc->sc_rmixer->hwbuf.used = 0; 6325 sc->sc_rmixer->mixseq = 0; 6326 sc->sc_rmixer->hwseq = 0; 6327 mutex_exit(sc->sc_intr_lock); 6328 6329 return error; 6330 } 6331 6332 /* 6333 * Flush this track. 6334 * Halts all operations, clears all buffers, reset error counters. 6335 * XXX I'm not sure... 6336 */ 6337 static void 6338 audio_track_clear(struct audio_softc *sc, audio_track_t *track) 6339 { 6340 6341 KASSERT(track); 6342 TRACET(3, track, "clear"); 6343 6344 audio_track_lock_enter(track); 6345 6346 /* Clear all internal parameters. */ 6347 track->usrbuf.used = 0; 6348 track->usrbuf.head = 0; 6349 if (track->codec.filter) { 6350 track->codec.srcbuf.used = 0; 6351 track->codec.srcbuf.head = 0; 6352 } 6353 if (track->chvol.filter) { 6354 track->chvol.srcbuf.used = 0; 6355 track->chvol.srcbuf.head = 0; 6356 } 6357 if (track->chmix.filter) { 6358 track->chmix.srcbuf.used = 0; 6359 track->chmix.srcbuf.head = 0; 6360 } 6361 if (track->freq.filter) { 6362 track->freq.srcbuf.used = 0; 6363 track->freq.srcbuf.head = 0; 6364 if (track->freq_step < 65536) 6365 track->freq_current = 65536; 6366 else 6367 track->freq_current = 0; 6368 memset(track->freq_prev, 0, sizeof(track->freq_prev)); 6369 memset(track->freq_curr, 0, sizeof(track->freq_curr)); 6370 } 6371 /* Clear buffer, then operation halts naturally. */ 6372 track->outbuf.used = 0; 6373 6374 /* Clear counters. */ 6375 track->stamp = 0; 6376 track->last_stamp = 0; 6377 track->dropframes = 0; 6378 6379 audio_track_lock_exit(track); 6380 } 6381 6382 /* 6383 * Drain the track. 6384 * track must be present and for playback. 6385 * If successful, it returns 0. Otherwise returns errno. 6386 * Must be called with sc_lock held. 6387 */ 6388 static int 6389 audio_track_drain(struct audio_softc *sc, audio_track_t *track) 6390 { 6391 audio_trackmixer_t *mixer; 6392 int done; 6393 int error; 6394 6395 KASSERT(track); 6396 TRACET(3, track, "start"); 6397 mixer = track->mixer; 6398 KASSERT(mutex_owned(sc->sc_lock)); 6399 6400 /* Ignore them if pause. */ 6401 if (track->is_pause) { 6402 TRACET(3, track, "pause -> clear"); 6403 track->pstate = AUDIO_STATE_CLEAR; 6404 } 6405 /* Terminate early here if there is no data in the track. */ 6406 if (track->pstate == AUDIO_STATE_CLEAR) { 6407 TRACET(3, track, "no need to drain"); 6408 return 0; 6409 } 6410 track->pstate = AUDIO_STATE_DRAINING; 6411 6412 for (;;) { 6413 /* I want to display it before condition evaluation. */ 6414 TRACET(3, track, "pid=%d.%d trkseq=%d hwseq=%d out=%d/%d/%d", 6415 (int)curproc->p_pid, (int)curlwp->l_lid, 6416 (int)track->seq, (int)mixer->hwseq, 6417 track->outbuf.head, track->outbuf.used, 6418 track->outbuf.capacity); 6419 6420 /* Condition to terminate */ 6421 audio_track_lock_enter(track); 6422 done = (track->usrbuf.used < frametobyte(&track->inputfmt, 1) && 6423 track->outbuf.used == 0 && 6424 track->seq <= mixer->hwseq); 6425 audio_track_lock_exit(track); 6426 if (done) 6427 break; 6428 6429 TRACET(3, track, "sleep"); 6430 error = audio_track_waitio(sc, track, "audio_drain"); 6431 if (error) 6432 return error; 6433 6434 /* XXX call audio_track_play here ? */ 6435 } 6436 6437 track->pstate = AUDIO_STATE_CLEAR; 6438 TRACET(3, track, "done"); 6439 return 0; 6440 } 6441 6442 /* 6443 * Send signal to process. 6444 * This is intended to be called only from audio_softintr_{rd,wr}. 6445 * Must be called without sc_intr_lock held. 6446 */ 6447 static inline void 6448 audio_psignal(struct audio_softc *sc, pid_t pid, int signum) 6449 { 6450 proc_t *p; 6451 6452 KASSERT(pid != 0); 6453 6454 /* 6455 * psignal() must be called without spin lock held. 6456 */ 6457 6458 mutex_enter(&proc_lock); 6459 p = proc_find(pid); 6460 if (p) 6461 psignal(p, signum); 6462 mutex_exit(&proc_lock); 6463 } 6464 6465 /* 6466 * This is software interrupt handler for record. 6467 * It is called from recording hardware interrupt everytime. 6468 * It does: 6469 * - Deliver SIGIO for all async processes. 6470 * - Notify to audio_read() that data has arrived. 6471 * - selnotify() for select/poll-ing processes. 6472 */ 6473 /* 6474 * XXX If a process issues FIOASYNC between hardware interrupt and 6475 * software interrupt, (stray) SIGIO will be sent to the process 6476 * despite the fact that it has not receive recorded data yet. 6477 */ 6478 static void 6479 audio_softintr_rd(void *cookie) 6480 { 6481 struct audio_softc *sc = cookie; 6482 audio_file_t *f; 6483 pid_t pid; 6484 6485 mutex_enter(sc->sc_lock); 6486 6487 SLIST_FOREACH(f, &sc->sc_files, entry) { 6488 audio_track_t *track = f->rtrack; 6489 6490 if (track == NULL) 6491 continue; 6492 6493 TRACET(4, track, "broadcast; inp=%d/%d/%d", 6494 track->input->head, 6495 track->input->used, 6496 track->input->capacity); 6497 6498 pid = f->async_audio; 6499 if (pid != 0) { 6500 TRACEF(4, f, "sending SIGIO %d", pid); 6501 audio_psignal(sc, pid, SIGIO); 6502 } 6503 } 6504 6505 /* Notify that data has arrived. */ 6506 selnotify(&sc->sc_rsel, 0, NOTE_SUBMIT); 6507 cv_broadcast(&sc->sc_rmixer->outcv); 6508 6509 mutex_exit(sc->sc_lock); 6510 } 6511 6512 /* 6513 * This is software interrupt handler for playback. 6514 * It is called from playback hardware interrupt everytime. 6515 * It does: 6516 * - Deliver SIGIO for all async and writable (used < lowat) processes. 6517 * - Notify to audio_write() that outbuf block available. 6518 * - selnotify() for select/poll-ing processes if there are any writable 6519 * (used < lowat) processes. Checking each descriptor will be done by 6520 * filt_audiowrite_event(). 6521 */ 6522 static void 6523 audio_softintr_wr(void *cookie) 6524 { 6525 struct audio_softc *sc = cookie; 6526 audio_file_t *f; 6527 bool found; 6528 pid_t pid; 6529 6530 TRACE(4, "called"); 6531 found = false; 6532 6533 mutex_enter(sc->sc_lock); 6534 6535 SLIST_FOREACH(f, &sc->sc_files, entry) { 6536 audio_track_t *track = f->ptrack; 6537 6538 if (track == NULL) 6539 continue; 6540 6541 TRACET(4, track, "broadcast; trkseq=%d out=%d/%d/%d", 6542 (int)track->seq, 6543 track->outbuf.head, 6544 track->outbuf.used, 6545 track->outbuf.capacity); 6546 6547 /* 6548 * Send a signal if the process is async mode and 6549 * used is lower than lowat. 6550 */ 6551 if (track->usrbuf.used <= track->usrbuf_usedlow && 6552 !track->is_pause) { 6553 /* For selnotify */ 6554 found = true; 6555 /* For SIGIO */ 6556 pid = f->async_audio; 6557 if (pid != 0) { 6558 TRACEF(4, f, "sending SIGIO %d", pid); 6559 audio_psignal(sc, pid, SIGIO); 6560 } 6561 } 6562 } 6563 6564 /* 6565 * Notify for select/poll when someone become writable. 6566 * It needs sc_lock (and not sc_intr_lock). 6567 */ 6568 if (found) { 6569 TRACE(4, "selnotify"); 6570 selnotify(&sc->sc_wsel, 0, NOTE_SUBMIT); 6571 } 6572 6573 /* Notify to audio_write() that outbuf available. */ 6574 cv_broadcast(&sc->sc_pmixer->outcv); 6575 6576 mutex_exit(sc->sc_lock); 6577 } 6578 6579 /* 6580 * Check (and convert) the format *p came from userland. 6581 * If successful, it writes back the converted format to *p if necessary and 6582 * returns 0. Otherwise returns errno (*p may be changed even in this case). 6583 */ 6584 static int 6585 audio_check_params(audio_format2_t *p) 6586 { 6587 6588 /* 6589 * Convert obsolete AUDIO_ENCODING_PCM encodings. 6590 * 6591 * AUDIO_ENCODING_PCM16 == AUDIO_ENCODING_LINEAR 6592 * So, it's always signed, as in SunOS. 6593 * 6594 * AUDIO_ENCODING_PCM8 == AUDIO_ENCODING_LINEAR8 6595 * So, it's always unsigned, as in SunOS. 6596 */ 6597 if (p->encoding == AUDIO_ENCODING_PCM16) { 6598 p->encoding = AUDIO_ENCODING_SLINEAR; 6599 } else if (p->encoding == AUDIO_ENCODING_PCM8) { 6600 if (p->precision == 8) 6601 p->encoding = AUDIO_ENCODING_ULINEAR; 6602 else 6603 return EINVAL; 6604 } 6605 6606 /* 6607 * Convert obsoleted AUDIO_ENCODING_[SU]LINEAR without endianness 6608 * suffix. 6609 */ 6610 if (p->encoding == AUDIO_ENCODING_SLINEAR) 6611 p->encoding = AUDIO_ENCODING_SLINEAR_NE; 6612 if (p->encoding == AUDIO_ENCODING_ULINEAR) 6613 p->encoding = AUDIO_ENCODING_ULINEAR_NE; 6614 6615 switch (p->encoding) { 6616 case AUDIO_ENCODING_ULAW: 6617 case AUDIO_ENCODING_ALAW: 6618 if (p->precision != 8) 6619 return EINVAL; 6620 break; 6621 case AUDIO_ENCODING_ADPCM: 6622 if (p->precision != 4 && p->precision != 8) 6623 return EINVAL; 6624 break; 6625 case AUDIO_ENCODING_SLINEAR_LE: 6626 case AUDIO_ENCODING_SLINEAR_BE: 6627 case AUDIO_ENCODING_ULINEAR_LE: 6628 case AUDIO_ENCODING_ULINEAR_BE: 6629 if (p->precision != 8 && p->precision != 16 && 6630 p->precision != 24 && p->precision != 32) 6631 return EINVAL; 6632 6633 /* 8bit format does not have endianness. */ 6634 if (p->precision == 8) { 6635 if (p->encoding == AUDIO_ENCODING_SLINEAR_OE) 6636 p->encoding = AUDIO_ENCODING_SLINEAR_NE; 6637 if (p->encoding == AUDIO_ENCODING_ULINEAR_OE) 6638 p->encoding = AUDIO_ENCODING_ULINEAR_NE; 6639 } 6640 6641 if (p->precision > p->stride) 6642 return EINVAL; 6643 break; 6644 case AUDIO_ENCODING_MPEG_L1_STREAM: 6645 case AUDIO_ENCODING_MPEG_L1_PACKETS: 6646 case AUDIO_ENCODING_MPEG_L1_SYSTEM: 6647 case AUDIO_ENCODING_MPEG_L2_STREAM: 6648 case AUDIO_ENCODING_MPEG_L2_PACKETS: 6649 case AUDIO_ENCODING_MPEG_L2_SYSTEM: 6650 case AUDIO_ENCODING_AC3: 6651 break; 6652 default: 6653 return EINVAL; 6654 } 6655 6656 /* sanity check # of channels*/ 6657 if (p->channels < 1 || p->channels > AUDIO_MAX_CHANNELS) 6658 return EINVAL; 6659 6660 return 0; 6661 } 6662 6663 /* 6664 * Initialize playback and record mixers. 6665 * mode (AUMODE_{PLAY,RECORD}) indicates the mixer to be initialized. 6666 * phwfmt and rhwfmt indicate the hardware format. pfil and rfil indicate 6667 * the filter registration information. These four must not be NULL. 6668 * If successful returns 0. Otherwise returns errno. 6669 * Must be called with sc_exlock held and without sc_lock held. 6670 * Must not be called if there are any tracks. 6671 * Caller should check that the initialization succeed by whether 6672 * sc_[pr]mixer is not NULL. 6673 */ 6674 static int 6675 audio_mixers_init(struct audio_softc *sc, int mode, 6676 const audio_format2_t *phwfmt, const audio_format2_t *rhwfmt, 6677 const audio_filter_reg_t *pfil, const audio_filter_reg_t *rfil) 6678 { 6679 int error; 6680 6681 KASSERT(phwfmt != NULL); 6682 KASSERT(rhwfmt != NULL); 6683 KASSERT(pfil != NULL); 6684 KASSERT(rfil != NULL); 6685 KASSERT(sc->sc_exlock); 6686 6687 if ((mode & AUMODE_PLAY)) { 6688 if (sc->sc_pmixer == NULL) { 6689 sc->sc_pmixer = kmem_zalloc(sizeof(*sc->sc_pmixer), 6690 KM_SLEEP); 6691 } else { 6692 /* destroy() doesn't free memory. */ 6693 audio_mixer_destroy(sc, sc->sc_pmixer); 6694 memset(sc->sc_pmixer, 0, sizeof(*sc->sc_pmixer)); 6695 } 6696 error = audio_mixer_init(sc, AUMODE_PLAY, phwfmt, pfil); 6697 if (error) { 6698 /* audio_mixer_init already displayed error code */ 6699 audio_printf(sc, "configuring playback mode failed\n"); 6700 kmem_free(sc->sc_pmixer, sizeof(*sc->sc_pmixer)); 6701 sc->sc_pmixer = NULL; 6702 return error; 6703 } 6704 } 6705 if ((mode & AUMODE_RECORD)) { 6706 if (sc->sc_rmixer == NULL) { 6707 sc->sc_rmixer = kmem_zalloc(sizeof(*sc->sc_rmixer), 6708 KM_SLEEP); 6709 } else { 6710 /* destroy() doesn't free memory. */ 6711 audio_mixer_destroy(sc, sc->sc_rmixer); 6712 memset(sc->sc_rmixer, 0, sizeof(*sc->sc_rmixer)); 6713 } 6714 error = audio_mixer_init(sc, AUMODE_RECORD, rhwfmt, rfil); 6715 if (error) { 6716 /* audio_mixer_init already displayed error code */ 6717 audio_printf(sc, "configuring record mode failed\n"); 6718 kmem_free(sc->sc_rmixer, sizeof(*sc->sc_rmixer)); 6719 sc->sc_rmixer = NULL; 6720 return error; 6721 } 6722 } 6723 6724 return 0; 6725 } 6726 6727 /* 6728 * Select a frequency. 6729 * Prioritize 48kHz and 44.1kHz. Otherwise choose the highest one. 6730 * XXX Better algorithm? 6731 */ 6732 static int 6733 audio_select_freq(const struct audio_format *fmt) 6734 { 6735 int freq; 6736 int high; 6737 int low; 6738 int j; 6739 6740 if (fmt->frequency_type == 0) { 6741 low = fmt->frequency[0]; 6742 high = fmt->frequency[1]; 6743 freq = 48000; 6744 if (low <= freq && freq <= high) { 6745 return freq; 6746 } 6747 freq = 44100; 6748 if (low <= freq && freq <= high) { 6749 return freq; 6750 } 6751 return high; 6752 } else { 6753 for (j = 0; j < fmt->frequency_type; j++) { 6754 if (fmt->frequency[j] == 48000) { 6755 return fmt->frequency[j]; 6756 } 6757 } 6758 high = 0; 6759 for (j = 0; j < fmt->frequency_type; j++) { 6760 if (fmt->frequency[j] == 44100) { 6761 return fmt->frequency[j]; 6762 } 6763 if (fmt->frequency[j] > high) { 6764 high = fmt->frequency[j]; 6765 } 6766 } 6767 return high; 6768 } 6769 } 6770 6771 /* 6772 * Choose the most preferred hardware format. 6773 * If successful, it will store the chosen format into *cand and return 0. 6774 * Otherwise, return errno. 6775 * Must be called without sc_lock held. 6776 */ 6777 static int 6778 audio_hw_probe(struct audio_softc *sc, audio_format2_t *cand, int mode) 6779 { 6780 audio_format_query_t query; 6781 int cand_score; 6782 int score; 6783 int i; 6784 int error; 6785 6786 /* 6787 * Score each formats and choose the highest one. 6788 * 6789 * +---- priority(0-3) 6790 * |+--- encoding/precision 6791 * ||+-- channels 6792 * score = 0x000000PEC 6793 */ 6794 6795 cand_score = 0; 6796 for (i = 0; ; i++) { 6797 memset(&query, 0, sizeof(query)); 6798 query.index = i; 6799 6800 mutex_enter(sc->sc_lock); 6801 error = sc->hw_if->query_format(sc->hw_hdl, &query); 6802 mutex_exit(sc->sc_lock); 6803 if (error == EINVAL) 6804 break; 6805 if (error) 6806 return error; 6807 6808 #if defined(AUDIO_DEBUG) 6809 DPRINTF(1, "fmt[%d] %c%c pri=%d %s,%d/%dbit,%dch,", i, 6810 (query.fmt.mode & AUMODE_PLAY) ? 'P' : '-', 6811 (query.fmt.mode & AUMODE_RECORD) ? 'R' : '-', 6812 query.fmt.priority, 6813 audio_encoding_name(query.fmt.encoding), 6814 query.fmt.validbits, 6815 query.fmt.precision, 6816 query.fmt.channels); 6817 if (query.fmt.frequency_type == 0) { 6818 DPRINTF(1, "{%d-%d", 6819 query.fmt.frequency[0], query.fmt.frequency[1]); 6820 } else { 6821 int j; 6822 for (j = 0; j < query.fmt.frequency_type; j++) { 6823 DPRINTF(1, "%c%d", 6824 (j == 0) ? '{' : ',', 6825 query.fmt.frequency[j]); 6826 } 6827 } 6828 DPRINTF(1, "}\n"); 6829 #endif 6830 6831 if ((query.fmt.mode & mode) == 0) { 6832 DPRINTF(1, "fmt[%d] skip; mode not match %d\n", i, 6833 mode); 6834 continue; 6835 } 6836 6837 if (query.fmt.priority < 0) { 6838 DPRINTF(1, "fmt[%d] skip; unsupported encoding\n", i); 6839 continue; 6840 } 6841 6842 /* Score */ 6843 score = (query.fmt.priority & 3) * 0x100; 6844 if (query.fmt.encoding == AUDIO_ENCODING_SLINEAR_NE && 6845 query.fmt.validbits == AUDIO_INTERNAL_BITS && 6846 query.fmt.precision == AUDIO_INTERNAL_BITS) { 6847 score += 0x20; 6848 } else if (query.fmt.encoding == AUDIO_ENCODING_SLINEAR_OE && 6849 query.fmt.validbits == AUDIO_INTERNAL_BITS && 6850 query.fmt.precision == AUDIO_INTERNAL_BITS) { 6851 score += 0x10; 6852 } 6853 6854 /* Do not prefer surround formats */ 6855 if (query.fmt.channels <= 2) 6856 score += query.fmt.channels; 6857 6858 if (score < cand_score) { 6859 DPRINTF(1, "fmt[%d] skip; score 0x%x < 0x%x\n", i, 6860 score, cand_score); 6861 continue; 6862 } 6863 6864 /* Update candidate */ 6865 cand_score = score; 6866 cand->encoding = query.fmt.encoding; 6867 cand->precision = query.fmt.validbits; 6868 cand->stride = query.fmt.precision; 6869 cand->channels = query.fmt.channels; 6870 cand->sample_rate = audio_select_freq(&query.fmt); 6871 DPRINTF(1, "fmt[%d] candidate (score=0x%x)" 6872 " pri=%d %s,%d/%d,%dch,%dHz\n", i, 6873 cand_score, query.fmt.priority, 6874 audio_encoding_name(query.fmt.encoding), 6875 cand->precision, cand->stride, 6876 cand->channels, cand->sample_rate); 6877 } 6878 6879 if (cand_score == 0) { 6880 DPRINTF(1, "%s no fmt\n", __func__); 6881 return ENXIO; 6882 } 6883 DPRINTF(1, "%s selected: %s,%d/%d,%dch,%dHz\n", __func__, 6884 audio_encoding_name(cand->encoding), 6885 cand->precision, cand->stride, cand->channels, cand->sample_rate); 6886 return 0; 6887 } 6888 6889 /* 6890 * Validate fmt with query_format. 6891 * If fmt is included in the result of query_format, returns 0. 6892 * Otherwise returns EINVAL. 6893 * Must be called without sc_lock held. 6894 */ 6895 static int 6896 audio_hw_validate_format(struct audio_softc *sc, int mode, 6897 const audio_format2_t *fmt) 6898 { 6899 audio_format_query_t query; 6900 struct audio_format *q; 6901 int index; 6902 int error; 6903 int j; 6904 6905 for (index = 0; ; index++) { 6906 query.index = index; 6907 mutex_enter(sc->sc_lock); 6908 error = sc->hw_if->query_format(sc->hw_hdl, &query); 6909 mutex_exit(sc->sc_lock); 6910 if (error == EINVAL) 6911 break; 6912 if (error) 6913 return error; 6914 6915 q = &query.fmt; 6916 /* 6917 * Note that fmt is audio_format2_t (precision/stride) but 6918 * q is audio_format_t (validbits/precision). 6919 */ 6920 if ((q->mode & mode) == 0) { 6921 continue; 6922 } 6923 if (fmt->encoding != q->encoding) { 6924 continue; 6925 } 6926 if (fmt->precision != q->validbits) { 6927 continue; 6928 } 6929 if (fmt->stride != q->precision) { 6930 continue; 6931 } 6932 if (fmt->channels != q->channels) { 6933 continue; 6934 } 6935 if (q->frequency_type == 0) { 6936 if (fmt->sample_rate < q->frequency[0] || 6937 fmt->sample_rate > q->frequency[1]) { 6938 continue; 6939 } 6940 } else { 6941 for (j = 0; j < q->frequency_type; j++) { 6942 if (fmt->sample_rate == q->frequency[j]) 6943 break; 6944 } 6945 if (j == query.fmt.frequency_type) { 6946 continue; 6947 } 6948 } 6949 6950 /* Matched. */ 6951 return 0; 6952 } 6953 6954 return EINVAL; 6955 } 6956 6957 /* 6958 * Set track mixer's format depending on ai->mode. 6959 * If AUMODE_PLAY is set in ai->mode, it set up the playback mixer 6960 * with ai.play.*. 6961 * If AUMODE_RECORD is set in ai->mode, it set up the recording mixer 6962 * with ai.record.*. 6963 * All other fields in ai are ignored. 6964 * If successful returns 0. Otherwise returns errno. 6965 * This function does not roll back even if it fails. 6966 * Must be called with sc_exlock held and without sc_lock held. 6967 */ 6968 static int 6969 audio_mixers_set_format(struct audio_softc *sc, const struct audio_info *ai) 6970 { 6971 audio_format2_t phwfmt; 6972 audio_format2_t rhwfmt; 6973 audio_filter_reg_t pfil; 6974 audio_filter_reg_t rfil; 6975 int mode; 6976 int error; 6977 6978 KASSERT(sc->sc_exlock); 6979 6980 /* 6981 * Even when setting either one of playback and recording, 6982 * both must be halted. 6983 */ 6984 if (sc->sc_popens + sc->sc_ropens > 0) 6985 return EBUSY; 6986 6987 if (!SPECIFIED(ai->mode) || ai->mode == 0) 6988 return ENOTTY; 6989 6990 mode = ai->mode; 6991 if ((mode & AUMODE_PLAY)) { 6992 phwfmt.encoding = ai->play.encoding; 6993 phwfmt.precision = ai->play.precision; 6994 phwfmt.stride = ai->play.precision; 6995 phwfmt.channels = ai->play.channels; 6996 phwfmt.sample_rate = ai->play.sample_rate; 6997 } 6998 if ((mode & AUMODE_RECORD)) { 6999 rhwfmt.encoding = ai->record.encoding; 7000 rhwfmt.precision = ai->record.precision; 7001 rhwfmt.stride = ai->record.precision; 7002 rhwfmt.channels = ai->record.channels; 7003 rhwfmt.sample_rate = ai->record.sample_rate; 7004 } 7005 7006 /* On non-independent devices, use the same format for both. */ 7007 if ((sc->sc_props & AUDIO_PROP_INDEPENDENT) == 0) { 7008 if (mode == AUMODE_RECORD) { 7009 phwfmt = rhwfmt; 7010 } else { 7011 rhwfmt = phwfmt; 7012 } 7013 mode = AUMODE_PLAY | AUMODE_RECORD; 7014 } 7015 7016 /* Then, unset the direction not exist on the hardware. */ 7017 if ((sc->sc_props & AUDIO_PROP_PLAYBACK) == 0) 7018 mode &= ~AUMODE_PLAY; 7019 if ((sc->sc_props & AUDIO_PROP_CAPTURE) == 0) 7020 mode &= ~AUMODE_RECORD; 7021 7022 /* debug */ 7023 if ((mode & AUMODE_PLAY)) { 7024 TRACE(1, "play=%s/%d/%d/%dch/%dHz", 7025 audio_encoding_name(phwfmt.encoding), 7026 phwfmt.precision, 7027 phwfmt.stride, 7028 phwfmt.channels, 7029 phwfmt.sample_rate); 7030 } 7031 if ((mode & AUMODE_RECORD)) { 7032 TRACE(1, "rec =%s/%d/%d/%dch/%dHz", 7033 audio_encoding_name(rhwfmt.encoding), 7034 rhwfmt.precision, 7035 rhwfmt.stride, 7036 rhwfmt.channels, 7037 rhwfmt.sample_rate); 7038 } 7039 7040 /* Check the format */ 7041 if ((mode & AUMODE_PLAY)) { 7042 if (audio_hw_validate_format(sc, AUMODE_PLAY, &phwfmt)) { 7043 TRACE(1, "invalid format"); 7044 return EINVAL; 7045 } 7046 } 7047 if ((mode & AUMODE_RECORD)) { 7048 if (audio_hw_validate_format(sc, AUMODE_RECORD, &rhwfmt)) { 7049 TRACE(1, "invalid format"); 7050 return EINVAL; 7051 } 7052 } 7053 7054 /* Configure the mixers. */ 7055 memset(&pfil, 0, sizeof(pfil)); 7056 memset(&rfil, 0, sizeof(rfil)); 7057 error = audio_hw_set_format(sc, mode, &phwfmt, &rhwfmt, &pfil, &rfil); 7058 if (error) 7059 return error; 7060 7061 error = audio_mixers_init(sc, mode, &phwfmt, &rhwfmt, &pfil, &rfil); 7062 if (error) 7063 return error; 7064 7065 /* 7066 * Reinitialize the sticky parameters for /dev/sound. 7067 * If the number of the hardware channels becomes less than the number 7068 * of channels that sticky parameters remember, subsequent /dev/sound 7069 * open will fail. To prevent this, reinitialize the sticky 7070 * parameters whenever the hardware format is changed. 7071 */ 7072 sc->sc_sound_pparams = params_to_format2(&audio_default); 7073 sc->sc_sound_rparams = params_to_format2(&audio_default); 7074 sc->sc_sound_ppause = false; 7075 sc->sc_sound_rpause = false; 7076 7077 return 0; 7078 } 7079 7080 /* 7081 * Store current mixers format into *ai. 7082 * Must be called with sc_exlock held. 7083 */ 7084 static void 7085 audio_mixers_get_format(struct audio_softc *sc, struct audio_info *ai) 7086 { 7087 7088 KASSERT(sc->sc_exlock); 7089 7090 /* 7091 * There is no stride information in audio_info but it doesn't matter. 7092 * trackmixer always treats stride and precision as the same. 7093 */ 7094 AUDIO_INITINFO(ai); 7095 ai->mode = 0; 7096 if (sc->sc_pmixer) { 7097 audio_format2_t *fmt = &sc->sc_pmixer->track_fmt; 7098 ai->play.encoding = fmt->encoding; 7099 ai->play.precision = fmt->precision; 7100 ai->play.channels = fmt->channels; 7101 ai->play.sample_rate = fmt->sample_rate; 7102 ai->mode |= AUMODE_PLAY; 7103 } 7104 if (sc->sc_rmixer) { 7105 audio_format2_t *fmt = &sc->sc_rmixer->track_fmt; 7106 ai->record.encoding = fmt->encoding; 7107 ai->record.precision = fmt->precision; 7108 ai->record.channels = fmt->channels; 7109 ai->record.sample_rate = fmt->sample_rate; 7110 ai->mode |= AUMODE_RECORD; 7111 } 7112 } 7113 7114 /* 7115 * audio_info details: 7116 * 7117 * ai.{play,record}.sample_rate (R/W) 7118 * ai.{play,record}.encoding (R/W) 7119 * ai.{play,record}.precision (R/W) 7120 * ai.{play,record}.channels (R/W) 7121 * These specify the playback or recording format. 7122 * Ignore members within an inactive track. 7123 * 7124 * ai.mode (R/W) 7125 * It specifies the playback or recording mode, AUMODE_*. 7126 * Currently, a mode change operation by ai.mode after opening is 7127 * prohibited. In addition, AUMODE_PLAY_ALL no longer makes sense. 7128 * However, it's possible to get or to set for backward compatibility. 7129 * 7130 * ai.{hiwat,lowat} (R/W) 7131 * These specify the high water mark and low water mark for playback 7132 * track. The unit is block. 7133 * 7134 * ai.{play,record}.gain (R/W) 7135 * It specifies the HW mixer volume in 0-255. 7136 * It is historical reason that the gain is connected to HW mixer. 7137 * 7138 * ai.{play,record}.balance (R/W) 7139 * It specifies the left-right balance of HW mixer in 0-64. 7140 * 32 means the center. 7141 * It is historical reason that the balance is connected to HW mixer. 7142 * 7143 * ai.{play,record}.port (R/W) 7144 * It specifies the input/output port of HW mixer. 7145 * 7146 * ai.monitor_gain (R/W) 7147 * It specifies the recording monitor gain(?) of HW mixer. 7148 * 7149 * ai.{play,record}.pause (R/W) 7150 * Non-zero means the track is paused. 7151 * 7152 * ai.play.seek (R/-) 7153 * It indicates the number of bytes written but not processed. 7154 * ai.record.seek (R/-) 7155 * It indicates the number of bytes to be able to read. 7156 * 7157 * ai.{play,record}.avail_ports (R/-) 7158 * Mixer info. 7159 * 7160 * ai.{play,record}.buffer_size (R/-) 7161 * It indicates the buffer size in bytes. Internally it means usrbuf. 7162 * 7163 * ai.{play,record}.samples (R/-) 7164 * It indicates the total number of bytes played or recorded. 7165 * 7166 * ai.{play,record}.eof (R/-) 7167 * It indicates the number of times reached EOF(?). 7168 * 7169 * ai.{play,record}.error (R/-) 7170 * Non-zero indicates overflow/underflow has occurred. 7171 * 7172 * ai.{play,record}.waiting (R/-) 7173 * Non-zero indicates that other process waits to open. 7174 * It will never happen anymore. 7175 * 7176 * ai.{play,record}.open (R/-) 7177 * Non-zero indicates the direction is opened by this process(?). 7178 * XXX Is this better to indicate that "the device is opened by 7179 * at least one process"? 7180 * 7181 * ai.{play,record}.active (R/-) 7182 * Non-zero indicates that I/O is currently active. 7183 * 7184 * ai.blocksize (R/-) 7185 * It indicates the block size in bytes. 7186 * XXX The blocksize of playback and recording may be different. 7187 */ 7188 7189 /* 7190 * Pause consideration: 7191 * 7192 * Pausing/unpausing never affect [pr]mixer. This single rule makes 7193 * operation simple. Note that playback and recording are asymmetric. 7194 * 7195 * For playback, 7196 * 1. Any playback open doesn't start pmixer regardless of initial pause 7197 * state of this track. 7198 * 2. The first write access among playback tracks only starts pmixer 7199 * regardless of this track's pause state. 7200 * 3. Even a pause of the last playback track doesn't stop pmixer. 7201 * 4. The last close of all playback tracks only stops pmixer. 7202 * 7203 * For recording, 7204 * 1. The first recording open only starts rmixer regardless of initial 7205 * pause state of this track. 7206 * 2. Even a pause of the last track doesn't stop rmixer. 7207 * 3. The last close of all recording tracks only stops rmixer. 7208 */ 7209 7210 /* 7211 * Set both track's parameters within a file depending on ai. 7212 * Update sc_sound_[pr]* if set. 7213 * Must be called with sc_exlock held and without sc_lock held. 7214 */ 7215 static int 7216 audio_file_setinfo(struct audio_softc *sc, audio_file_t *file, 7217 const struct audio_info *ai) 7218 { 7219 const struct audio_prinfo *pi; 7220 const struct audio_prinfo *ri; 7221 audio_track_t *ptrack; 7222 audio_track_t *rtrack; 7223 audio_format2_t pfmt; 7224 audio_format2_t rfmt; 7225 int pchanges; 7226 int rchanges; 7227 int mode; 7228 struct audio_info saved_ai; 7229 audio_format2_t saved_pfmt; 7230 audio_format2_t saved_rfmt; 7231 int error; 7232 7233 KASSERT(sc->sc_exlock); 7234 7235 pi = &ai->play; 7236 ri = &ai->record; 7237 pchanges = 0; 7238 rchanges = 0; 7239 7240 ptrack = file->ptrack; 7241 rtrack = file->rtrack; 7242 7243 #if defined(AUDIO_DEBUG) 7244 if (audiodebug >= 2) { 7245 char buf[256]; 7246 char p[64]; 7247 int buflen; 7248 int plen; 7249 #define SPRINTF(var, fmt...) do { \ 7250 var##len += snprintf(var + var##len, sizeof(var) - var##len, fmt); \ 7251 } while (0) 7252 7253 buflen = 0; 7254 plen = 0; 7255 if (SPECIFIED(pi->encoding)) 7256 SPRINTF(p, "/%s", audio_encoding_name(pi->encoding)); 7257 if (SPECIFIED(pi->precision)) 7258 SPRINTF(p, "/%dbit", pi->precision); 7259 if (SPECIFIED(pi->channels)) 7260 SPRINTF(p, "/%dch", pi->channels); 7261 if (SPECIFIED(pi->sample_rate)) 7262 SPRINTF(p, "/%dHz", pi->sample_rate); 7263 if (plen > 0) 7264 SPRINTF(buf, ",play.param=%s", p + 1); 7265 7266 plen = 0; 7267 if (SPECIFIED(ri->encoding)) 7268 SPRINTF(p, "/%s", audio_encoding_name(ri->encoding)); 7269 if (SPECIFIED(ri->precision)) 7270 SPRINTF(p, "/%dbit", ri->precision); 7271 if (SPECIFIED(ri->channels)) 7272 SPRINTF(p, "/%dch", ri->channels); 7273 if (SPECIFIED(ri->sample_rate)) 7274 SPRINTF(p, "/%dHz", ri->sample_rate); 7275 if (plen > 0) 7276 SPRINTF(buf, ",record.param=%s", p + 1); 7277 7278 if (SPECIFIED(ai->mode)) 7279 SPRINTF(buf, ",mode=%d", ai->mode); 7280 if (SPECIFIED(ai->hiwat)) 7281 SPRINTF(buf, ",hiwat=%d", ai->hiwat); 7282 if (SPECIFIED(ai->lowat)) 7283 SPRINTF(buf, ",lowat=%d", ai->lowat); 7284 if (SPECIFIED(ai->play.gain)) 7285 SPRINTF(buf, ",play.gain=%d", ai->play.gain); 7286 if (SPECIFIED(ai->record.gain)) 7287 SPRINTF(buf, ",record.gain=%d", ai->record.gain); 7288 if (SPECIFIED_CH(ai->play.balance)) 7289 SPRINTF(buf, ",play.balance=%d", ai->play.balance); 7290 if (SPECIFIED_CH(ai->record.balance)) 7291 SPRINTF(buf, ",record.balance=%d", ai->record.balance); 7292 if (SPECIFIED(ai->play.port)) 7293 SPRINTF(buf, ",play.port=%d", ai->play.port); 7294 if (SPECIFIED(ai->record.port)) 7295 SPRINTF(buf, ",record.port=%d", ai->record.port); 7296 if (SPECIFIED(ai->monitor_gain)) 7297 SPRINTF(buf, ",monitor_gain=%d", ai->monitor_gain); 7298 if (SPECIFIED_CH(ai->play.pause)) 7299 SPRINTF(buf, ",play.pause=%d", ai->play.pause); 7300 if (SPECIFIED_CH(ai->record.pause)) 7301 SPRINTF(buf, ",record.pause=%d", ai->record.pause); 7302 7303 if (buflen > 0) 7304 TRACE(2, "specified %s", buf + 1); 7305 } 7306 #endif 7307 7308 AUDIO_INITINFO(&saved_ai); 7309 /* XXX shut up gcc */ 7310 memset(&saved_pfmt, 0, sizeof(saved_pfmt)); 7311 memset(&saved_rfmt, 0, sizeof(saved_rfmt)); 7312 7313 /* 7314 * Set default value and save current parameters. 7315 * For backward compatibility, use sticky parameters for nonexistent 7316 * track. 7317 */ 7318 if (ptrack) { 7319 pfmt = ptrack->usrbuf.fmt; 7320 saved_pfmt = ptrack->usrbuf.fmt; 7321 saved_ai.play.pause = ptrack->is_pause; 7322 } else { 7323 pfmt = sc->sc_sound_pparams; 7324 } 7325 if (rtrack) { 7326 rfmt = rtrack->usrbuf.fmt; 7327 saved_rfmt = rtrack->usrbuf.fmt; 7328 saved_ai.record.pause = rtrack->is_pause; 7329 } else { 7330 rfmt = sc->sc_sound_rparams; 7331 } 7332 saved_ai.mode = file->mode; 7333 7334 /* 7335 * Overwrite if specified. 7336 */ 7337 mode = file->mode; 7338 if (SPECIFIED(ai->mode)) { 7339 /* 7340 * Setting ai->mode no longer does anything because it's 7341 * prohibited to change playback/recording mode after open 7342 * and AUMODE_PLAY_ALL is obsoleted. However, it still 7343 * keeps the state of AUMODE_PLAY_ALL itself for backward 7344 * compatibility. 7345 * In the internal, only file->mode has the state of 7346 * AUMODE_PLAY_ALL flag and track->mode in both track does 7347 * not have. 7348 */ 7349 if ((file->mode & AUMODE_PLAY)) { 7350 mode = (file->mode & (AUMODE_PLAY | AUMODE_RECORD)) 7351 | (ai->mode & AUMODE_PLAY_ALL); 7352 } 7353 } 7354 7355 pchanges = audio_track_setinfo_check(ptrack, &pfmt, pi); 7356 if (pchanges == -1) { 7357 #if defined(AUDIO_DEBUG) 7358 TRACEF(1, file, "check play.params failed: " 7359 "%s %ubit %uch %uHz", 7360 audio_encoding_name(pi->encoding), 7361 pi->precision, 7362 pi->channels, 7363 pi->sample_rate); 7364 #endif 7365 return EINVAL; 7366 } 7367 7368 rchanges = audio_track_setinfo_check(rtrack, &rfmt, ri); 7369 if (rchanges == -1) { 7370 #if defined(AUDIO_DEBUG) 7371 TRACEF(1, file, "check record.params failed: " 7372 "%s %ubit %uch %uHz", 7373 audio_encoding_name(ri->encoding), 7374 ri->precision, 7375 ri->channels, 7376 ri->sample_rate); 7377 #endif 7378 return EINVAL; 7379 } 7380 7381 if (SPECIFIED(ai->mode)) { 7382 pchanges = 1; 7383 rchanges = 1; 7384 } 7385 7386 /* 7387 * Even when setting either one of playback and recording, 7388 * both track must be halted. 7389 */ 7390 if (pchanges || rchanges) { 7391 audio_file_clear(sc, file); 7392 #if defined(AUDIO_DEBUG) 7393 char nbuf[16]; 7394 char fmtbuf[64]; 7395 if (pchanges) { 7396 if (ptrack) { 7397 snprintf(nbuf, sizeof(nbuf), "%d", ptrack->id); 7398 } else { 7399 snprintf(nbuf, sizeof(nbuf), "-"); 7400 } 7401 audio_format2_tostr(fmtbuf, sizeof(fmtbuf), &pfmt); 7402 DPRINTF(1, "audio track#%s play mode: %s\n", 7403 nbuf, fmtbuf); 7404 } 7405 if (rchanges) { 7406 if (rtrack) { 7407 snprintf(nbuf, sizeof(nbuf), "%d", rtrack->id); 7408 } else { 7409 snprintf(nbuf, sizeof(nbuf), "-"); 7410 } 7411 audio_format2_tostr(fmtbuf, sizeof(fmtbuf), &rfmt); 7412 DPRINTF(1, "audio track#%s rec mode: %s\n", 7413 nbuf, fmtbuf); 7414 } 7415 #endif 7416 } 7417 7418 /* Set mixer parameters */ 7419 mutex_enter(sc->sc_lock); 7420 error = audio_hw_setinfo(sc, ai, &saved_ai); 7421 mutex_exit(sc->sc_lock); 7422 if (error) 7423 goto abort1; 7424 7425 /* 7426 * Set to track and update sticky parameters. 7427 */ 7428 error = 0; 7429 file->mode = mode; 7430 7431 if (SPECIFIED_CH(pi->pause)) { 7432 if (ptrack) 7433 ptrack->is_pause = pi->pause; 7434 sc->sc_sound_ppause = pi->pause; 7435 } 7436 if (pchanges) { 7437 if (ptrack) { 7438 audio_track_lock_enter(ptrack); 7439 error = audio_track_set_format(ptrack, &pfmt); 7440 audio_track_lock_exit(ptrack); 7441 if (error) { 7442 TRACET(1, ptrack, "set play.params failed"); 7443 goto abort2; 7444 } 7445 } 7446 sc->sc_sound_pparams = pfmt; 7447 } 7448 /* Change water marks after initializing the buffers. */ 7449 if (SPECIFIED(ai->hiwat) || SPECIFIED(ai->lowat)) { 7450 if (ptrack) 7451 audio_track_setinfo_water(ptrack, ai); 7452 } 7453 7454 if (SPECIFIED_CH(ri->pause)) { 7455 if (rtrack) 7456 rtrack->is_pause = ri->pause; 7457 sc->sc_sound_rpause = ri->pause; 7458 } 7459 if (rchanges) { 7460 if (rtrack) { 7461 audio_track_lock_enter(rtrack); 7462 error = audio_track_set_format(rtrack, &rfmt); 7463 audio_track_lock_exit(rtrack); 7464 if (error) { 7465 TRACET(1, rtrack, "set record.params failed"); 7466 goto abort3; 7467 } 7468 } 7469 sc->sc_sound_rparams = rfmt; 7470 } 7471 7472 return 0; 7473 7474 /* Rollback */ 7475 abort3: 7476 if (error != ENOMEM) { 7477 rtrack->is_pause = saved_ai.record.pause; 7478 audio_track_lock_enter(rtrack); 7479 audio_track_set_format(rtrack, &saved_rfmt); 7480 audio_track_lock_exit(rtrack); 7481 } 7482 sc->sc_sound_rpause = saved_ai.record.pause; 7483 sc->sc_sound_rparams = saved_rfmt; 7484 abort2: 7485 if (ptrack && error != ENOMEM) { 7486 ptrack->is_pause = saved_ai.play.pause; 7487 audio_track_lock_enter(ptrack); 7488 audio_track_set_format(ptrack, &saved_pfmt); 7489 audio_track_lock_exit(ptrack); 7490 } 7491 sc->sc_sound_ppause = saved_ai.play.pause; 7492 sc->sc_sound_pparams = saved_pfmt; 7493 file->mode = saved_ai.mode; 7494 abort1: 7495 mutex_enter(sc->sc_lock); 7496 audio_hw_setinfo(sc, &saved_ai, NULL); 7497 mutex_exit(sc->sc_lock); 7498 7499 return error; 7500 } 7501 7502 /* 7503 * Write SPECIFIED() parameters within info back to fmt. 7504 * Note that track can be NULL here. 7505 * Return value of 1 indicates that fmt is modified. 7506 * Return value of 0 indicates that fmt is not modified. 7507 * Return value of -1 indicates that error EINVAL has occurred. 7508 */ 7509 static int 7510 audio_track_setinfo_check(audio_track_t *track, 7511 audio_format2_t *fmt, const struct audio_prinfo *info) 7512 { 7513 const audio_format2_t *hwfmt; 7514 int changes; 7515 7516 changes = 0; 7517 if (SPECIFIED(info->sample_rate)) { 7518 if (info->sample_rate < AUDIO_MIN_FREQUENCY) 7519 return -1; 7520 if (info->sample_rate > AUDIO_MAX_FREQUENCY) 7521 return -1; 7522 fmt->sample_rate = info->sample_rate; 7523 changes = 1; 7524 } 7525 if (SPECIFIED(info->encoding)) { 7526 fmt->encoding = info->encoding; 7527 changes = 1; 7528 } 7529 if (SPECIFIED(info->precision)) { 7530 fmt->precision = info->precision; 7531 /* we don't have API to specify stride */ 7532 fmt->stride = info->precision; 7533 changes = 1; 7534 } 7535 if (SPECIFIED(info->channels)) { 7536 /* 7537 * We can convert between monaural and stereo each other. 7538 * We can reduce than the number of channels that the hardware 7539 * supports. 7540 */ 7541 if (info->channels > 2) { 7542 if (track) { 7543 hwfmt = &track->mixer->hwbuf.fmt; 7544 if (info->channels > hwfmt->channels) 7545 return -1; 7546 } else { 7547 /* 7548 * This should never happen. 7549 * If track == NULL, channels should be <= 2. 7550 */ 7551 return -1; 7552 } 7553 } 7554 fmt->channels = info->channels; 7555 changes = 1; 7556 } 7557 7558 if (changes) { 7559 if (audio_check_params(fmt) != 0) 7560 return -1; 7561 } 7562 7563 return changes; 7564 } 7565 7566 /* 7567 * Change water marks for playback track if specified. 7568 */ 7569 static void 7570 audio_track_setinfo_water(audio_track_t *track, const struct audio_info *ai) 7571 { 7572 u_int blks; 7573 u_int maxblks; 7574 u_int blksize; 7575 7576 KASSERT(audio_track_is_playback(track)); 7577 7578 blksize = track->usrbuf_blksize; 7579 maxblks = track->usrbuf.capacity / blksize; 7580 7581 if (SPECIFIED(ai->hiwat)) { 7582 blks = ai->hiwat; 7583 if (blks > maxblks) 7584 blks = maxblks; 7585 if (blks < 2) 7586 blks = 2; 7587 track->usrbuf_usedhigh = blks * blksize; 7588 } 7589 if (SPECIFIED(ai->lowat)) { 7590 blks = ai->lowat; 7591 if (blks > maxblks - 1) 7592 blks = maxblks - 1; 7593 track->usrbuf_usedlow = blks * blksize; 7594 } 7595 if (SPECIFIED(ai->hiwat) || SPECIFIED(ai->lowat)) { 7596 if (track->usrbuf_usedlow > track->usrbuf_usedhigh - blksize) { 7597 track->usrbuf_usedlow = track->usrbuf_usedhigh - 7598 blksize; 7599 } 7600 } 7601 } 7602 7603 /* 7604 * Set hardware part of *newai. 7605 * The parameters handled here are *.port, *.gain, *.balance and monitor_gain. 7606 * If oldai is specified, previous parameters are stored. 7607 * This function itself does not roll back if error occurred. 7608 * Must be called with sc_lock && sc_exlock held. 7609 */ 7610 static int 7611 audio_hw_setinfo(struct audio_softc *sc, const struct audio_info *newai, 7612 struct audio_info *oldai) 7613 { 7614 const struct audio_prinfo *newpi; 7615 const struct audio_prinfo *newri; 7616 struct audio_prinfo *oldpi; 7617 struct audio_prinfo *oldri; 7618 u_int pgain; 7619 u_int rgain; 7620 u_char pbalance; 7621 u_char rbalance; 7622 int error; 7623 7624 KASSERT(mutex_owned(sc->sc_lock)); 7625 KASSERT(sc->sc_exlock); 7626 7627 /* XXX shut up gcc */ 7628 oldpi = NULL; 7629 oldri = NULL; 7630 7631 newpi = &newai->play; 7632 newri = &newai->record; 7633 if (oldai) { 7634 oldpi = &oldai->play; 7635 oldri = &oldai->record; 7636 } 7637 error = 0; 7638 7639 /* 7640 * It looks like unnecessary to halt HW mixers to set HW mixers. 7641 * mixer_ioctl(MIXER_WRITE) also doesn't halt. 7642 */ 7643 7644 if (SPECIFIED(newpi->port)) { 7645 if (oldai) 7646 oldpi->port = au_get_port(sc, &sc->sc_outports); 7647 error = au_set_port(sc, &sc->sc_outports, newpi->port); 7648 if (error) { 7649 audio_printf(sc, 7650 "setting play.port=%d failed: errno=%d\n", 7651 newpi->port, error); 7652 goto abort; 7653 } 7654 } 7655 if (SPECIFIED(newri->port)) { 7656 if (oldai) 7657 oldri->port = au_get_port(sc, &sc->sc_inports); 7658 error = au_set_port(sc, &sc->sc_inports, newri->port); 7659 if (error) { 7660 audio_printf(sc, 7661 "setting record.port=%d failed: errno=%d\n", 7662 newri->port, error); 7663 goto abort; 7664 } 7665 } 7666 7667 /* play.{gain,balance} */ 7668 if (SPECIFIED(newpi->gain) || SPECIFIED_CH(newpi->balance)) { 7669 au_get_gain(sc, &sc->sc_outports, &pgain, &pbalance); 7670 if (oldai) { 7671 oldpi->gain = pgain; 7672 oldpi->balance = pbalance; 7673 } 7674 7675 if (SPECIFIED(newpi->gain)) 7676 pgain = newpi->gain; 7677 if (SPECIFIED_CH(newpi->balance)) 7678 pbalance = newpi->balance; 7679 error = au_set_gain(sc, &sc->sc_outports, pgain, pbalance); 7680 if (error) { 7681 audio_printf(sc, 7682 "setting play.gain=%d/balance=%d failed: " 7683 "errno=%d\n", 7684 pgain, pbalance, error); 7685 goto abort; 7686 } 7687 } 7688 7689 /* record.{gain,balance} */ 7690 if (SPECIFIED(newri->gain) || SPECIFIED_CH(newri->balance)) { 7691 au_get_gain(sc, &sc->sc_inports, &rgain, &rbalance); 7692 if (oldai) { 7693 oldri->gain = rgain; 7694 oldri->balance = rbalance; 7695 } 7696 7697 if (SPECIFIED(newri->gain)) 7698 rgain = newri->gain; 7699 if (SPECIFIED_CH(newri->balance)) 7700 rbalance = newri->balance; 7701 error = au_set_gain(sc, &sc->sc_inports, rgain, rbalance); 7702 if (error) { 7703 audio_printf(sc, 7704 "setting record.gain=%d/balance=%d failed: " 7705 "errno=%d\n", 7706 rgain, rbalance, error); 7707 goto abort; 7708 } 7709 } 7710 7711 if (SPECIFIED(newai->monitor_gain) && sc->sc_monitor_port != -1) { 7712 if (oldai) 7713 oldai->monitor_gain = au_get_monitor_gain(sc); 7714 error = au_set_monitor_gain(sc, newai->monitor_gain); 7715 if (error) { 7716 audio_printf(sc, 7717 "setting monitor_gain=%d failed: errno=%d\n", 7718 newai->monitor_gain, error); 7719 goto abort; 7720 } 7721 } 7722 7723 /* XXX TODO */ 7724 /* sc->sc_ai = *ai; */ 7725 7726 error = 0; 7727 abort: 7728 return error; 7729 } 7730 7731 /* 7732 * Setup the hardware with mixer format phwfmt, rhwfmt. 7733 * The arguments have following restrictions: 7734 * - setmode is the direction you want to set, AUMODE_PLAY or AUMODE_RECORD, 7735 * or both. 7736 * - phwfmt and rhwfmt must not be NULL regardless of setmode. 7737 * - On non-independent devices, phwfmt and rhwfmt must have the same 7738 * parameters. 7739 * - pfil and rfil must be zero-filled. 7740 * If successful, 7741 * - pfil, rfil will be filled with filter information specified by the 7742 * hardware driver if necessary. 7743 * and then returns 0. Otherwise returns errno. 7744 * Must be called without sc_lock held. 7745 */ 7746 static int 7747 audio_hw_set_format(struct audio_softc *sc, int setmode, 7748 const audio_format2_t *phwfmt, const audio_format2_t *rhwfmt, 7749 audio_filter_reg_t *pfil, audio_filter_reg_t *rfil) 7750 { 7751 audio_params_t pp, rp; 7752 int error; 7753 7754 KASSERT(phwfmt != NULL); 7755 KASSERT(rhwfmt != NULL); 7756 7757 pp = format2_to_params(phwfmt); 7758 rp = format2_to_params(rhwfmt); 7759 7760 mutex_enter(sc->sc_lock); 7761 error = sc->hw_if->set_format(sc->hw_hdl, setmode, 7762 &pp, &rp, pfil, rfil); 7763 if (error) { 7764 mutex_exit(sc->sc_lock); 7765 audio_printf(sc, "set_format failed: errno=%d\n", error); 7766 return error; 7767 } 7768 7769 if (sc->hw_if->commit_settings) { 7770 error = sc->hw_if->commit_settings(sc->hw_hdl); 7771 if (error) { 7772 mutex_exit(sc->sc_lock); 7773 audio_printf(sc, 7774 "commit_settings failed: errno=%d\n", error); 7775 return error; 7776 } 7777 } 7778 mutex_exit(sc->sc_lock); 7779 7780 return 0; 7781 } 7782 7783 /* 7784 * Fill audio_info structure. If need_mixerinfo is true, it will also 7785 * fill the hardware mixer information. 7786 * Must be called with sc_exlock held and without sc_lock held. 7787 */ 7788 static int 7789 audiogetinfo(struct audio_softc *sc, struct audio_info *ai, int need_mixerinfo, 7790 audio_file_t *file) 7791 { 7792 struct audio_prinfo *ri, *pi; 7793 audio_track_t *track; 7794 audio_track_t *ptrack; 7795 audio_track_t *rtrack; 7796 int gain; 7797 7798 KASSERT(sc->sc_exlock); 7799 7800 ri = &ai->record; 7801 pi = &ai->play; 7802 ptrack = file->ptrack; 7803 rtrack = file->rtrack; 7804 7805 memset(ai, 0, sizeof(*ai)); 7806 7807 if (ptrack) { 7808 pi->sample_rate = ptrack->usrbuf.fmt.sample_rate; 7809 pi->channels = ptrack->usrbuf.fmt.channels; 7810 pi->precision = ptrack->usrbuf.fmt.precision; 7811 pi->encoding = ptrack->usrbuf.fmt.encoding; 7812 pi->pause = ptrack->is_pause; 7813 } else { 7814 /* Use sticky parameters if the track is not available. */ 7815 pi->sample_rate = sc->sc_sound_pparams.sample_rate; 7816 pi->channels = sc->sc_sound_pparams.channels; 7817 pi->precision = sc->sc_sound_pparams.precision; 7818 pi->encoding = sc->sc_sound_pparams.encoding; 7819 pi->pause = sc->sc_sound_ppause; 7820 } 7821 if (rtrack) { 7822 ri->sample_rate = rtrack->usrbuf.fmt.sample_rate; 7823 ri->channels = rtrack->usrbuf.fmt.channels; 7824 ri->precision = rtrack->usrbuf.fmt.precision; 7825 ri->encoding = rtrack->usrbuf.fmt.encoding; 7826 ri->pause = rtrack->is_pause; 7827 } else { 7828 /* Use sticky parameters if the track is not available. */ 7829 ri->sample_rate = sc->sc_sound_rparams.sample_rate; 7830 ri->channels = sc->sc_sound_rparams.channels; 7831 ri->precision = sc->sc_sound_rparams.precision; 7832 ri->encoding = sc->sc_sound_rparams.encoding; 7833 ri->pause = sc->sc_sound_rpause; 7834 } 7835 7836 if (ptrack) { 7837 pi->seek = ptrack->usrbuf.used; 7838 pi->samples = ptrack->stamp * ptrack->usrbuf_blksize; 7839 pi->eof = ptrack->eofcounter; 7840 pi->error = (ptrack->dropframes != 0) ? 1 : 0; 7841 pi->open = 1; 7842 pi->buffer_size = ptrack->usrbuf.capacity; 7843 } 7844 pi->waiting = 0; /* open never hangs */ 7845 pi->active = sc->sc_pbusy; 7846 7847 if (rtrack) { 7848 ri->seek = audio_track_readablebytes(rtrack); 7849 ri->samples = rtrack->stamp * rtrack->usrbuf_blksize; 7850 ri->eof = 0; 7851 ri->error = (rtrack->dropframes != 0) ? 1 : 0; 7852 ri->open = 1; 7853 ri->buffer_size = audio_track_inputblk_as_usrbyte(rtrack, 7854 rtrack->input->capacity); 7855 } 7856 ri->waiting = 0; /* open never hangs */ 7857 ri->active = sc->sc_rbusy; 7858 7859 /* 7860 * XXX There may be different number of channels between playback 7861 * and recording, so that blocksize also may be different. 7862 * But struct audio_info has an united blocksize... 7863 * Here, I use play info precedencely if ptrack is available, 7864 * otherwise record info. 7865 * 7866 * XXX hiwat/lowat is a playback-only parameter. What should I 7867 * return for a record-only descriptor? 7868 */ 7869 track = ptrack ? ptrack : rtrack; 7870 if (track) { 7871 ai->blocksize = track->usrbuf_blksize; 7872 ai->hiwat = track->usrbuf_usedhigh / track->usrbuf_blksize; 7873 ai->lowat = track->usrbuf_usedlow / track->usrbuf_blksize; 7874 } 7875 ai->mode = file->mode; 7876 7877 /* 7878 * For backward compatibility, we have to pad these five fields 7879 * a fake non-zero value even if there are no tracks. 7880 */ 7881 if (ptrack == NULL) 7882 pi->buffer_size = 65536; 7883 if (rtrack == NULL) 7884 ri->buffer_size = 65536; 7885 if (ptrack == NULL && rtrack == NULL) { 7886 ai->blocksize = 2048; 7887 ai->hiwat = ai->play.buffer_size / ai->blocksize; 7888 ai->lowat = ai->hiwat * 3 / 4; 7889 } 7890 7891 if (need_mixerinfo) { 7892 mutex_enter(sc->sc_lock); 7893 7894 pi->port = au_get_port(sc, &sc->sc_outports); 7895 ri->port = au_get_port(sc, &sc->sc_inports); 7896 7897 pi->avail_ports = sc->sc_outports.allports; 7898 ri->avail_ports = sc->sc_inports.allports; 7899 7900 au_get_gain(sc, &sc->sc_outports, &pi->gain, &pi->balance); 7901 au_get_gain(sc, &sc->sc_inports, &ri->gain, &ri->balance); 7902 7903 if (sc->sc_monitor_port != -1) { 7904 gain = au_get_monitor_gain(sc); 7905 if (gain != -1) 7906 ai->monitor_gain = gain; 7907 } 7908 mutex_exit(sc->sc_lock); 7909 } 7910 7911 return 0; 7912 } 7913 7914 /* 7915 * Return true if playback is configured. 7916 * This function can be used after audioattach. 7917 */ 7918 static bool 7919 audio_can_playback(struct audio_softc *sc) 7920 { 7921 7922 return (sc->sc_pmixer != NULL); 7923 } 7924 7925 /* 7926 * Return true if recording is configured. 7927 * This function can be used after audioattach. 7928 */ 7929 static bool 7930 audio_can_capture(struct audio_softc *sc) 7931 { 7932 7933 return (sc->sc_rmixer != NULL); 7934 } 7935 7936 /* 7937 * Get the afp->index'th item from the valid one of format[]. 7938 * If found, stores it to afp->fmt and returns 0. Otherwise return EINVAL. 7939 * 7940 * This is common routines for query_format. 7941 * If your hardware driver has struct audio_format[], the simplest case 7942 * you can write your query_format interface as follows: 7943 * 7944 * struct audio_format foo_format[] = { ... }; 7945 * 7946 * int 7947 * foo_query_format(void *hdl, audio_format_query_t *afp) 7948 * { 7949 * return audio_query_format(foo_format, __arraycount(foo_format), afp); 7950 * } 7951 */ 7952 int 7953 audio_query_format(const struct audio_format *format, int nformats, 7954 audio_format_query_t *afp) 7955 { 7956 const struct audio_format *f; 7957 int idx; 7958 int i; 7959 7960 idx = 0; 7961 for (i = 0; i < nformats; i++) { 7962 f = &format[i]; 7963 if (!AUFMT_IS_VALID(f)) 7964 continue; 7965 if (afp->index == idx) { 7966 afp->fmt = *f; 7967 return 0; 7968 } 7969 idx++; 7970 } 7971 return EINVAL; 7972 } 7973 7974 /* 7975 * This function is provided for the hardware driver's set_format() to 7976 * find index matches with 'param' from array of audio_format_t 'formats'. 7977 * 'mode' is either of AUMODE_PLAY or AUMODE_RECORD. 7978 * It returns the matched index and never fails. Because param passed to 7979 * set_format() is selected from query_format(). 7980 * This function will be an alternative to auconv_set_converter() to 7981 * find index. 7982 */ 7983 int 7984 audio_indexof_format(const struct audio_format *formats, int nformats, 7985 int mode, const audio_params_t *param) 7986 { 7987 const struct audio_format *f; 7988 int index; 7989 int j; 7990 7991 for (index = 0; index < nformats; index++) { 7992 f = &formats[index]; 7993 7994 if (!AUFMT_IS_VALID(f)) 7995 continue; 7996 if ((f->mode & mode) == 0) 7997 continue; 7998 if (f->encoding != param->encoding) 7999 continue; 8000 if (f->validbits != param->precision) 8001 continue; 8002 if (f->channels != param->channels) 8003 continue; 8004 8005 if (f->frequency_type == 0) { 8006 if (param->sample_rate < f->frequency[0] || 8007 param->sample_rate > f->frequency[1]) 8008 continue; 8009 } else { 8010 for (j = 0; j < f->frequency_type; j++) { 8011 if (param->sample_rate == f->frequency[j]) 8012 break; 8013 } 8014 if (j == f->frequency_type) 8015 continue; 8016 } 8017 8018 /* Then, matched */ 8019 return index; 8020 } 8021 8022 /* Not matched. This should not be happened. */ 8023 panic("%s: cannot find matched format\n", __func__); 8024 } 8025 8026 /* 8027 * Get or set hardware blocksize in msec. 8028 * XXX It's for debug. 8029 */ 8030 static int 8031 audio_sysctl_blk_ms(SYSCTLFN_ARGS) 8032 { 8033 struct sysctlnode node; 8034 struct audio_softc *sc; 8035 audio_format2_t phwfmt; 8036 audio_format2_t rhwfmt; 8037 audio_filter_reg_t pfil; 8038 audio_filter_reg_t rfil; 8039 int t; 8040 int old_blk_ms; 8041 int mode; 8042 int error; 8043 8044 node = *rnode; 8045 sc = node.sysctl_data; 8046 8047 error = audio_exlock_enter(sc); 8048 if (error) 8049 return error; 8050 8051 old_blk_ms = sc->sc_blk_ms; 8052 t = old_blk_ms; 8053 node.sysctl_data = &t; 8054 error = sysctl_lookup(SYSCTLFN_CALL(&node)); 8055 if (error || newp == NULL) 8056 goto abort; 8057 8058 if (t < 0) { 8059 error = EINVAL; 8060 goto abort; 8061 } 8062 8063 if (sc->sc_popens + sc->sc_ropens > 0) { 8064 error = EBUSY; 8065 goto abort; 8066 } 8067 sc->sc_blk_ms = t; 8068 mode = 0; 8069 if (sc->sc_pmixer) { 8070 mode |= AUMODE_PLAY; 8071 phwfmt = sc->sc_pmixer->hwbuf.fmt; 8072 } 8073 if (sc->sc_rmixer) { 8074 mode |= AUMODE_RECORD; 8075 rhwfmt = sc->sc_rmixer->hwbuf.fmt; 8076 } 8077 8078 /* re-init hardware */ 8079 memset(&pfil, 0, sizeof(pfil)); 8080 memset(&rfil, 0, sizeof(rfil)); 8081 error = audio_hw_set_format(sc, mode, &phwfmt, &rhwfmt, &pfil, &rfil); 8082 if (error) { 8083 goto abort; 8084 } 8085 8086 /* re-init track mixer */ 8087 error = audio_mixers_init(sc, mode, &phwfmt, &rhwfmt, &pfil, &rfil); 8088 if (error) { 8089 /* Rollback */ 8090 sc->sc_blk_ms = old_blk_ms; 8091 audio_mixers_init(sc, mode, &phwfmt, &rhwfmt, &pfil, &rfil); 8092 goto abort; 8093 } 8094 error = 0; 8095 abort: 8096 audio_exlock_exit(sc); 8097 return error; 8098 } 8099 8100 /* 8101 * Get or set multiuser mode. 8102 */ 8103 static int 8104 audio_sysctl_multiuser(SYSCTLFN_ARGS) 8105 { 8106 struct sysctlnode node; 8107 struct audio_softc *sc; 8108 bool t; 8109 int error; 8110 8111 node = *rnode; 8112 sc = node.sysctl_data; 8113 8114 error = audio_exlock_enter(sc); 8115 if (error) 8116 return error; 8117 8118 t = sc->sc_multiuser; 8119 node.sysctl_data = &t; 8120 error = sysctl_lookup(SYSCTLFN_CALL(&node)); 8121 if (error || newp == NULL) 8122 goto abort; 8123 8124 sc->sc_multiuser = t; 8125 error = 0; 8126 abort: 8127 audio_exlock_exit(sc); 8128 return error; 8129 } 8130 8131 #if defined(AUDIO_DEBUG) 8132 /* 8133 * Get or set debug verbose level. (0..4) 8134 * XXX It's for debug. 8135 * XXX It is not separated per device. 8136 */ 8137 static int 8138 audio_sysctl_debug(SYSCTLFN_ARGS) 8139 { 8140 struct sysctlnode node; 8141 int t; 8142 int error; 8143 8144 node = *rnode; 8145 t = audiodebug; 8146 node.sysctl_data = &t; 8147 error = sysctl_lookup(SYSCTLFN_CALL(&node)); 8148 if (error || newp == NULL) 8149 return error; 8150 8151 if (t < 0 || t > 4) 8152 return EINVAL; 8153 audiodebug = t; 8154 printf("audio: audiodebug = %d\n", audiodebug); 8155 return 0; 8156 } 8157 #endif /* AUDIO_DEBUG */ 8158 8159 #ifdef AUDIO_PM_IDLE 8160 static void 8161 audio_idle(void *arg) 8162 { 8163 device_t dv = arg; 8164 struct audio_softc *sc = device_private(dv); 8165 8166 #ifdef PNP_DEBUG 8167 extern int pnp_debug_idle; 8168 if (pnp_debug_idle) 8169 printf("%s: idle handler called\n", device_xname(dv)); 8170 #endif 8171 8172 sc->sc_idle = true; 8173 8174 /* XXX joerg Make pmf_device_suspend handle children? */ 8175 if (!pmf_device_suspend(dv, PMF_Q_SELF)) 8176 return; 8177 8178 if (!pmf_device_suspend(sc->hw_dev, PMF_Q_SELF)) 8179 pmf_device_resume(dv, PMF_Q_SELF); 8180 } 8181 8182 static void 8183 audio_activity(device_t dv, devactive_t type) 8184 { 8185 struct audio_softc *sc = device_private(dv); 8186 8187 if (type != DVA_SYSTEM) 8188 return; 8189 8190 callout_schedule(&sc->sc_idle_counter, audio_idle_timeout * hz); 8191 8192 sc->sc_idle = false; 8193 if (!device_is_active(dv)) { 8194 /* XXX joerg How to deal with a failing resume... */ 8195 pmf_device_resume(sc->hw_dev, PMF_Q_SELF); 8196 pmf_device_resume(dv, PMF_Q_SELF); 8197 } 8198 } 8199 #endif 8200 8201 static bool 8202 audio_suspend(device_t dv, const pmf_qual_t *qual) 8203 { 8204 struct audio_softc *sc = device_private(dv); 8205 int error; 8206 8207 error = audio_exlock_mutex_enter(sc); 8208 if (error) 8209 return error; 8210 sc->sc_suspending = true; 8211 audio_mixer_capture(sc); 8212 8213 if (sc->sc_pbusy) { 8214 audio_pmixer_halt(sc); 8215 /* Reuse this as need-to-restart flag while suspending */ 8216 sc->sc_pbusy = true; 8217 } 8218 if (sc->sc_rbusy) { 8219 audio_rmixer_halt(sc); 8220 /* Reuse this as need-to-restart flag while suspending */ 8221 sc->sc_rbusy = true; 8222 } 8223 8224 #ifdef AUDIO_PM_IDLE 8225 callout_halt(&sc->sc_idle_counter, sc->sc_lock); 8226 #endif 8227 audio_exlock_mutex_exit(sc); 8228 8229 return true; 8230 } 8231 8232 static bool 8233 audio_resume(device_t dv, const pmf_qual_t *qual) 8234 { 8235 struct audio_softc *sc = device_private(dv); 8236 struct audio_info ai; 8237 int error; 8238 8239 error = audio_exlock_mutex_enter(sc); 8240 if (error) 8241 return error; 8242 8243 sc->sc_suspending = false; 8244 audio_mixer_restore(sc); 8245 /* XXX ? */ 8246 AUDIO_INITINFO(&ai); 8247 audio_hw_setinfo(sc, &ai, NULL); 8248 8249 /* 8250 * During from suspend to resume here, sc_[pr]busy is used as 8251 * need-to-restart flag temporarily. After this point, 8252 * sc_[pr]busy is returned to its original usage (busy flag). 8253 * And note that sc_[pr]busy must be false to call [pr]mixer_start(). 8254 */ 8255 if (sc->sc_pbusy) { 8256 /* pmixer_start() requires pbusy is false */ 8257 sc->sc_pbusy = false; 8258 audio_pmixer_start(sc, true); 8259 } 8260 if (sc->sc_rbusy) { 8261 /* rmixer_start() requires rbusy is false */ 8262 sc->sc_rbusy = false; 8263 audio_rmixer_start(sc); 8264 } 8265 8266 audio_exlock_mutex_exit(sc); 8267 8268 return true; 8269 } 8270 8271 #if defined(AUDIO_DEBUG) 8272 static void 8273 audio_format2_tostr(char *buf, size_t bufsize, const audio_format2_t *fmt) 8274 { 8275 int n; 8276 8277 n = 0; 8278 n += snprintf(buf + n, bufsize - n, "%s", 8279 audio_encoding_name(fmt->encoding)); 8280 if (fmt->precision == fmt->stride) { 8281 n += snprintf(buf + n, bufsize - n, " %dbit", fmt->precision); 8282 } else { 8283 n += snprintf(buf + n, bufsize - n, " %d/%dbit", 8284 fmt->precision, fmt->stride); 8285 } 8286 8287 snprintf(buf + n, bufsize - n, " %uch %uHz", 8288 fmt->channels, fmt->sample_rate); 8289 } 8290 #endif 8291 8292 #if defined(AUDIO_DEBUG) 8293 static void 8294 audio_print_format2(const char *s, const audio_format2_t *fmt) 8295 { 8296 char fmtstr[64]; 8297 8298 audio_format2_tostr(fmtstr, sizeof(fmtstr), fmt); 8299 printf("%s %s\n", s, fmtstr); 8300 } 8301 #endif 8302 8303 #ifdef DIAGNOSTIC 8304 void 8305 audio_diagnostic_format2(const char *where, const audio_format2_t *fmt) 8306 { 8307 8308 KASSERTMSG(fmt, "called from %s", where); 8309 8310 /* XXX MSM6258 vs(4) only has 4bit stride format. */ 8311 if (fmt->encoding == AUDIO_ENCODING_ADPCM) { 8312 KASSERTMSG(fmt->stride == 4 || fmt->stride == 8, 8313 "called from %s: fmt->stride=%d", where, fmt->stride); 8314 } else { 8315 KASSERTMSG(fmt->stride % NBBY == 0, 8316 "called from %s: fmt->stride=%d", where, fmt->stride); 8317 } 8318 KASSERTMSG(fmt->precision <= fmt->stride, 8319 "called from %s: fmt->precision=%d fmt->stride=%d", 8320 where, fmt->precision, fmt->stride); 8321 KASSERTMSG(1 <= fmt->channels && fmt->channels <= AUDIO_MAX_CHANNELS, 8322 "called from %s: fmt->channels=%d", where, fmt->channels); 8323 8324 /* XXX No check for encodings? */ 8325 } 8326 8327 void 8328 audio_diagnostic_filter_arg(const char *where, const audio_filter_arg_t *arg) 8329 { 8330 8331 KASSERT(arg != NULL); 8332 KASSERT(arg->src != NULL); 8333 KASSERT(arg->dst != NULL); 8334 audio_diagnostic_format2(where, arg->srcfmt); 8335 audio_diagnostic_format2(where, arg->dstfmt); 8336 KASSERT(arg->count > 0); 8337 } 8338 8339 void 8340 audio_diagnostic_ring(const char *where, const audio_ring_t *ring) 8341 { 8342 8343 KASSERTMSG(ring, "called from %s", where); 8344 audio_diagnostic_format2(where, &ring->fmt); 8345 KASSERTMSG(0 <= ring->capacity && ring->capacity < INT_MAX / 2, 8346 "called from %s: ring->capacity=%d", where, ring->capacity); 8347 KASSERTMSG(0 <= ring->used && ring->used <= ring->capacity, 8348 "called from %s: ring->used=%d ring->capacity=%d", 8349 where, ring->used, ring->capacity); 8350 if (ring->capacity == 0) { 8351 KASSERTMSG(ring->mem == NULL, 8352 "called from %s: capacity == 0 but mem != NULL", where); 8353 } else { 8354 KASSERTMSG(ring->mem != NULL, 8355 "called from %s: capacity != 0 but mem == NULL", where); 8356 KASSERTMSG(0 <= ring->head && ring->head < ring->capacity, 8357 "called from %s: ring->head=%d ring->capacity=%d", 8358 where, ring->head, ring->capacity); 8359 } 8360 } 8361 #endif /* DIAGNOSTIC */ 8362 8363 8364 /* 8365 * Mixer driver 8366 */ 8367 8368 /* 8369 * Must be called without sc_lock held. 8370 */ 8371 int 8372 mixer_open(dev_t dev, struct audio_softc *sc, int flags, int ifmt, 8373 struct lwp *l) 8374 { 8375 struct file *fp; 8376 audio_file_t *af; 8377 int error, fd; 8378 8379 TRACE(1, "flags=0x%x", flags); 8380 8381 error = fd_allocfile(&fp, &fd); 8382 if (error) 8383 return error; 8384 8385 af = kmem_zalloc(sizeof(*af), KM_SLEEP); 8386 af->sc = sc; 8387 af->dev = dev; 8388 8389 mutex_enter(sc->sc_lock); 8390 if (sc->sc_dying) { 8391 mutex_exit(sc->sc_lock); 8392 kmem_free(af, sizeof(*af)); 8393 fd_abort(curproc, fp, fd); 8394 return ENXIO; 8395 } 8396 mutex_enter(sc->sc_intr_lock); 8397 SLIST_INSERT_HEAD(&sc->sc_files, af, entry); 8398 mutex_exit(sc->sc_intr_lock); 8399 mutex_exit(sc->sc_lock); 8400 8401 error = fd_clone(fp, fd, flags, &audio_fileops, af); 8402 KASSERT(error == EMOVEFD); 8403 8404 return error; 8405 } 8406 8407 /* 8408 * Add a process to those to be signalled on mixer activity. 8409 * If the process has already been added, do nothing. 8410 * Must be called with sc_exlock held and without sc_lock held. 8411 */ 8412 static void 8413 mixer_async_add(struct audio_softc *sc, pid_t pid) 8414 { 8415 int i; 8416 8417 KASSERT(sc->sc_exlock); 8418 8419 /* If already exists, returns without doing anything. */ 8420 for (i = 0; i < sc->sc_am_used; i++) { 8421 if (sc->sc_am[i] == pid) 8422 return; 8423 } 8424 8425 /* Extend array if necessary. */ 8426 if (sc->sc_am_used >= sc->sc_am_capacity) { 8427 sc->sc_am_capacity += AM_CAPACITY; 8428 sc->sc_am = kern_realloc(sc->sc_am, 8429 sc->sc_am_capacity * sizeof(pid_t), M_WAITOK); 8430 TRACE(2, "realloc am_capacity=%d", sc->sc_am_capacity); 8431 } 8432 8433 TRACE(2, "am[%d]=%d", sc->sc_am_used, (int)pid); 8434 sc->sc_am[sc->sc_am_used++] = pid; 8435 } 8436 8437 /* 8438 * Remove a process from those to be signalled on mixer activity. 8439 * If the process has not been added, do nothing. 8440 * Must be called with sc_exlock held and without sc_lock held. 8441 */ 8442 static void 8443 mixer_async_remove(struct audio_softc *sc, pid_t pid) 8444 { 8445 int i; 8446 8447 KASSERT(sc->sc_exlock); 8448 8449 for (i = 0; i < sc->sc_am_used; i++) { 8450 if (sc->sc_am[i] == pid) { 8451 sc->sc_am[i] = sc->sc_am[--sc->sc_am_used]; 8452 TRACE(2, "am[%d](%d) removed, used=%d", 8453 i, (int)pid, sc->sc_am_used); 8454 8455 /* Empty array if no longer necessary. */ 8456 if (sc->sc_am_used == 0) { 8457 kern_free(sc->sc_am); 8458 sc->sc_am = NULL; 8459 sc->sc_am_capacity = 0; 8460 TRACE(2, "released"); 8461 } 8462 return; 8463 } 8464 } 8465 } 8466 8467 /* 8468 * Signal all processes waiting for the mixer. 8469 * Must be called with sc_exlock held. 8470 */ 8471 static void 8472 mixer_signal(struct audio_softc *sc) 8473 { 8474 proc_t *p; 8475 int i; 8476 8477 KASSERT(sc->sc_exlock); 8478 8479 for (i = 0; i < sc->sc_am_used; i++) { 8480 mutex_enter(&proc_lock); 8481 p = proc_find(sc->sc_am[i]); 8482 if (p) 8483 psignal(p, SIGIO); 8484 mutex_exit(&proc_lock); 8485 } 8486 } 8487 8488 /* 8489 * Close a mixer device 8490 */ 8491 int 8492 mixer_close(struct audio_softc *sc, audio_file_t *file) 8493 { 8494 int error; 8495 8496 error = audio_exlock_enter(sc); 8497 if (error) 8498 return error; 8499 TRACE(1, "called"); 8500 mixer_async_remove(sc, curproc->p_pid); 8501 audio_exlock_exit(sc); 8502 8503 return 0; 8504 } 8505 8506 /* 8507 * Must be called without sc_lock nor sc_exlock held. 8508 */ 8509 int 8510 mixer_ioctl(struct audio_softc *sc, u_long cmd, void *addr, int flag, 8511 struct lwp *l) 8512 { 8513 mixer_devinfo_t *mi; 8514 mixer_ctrl_t *mc; 8515 int val; 8516 int error; 8517 8518 #if defined(AUDIO_DEBUG) 8519 char pre[64]; 8520 snprintf(pre, sizeof(pre), "pid=%d.%d", 8521 (int)curproc->p_pid, (int)l->l_lid); 8522 #endif 8523 error = EINVAL; 8524 8525 /* we can return cached values if we are sleeping */ 8526 if (cmd != AUDIO_MIXER_READ) { 8527 mutex_enter(sc->sc_lock); 8528 device_active(sc->sc_dev, DVA_SYSTEM); 8529 mutex_exit(sc->sc_lock); 8530 } 8531 8532 switch (cmd) { 8533 case FIOASYNC: 8534 val = *(int *)addr; 8535 TRACE(2, "%s FIOASYNC %s", pre, val ? "on" : "off"); 8536 error = audio_exlock_enter(sc); 8537 if (error) 8538 break; 8539 if (val) { 8540 mixer_async_add(sc, curproc->p_pid); 8541 } else { 8542 mixer_async_remove(sc, curproc->p_pid); 8543 } 8544 audio_exlock_exit(sc); 8545 break; 8546 8547 case AUDIO_GETDEV: 8548 TRACE(2, "%s AUDIO_GETDEV", pre); 8549 error = sc->hw_if->getdev(sc->hw_hdl, (audio_device_t *)addr); 8550 break; 8551 8552 case AUDIO_MIXER_DEVINFO: 8553 TRACE(2, "%s AUDIO_MIXER_DEVINFO", pre); 8554 mi = (mixer_devinfo_t *)addr; 8555 8556 mi->un.v.delta = 0; /* default */ 8557 mutex_enter(sc->sc_lock); 8558 error = audio_query_devinfo(sc, mi); 8559 mutex_exit(sc->sc_lock); 8560 break; 8561 8562 case AUDIO_MIXER_READ: 8563 TRACE(2, "%s AUDIO_MIXER_READ", pre); 8564 mc = (mixer_ctrl_t *)addr; 8565 8566 error = audio_exlock_mutex_enter(sc); 8567 if (error) 8568 break; 8569 if (device_is_active(sc->hw_dev)) 8570 error = audio_get_port(sc, mc); 8571 else if (mc->dev < 0 || mc->dev >= sc->sc_nmixer_states) 8572 error = ENXIO; 8573 else { 8574 int dev = mc->dev; 8575 memcpy(mc, &sc->sc_mixer_state[dev], 8576 sizeof(mixer_ctrl_t)); 8577 error = 0; 8578 } 8579 audio_exlock_mutex_exit(sc); 8580 break; 8581 8582 case AUDIO_MIXER_WRITE: 8583 TRACE(2, "%s AUDIO_MIXER_WRITE", pre); 8584 error = audio_exlock_mutex_enter(sc); 8585 if (error) 8586 break; 8587 error = audio_set_port(sc, (mixer_ctrl_t *)addr); 8588 if (error) { 8589 audio_exlock_mutex_exit(sc); 8590 break; 8591 } 8592 8593 if (sc->hw_if->commit_settings) { 8594 error = sc->hw_if->commit_settings(sc->hw_hdl); 8595 if (error) { 8596 audio_exlock_mutex_exit(sc); 8597 break; 8598 } 8599 } 8600 mutex_exit(sc->sc_lock); 8601 mixer_signal(sc); 8602 audio_exlock_exit(sc); 8603 break; 8604 8605 default: 8606 TRACE(2, "(%lu,'%c',%lu)", 8607 IOCPARM_LEN(cmd), (char)IOCGROUP(cmd), cmd & 0xff); 8608 if (sc->hw_if->dev_ioctl) { 8609 mutex_enter(sc->sc_lock); 8610 error = sc->hw_if->dev_ioctl(sc->hw_hdl, 8611 cmd, addr, flag, l); 8612 mutex_exit(sc->sc_lock); 8613 } else 8614 error = EINVAL; 8615 break; 8616 } 8617 8618 if (error) 8619 TRACE(2, "error=%d", error); 8620 return error; 8621 } 8622 8623 /* 8624 * Must be called with sc_lock held. 8625 */ 8626 int 8627 au_portof(struct audio_softc *sc, char *name, int class) 8628 { 8629 mixer_devinfo_t mi; 8630 8631 KASSERT(mutex_owned(sc->sc_lock)); 8632 8633 for (mi.index = 0; audio_query_devinfo(sc, &mi) == 0; mi.index++) { 8634 if (mi.mixer_class == class && strcmp(mi.label.name, name) == 0) 8635 return mi.index; 8636 } 8637 return -1; 8638 } 8639 8640 /* 8641 * Must be called with sc_lock held. 8642 */ 8643 void 8644 au_setup_ports(struct audio_softc *sc, struct au_mixer_ports *ports, 8645 mixer_devinfo_t *mi, const struct portname *tbl) 8646 { 8647 int i, j; 8648 8649 KASSERT(mutex_owned(sc->sc_lock)); 8650 8651 ports->index = mi->index; 8652 if (mi->type == AUDIO_MIXER_ENUM) { 8653 ports->isenum = true; 8654 for(i = 0; tbl[i].name; i++) 8655 for(j = 0; j < mi->un.e.num_mem; j++) 8656 if (strcmp(mi->un.e.member[j].label.name, 8657 tbl[i].name) == 0) { 8658 ports->allports |= tbl[i].mask; 8659 ports->aumask[ports->nports] = tbl[i].mask; 8660 ports->misel[ports->nports] = 8661 mi->un.e.member[j].ord; 8662 ports->miport[ports->nports] = 8663 au_portof(sc, mi->un.e.member[j].label.name, 8664 mi->mixer_class); 8665 if (ports->mixerout != -1 && 8666 ports->miport[ports->nports] != -1) 8667 ports->isdual = true; 8668 ++ports->nports; 8669 } 8670 } else if (mi->type == AUDIO_MIXER_SET) { 8671 for(i = 0; tbl[i].name; i++) 8672 for(j = 0; j < mi->un.s.num_mem; j++) 8673 if (strcmp(mi->un.s.member[j].label.name, 8674 tbl[i].name) == 0) { 8675 ports->allports |= tbl[i].mask; 8676 ports->aumask[ports->nports] = tbl[i].mask; 8677 ports->misel[ports->nports] = 8678 mi->un.s.member[j].mask; 8679 ports->miport[ports->nports] = 8680 au_portof(sc, mi->un.s.member[j].label.name, 8681 mi->mixer_class); 8682 ++ports->nports; 8683 } 8684 } 8685 } 8686 8687 /* 8688 * Must be called with sc_lock && sc_exlock held. 8689 */ 8690 int 8691 au_set_lr_value(struct audio_softc *sc, mixer_ctrl_t *ct, int l, int r) 8692 { 8693 8694 KASSERT(mutex_owned(sc->sc_lock)); 8695 KASSERT(sc->sc_exlock); 8696 8697 ct->type = AUDIO_MIXER_VALUE; 8698 ct->un.value.num_channels = 2; 8699 ct->un.value.level[AUDIO_MIXER_LEVEL_LEFT] = l; 8700 ct->un.value.level[AUDIO_MIXER_LEVEL_RIGHT] = r; 8701 if (audio_set_port(sc, ct) == 0) 8702 return 0; 8703 ct->un.value.num_channels = 1; 8704 ct->un.value.level[AUDIO_MIXER_LEVEL_MONO] = (l+r)/2; 8705 return audio_set_port(sc, ct); 8706 } 8707 8708 /* 8709 * Must be called with sc_lock && sc_exlock held. 8710 */ 8711 int 8712 au_get_lr_value(struct audio_softc *sc, mixer_ctrl_t *ct, int *l, int *r) 8713 { 8714 int error; 8715 8716 KASSERT(mutex_owned(sc->sc_lock)); 8717 KASSERT(sc->sc_exlock); 8718 8719 ct->un.value.num_channels = 2; 8720 if (audio_get_port(sc, ct) == 0) { 8721 *l = ct->un.value.level[AUDIO_MIXER_LEVEL_LEFT]; 8722 *r = ct->un.value.level[AUDIO_MIXER_LEVEL_RIGHT]; 8723 } else { 8724 ct->un.value.num_channels = 1; 8725 error = audio_get_port(sc, ct); 8726 if (error) 8727 return error; 8728 *r = *l = ct->un.value.level[AUDIO_MIXER_LEVEL_MONO]; 8729 } 8730 return 0; 8731 } 8732 8733 /* 8734 * Must be called with sc_lock && sc_exlock held. 8735 */ 8736 int 8737 au_set_gain(struct audio_softc *sc, struct au_mixer_ports *ports, 8738 int gain, int balance) 8739 { 8740 mixer_ctrl_t ct; 8741 int i, error; 8742 int l, r; 8743 u_int mask; 8744 int nset; 8745 8746 KASSERT(mutex_owned(sc->sc_lock)); 8747 KASSERT(sc->sc_exlock); 8748 8749 if (balance == AUDIO_MID_BALANCE) { 8750 l = r = gain; 8751 } else if (balance < AUDIO_MID_BALANCE) { 8752 l = gain; 8753 r = (balance * gain) / AUDIO_MID_BALANCE; 8754 } else { 8755 r = gain; 8756 l = ((AUDIO_RIGHT_BALANCE - balance) * gain) 8757 / AUDIO_MID_BALANCE; 8758 } 8759 TRACE(2, "gain=%d balance=%d, l=%d r=%d", gain, balance, l, r); 8760 8761 if (ports->index == -1) { 8762 usemaster: 8763 if (ports->master == -1) 8764 return 0; /* just ignore it silently */ 8765 ct.dev = ports->master; 8766 error = au_set_lr_value(sc, &ct, l, r); 8767 } else { 8768 ct.dev = ports->index; 8769 if (ports->isenum) { 8770 ct.type = AUDIO_MIXER_ENUM; 8771 error = audio_get_port(sc, &ct); 8772 if (error) 8773 return error; 8774 if (ports->isdual) { 8775 if (ports->cur_port == -1) 8776 ct.dev = ports->master; 8777 else 8778 ct.dev = ports->miport[ports->cur_port]; 8779 error = au_set_lr_value(sc, &ct, l, r); 8780 } else { 8781 for(i = 0; i < ports->nports; i++) 8782 if (ports->misel[i] == ct.un.ord) { 8783 ct.dev = ports->miport[i]; 8784 if (ct.dev == -1 || 8785 au_set_lr_value(sc, &ct, l, r)) 8786 goto usemaster; 8787 else 8788 break; 8789 } 8790 } 8791 } else { 8792 ct.type = AUDIO_MIXER_SET; 8793 error = audio_get_port(sc, &ct); 8794 if (error) 8795 return error; 8796 mask = ct.un.mask; 8797 nset = 0; 8798 for(i = 0; i < ports->nports; i++) { 8799 if (ports->misel[i] & mask) { 8800 ct.dev = ports->miport[i]; 8801 if (ct.dev != -1 && 8802 au_set_lr_value(sc, &ct, l, r) == 0) 8803 nset++; 8804 } 8805 } 8806 if (nset == 0) 8807 goto usemaster; 8808 } 8809 } 8810 if (!error) 8811 mixer_signal(sc); 8812 return error; 8813 } 8814 8815 /* 8816 * Must be called with sc_lock && sc_exlock held. 8817 */ 8818 void 8819 au_get_gain(struct audio_softc *sc, struct au_mixer_ports *ports, 8820 u_int *pgain, u_char *pbalance) 8821 { 8822 mixer_ctrl_t ct; 8823 int i, l, r, n; 8824 int lgain, rgain; 8825 8826 KASSERT(mutex_owned(sc->sc_lock)); 8827 KASSERT(sc->sc_exlock); 8828 8829 lgain = AUDIO_MAX_GAIN / 2; 8830 rgain = AUDIO_MAX_GAIN / 2; 8831 if (ports->index == -1) { 8832 usemaster: 8833 if (ports->master == -1) 8834 goto bad; 8835 ct.dev = ports->master; 8836 ct.type = AUDIO_MIXER_VALUE; 8837 if (au_get_lr_value(sc, &ct, &lgain, &rgain)) 8838 goto bad; 8839 } else { 8840 ct.dev = ports->index; 8841 if (ports->isenum) { 8842 ct.type = AUDIO_MIXER_ENUM; 8843 if (audio_get_port(sc, &ct)) 8844 goto bad; 8845 ct.type = AUDIO_MIXER_VALUE; 8846 if (ports->isdual) { 8847 if (ports->cur_port == -1) 8848 ct.dev = ports->master; 8849 else 8850 ct.dev = ports->miport[ports->cur_port]; 8851 au_get_lr_value(sc, &ct, &lgain, &rgain); 8852 } else { 8853 for(i = 0; i < ports->nports; i++) 8854 if (ports->misel[i] == ct.un.ord) { 8855 ct.dev = ports->miport[i]; 8856 if (ct.dev == -1 || 8857 au_get_lr_value(sc, &ct, 8858 &lgain, &rgain)) 8859 goto usemaster; 8860 else 8861 break; 8862 } 8863 } 8864 } else { 8865 ct.type = AUDIO_MIXER_SET; 8866 if (audio_get_port(sc, &ct)) 8867 goto bad; 8868 ct.type = AUDIO_MIXER_VALUE; 8869 lgain = rgain = n = 0; 8870 for(i = 0; i < ports->nports; i++) { 8871 if (ports->misel[i] & ct.un.mask) { 8872 ct.dev = ports->miport[i]; 8873 if (ct.dev == -1 || 8874 au_get_lr_value(sc, &ct, &l, &r)) 8875 goto usemaster; 8876 else { 8877 lgain += l; 8878 rgain += r; 8879 n++; 8880 } 8881 } 8882 } 8883 if (n != 0) { 8884 lgain /= n; 8885 rgain /= n; 8886 } 8887 } 8888 } 8889 bad: 8890 if (lgain == rgain) { /* handles lgain==rgain==0 */ 8891 *pgain = lgain; 8892 *pbalance = AUDIO_MID_BALANCE; 8893 } else if (lgain < rgain) { 8894 *pgain = rgain; 8895 /* balance should be > AUDIO_MID_BALANCE */ 8896 *pbalance = AUDIO_RIGHT_BALANCE - 8897 (AUDIO_MID_BALANCE * lgain) / rgain; 8898 } else /* lgain > rgain */ { 8899 *pgain = lgain; 8900 /* balance should be < AUDIO_MID_BALANCE */ 8901 *pbalance = (AUDIO_MID_BALANCE * rgain) / lgain; 8902 } 8903 } 8904 8905 /* 8906 * Must be called with sc_lock && sc_exlock held. 8907 */ 8908 int 8909 au_set_port(struct audio_softc *sc, struct au_mixer_ports *ports, u_int port) 8910 { 8911 mixer_ctrl_t ct; 8912 int i, error, use_mixerout; 8913 8914 KASSERT(mutex_owned(sc->sc_lock)); 8915 KASSERT(sc->sc_exlock); 8916 8917 use_mixerout = 1; 8918 if (port == 0) { 8919 if (ports->allports == 0) 8920 return 0; /* Allow this special case. */ 8921 else if (ports->isdual) { 8922 if (ports->cur_port == -1) { 8923 return 0; 8924 } else { 8925 port = ports->aumask[ports->cur_port]; 8926 ports->cur_port = -1; 8927 use_mixerout = 0; 8928 } 8929 } 8930 } 8931 if (ports->index == -1) 8932 return EINVAL; 8933 ct.dev = ports->index; 8934 if (ports->isenum) { 8935 if (port & (port-1)) 8936 return EINVAL; /* Only one port allowed */ 8937 ct.type = AUDIO_MIXER_ENUM; 8938 error = EINVAL; 8939 for(i = 0; i < ports->nports; i++) 8940 if (ports->aumask[i] == port) { 8941 if (ports->isdual && use_mixerout) { 8942 ct.un.ord = ports->mixerout; 8943 ports->cur_port = i; 8944 } else { 8945 ct.un.ord = ports->misel[i]; 8946 } 8947 error = audio_set_port(sc, &ct); 8948 break; 8949 } 8950 } else { 8951 ct.type = AUDIO_MIXER_SET; 8952 ct.un.mask = 0; 8953 for(i = 0; i < ports->nports; i++) 8954 if (ports->aumask[i] & port) 8955 ct.un.mask |= ports->misel[i]; 8956 if (port != 0 && ct.un.mask == 0) 8957 error = EINVAL; 8958 else 8959 error = audio_set_port(sc, &ct); 8960 } 8961 if (!error) 8962 mixer_signal(sc); 8963 return error; 8964 } 8965 8966 /* 8967 * Must be called with sc_lock && sc_exlock held. 8968 */ 8969 int 8970 au_get_port(struct audio_softc *sc, struct au_mixer_ports *ports) 8971 { 8972 mixer_ctrl_t ct; 8973 int i, aumask; 8974 8975 KASSERT(mutex_owned(sc->sc_lock)); 8976 KASSERT(sc->sc_exlock); 8977 8978 if (ports->index == -1) 8979 return 0; 8980 ct.dev = ports->index; 8981 ct.type = ports->isenum ? AUDIO_MIXER_ENUM : AUDIO_MIXER_SET; 8982 if (audio_get_port(sc, &ct)) 8983 return 0; 8984 aumask = 0; 8985 if (ports->isenum) { 8986 if (ports->isdual && ports->cur_port != -1) { 8987 if (ports->mixerout == ct.un.ord) 8988 aumask = ports->aumask[ports->cur_port]; 8989 else 8990 ports->cur_port = -1; 8991 } 8992 if (aumask == 0) 8993 for(i = 0; i < ports->nports; i++) 8994 if (ports->misel[i] == ct.un.ord) 8995 aumask = ports->aumask[i]; 8996 } else { 8997 for(i = 0; i < ports->nports; i++) 8998 if (ct.un.mask & ports->misel[i]) 8999 aumask |= ports->aumask[i]; 9000 } 9001 return aumask; 9002 } 9003 9004 /* 9005 * It returns 0 if success, otherwise errno. 9006 * Must be called only if sc->sc_monitor_port != -1. 9007 * Must be called with sc_lock && sc_exlock held. 9008 */ 9009 static int 9010 au_set_monitor_gain(struct audio_softc *sc, int monitor_gain) 9011 { 9012 mixer_ctrl_t ct; 9013 9014 KASSERT(mutex_owned(sc->sc_lock)); 9015 KASSERT(sc->sc_exlock); 9016 9017 ct.dev = sc->sc_monitor_port; 9018 ct.type = AUDIO_MIXER_VALUE; 9019 ct.un.value.num_channels = 1; 9020 ct.un.value.level[AUDIO_MIXER_LEVEL_MONO] = monitor_gain; 9021 return audio_set_port(sc, &ct); 9022 } 9023 9024 /* 9025 * It returns monitor gain if success, otherwise -1. 9026 * Must be called only if sc->sc_monitor_port != -1. 9027 * Must be called with sc_lock && sc_exlock held. 9028 */ 9029 static int 9030 au_get_monitor_gain(struct audio_softc *sc) 9031 { 9032 mixer_ctrl_t ct; 9033 9034 KASSERT(mutex_owned(sc->sc_lock)); 9035 KASSERT(sc->sc_exlock); 9036 9037 ct.dev = sc->sc_monitor_port; 9038 ct.type = AUDIO_MIXER_VALUE; 9039 ct.un.value.num_channels = 1; 9040 if (audio_get_port(sc, &ct)) 9041 return -1; 9042 return ct.un.value.level[AUDIO_MIXER_LEVEL_MONO]; 9043 } 9044 9045 /* 9046 * Must be called with sc_lock && sc_exlock held. 9047 */ 9048 static int 9049 audio_set_port(struct audio_softc *sc, mixer_ctrl_t *mc) 9050 { 9051 9052 KASSERT(mutex_owned(sc->sc_lock)); 9053 KASSERT(sc->sc_exlock); 9054 9055 return sc->hw_if->set_port(sc->hw_hdl, mc); 9056 } 9057 9058 /* 9059 * Must be called with sc_lock && sc_exlock held. 9060 */ 9061 static int 9062 audio_get_port(struct audio_softc *sc, mixer_ctrl_t *mc) 9063 { 9064 9065 KASSERT(mutex_owned(sc->sc_lock)); 9066 KASSERT(sc->sc_exlock); 9067 9068 return sc->hw_if->get_port(sc->hw_hdl, mc); 9069 } 9070 9071 /* 9072 * Must be called with sc_lock && sc_exlock held. 9073 */ 9074 static void 9075 audio_mixer_capture(struct audio_softc *sc) 9076 { 9077 mixer_devinfo_t mi; 9078 mixer_ctrl_t *mc; 9079 9080 KASSERT(mutex_owned(sc->sc_lock)); 9081 KASSERT(sc->sc_exlock); 9082 9083 for (mi.index = 0;; mi.index++) { 9084 if (audio_query_devinfo(sc, &mi) != 0) 9085 break; 9086 KASSERT(mi.index < sc->sc_nmixer_states); 9087 if (mi.type == AUDIO_MIXER_CLASS) 9088 continue; 9089 mc = &sc->sc_mixer_state[mi.index]; 9090 mc->dev = mi.index; 9091 mc->type = mi.type; 9092 mc->un.value.num_channels = mi.un.v.num_channels; 9093 (void)audio_get_port(sc, mc); 9094 } 9095 9096 return; 9097 } 9098 9099 /* 9100 * Must be called with sc_lock && sc_exlock held. 9101 */ 9102 static void 9103 audio_mixer_restore(struct audio_softc *sc) 9104 { 9105 mixer_devinfo_t mi; 9106 mixer_ctrl_t *mc; 9107 9108 KASSERT(mutex_owned(sc->sc_lock)); 9109 KASSERT(sc->sc_exlock); 9110 9111 for (mi.index = 0; ; mi.index++) { 9112 if (audio_query_devinfo(sc, &mi) != 0) 9113 break; 9114 if (mi.type == AUDIO_MIXER_CLASS) 9115 continue; 9116 mc = &sc->sc_mixer_state[mi.index]; 9117 (void)audio_set_port(sc, mc); 9118 } 9119 if (sc->hw_if->commit_settings) 9120 sc->hw_if->commit_settings(sc->hw_hdl); 9121 9122 return; 9123 } 9124 9125 static void 9126 audio_volume_down(device_t dv) 9127 { 9128 struct audio_softc *sc = device_private(dv); 9129 mixer_devinfo_t mi; 9130 int newgain; 9131 u_int gain; 9132 u_char balance; 9133 9134 if (audio_exlock_mutex_enter(sc) != 0) 9135 return; 9136 if (sc->sc_outports.index == -1 && sc->sc_outports.master != -1) { 9137 mi.index = sc->sc_outports.master; 9138 mi.un.v.delta = 0; 9139 if (audio_query_devinfo(sc, &mi) == 0) { 9140 au_get_gain(sc, &sc->sc_outports, &gain, &balance); 9141 newgain = gain - mi.un.v.delta; 9142 if (newgain < AUDIO_MIN_GAIN) 9143 newgain = AUDIO_MIN_GAIN; 9144 au_set_gain(sc, &sc->sc_outports, newgain, balance); 9145 } 9146 } 9147 audio_exlock_mutex_exit(sc); 9148 } 9149 9150 static void 9151 audio_volume_up(device_t dv) 9152 { 9153 struct audio_softc *sc = device_private(dv); 9154 mixer_devinfo_t mi; 9155 u_int gain, newgain; 9156 u_char balance; 9157 9158 if (audio_exlock_mutex_enter(sc) != 0) 9159 return; 9160 if (sc->sc_outports.index == -1 && sc->sc_outports.master != -1) { 9161 mi.index = sc->sc_outports.master; 9162 mi.un.v.delta = 0; 9163 if (audio_query_devinfo(sc, &mi) == 0) { 9164 au_get_gain(sc, &sc->sc_outports, &gain, &balance); 9165 newgain = gain + mi.un.v.delta; 9166 if (newgain > AUDIO_MAX_GAIN) 9167 newgain = AUDIO_MAX_GAIN; 9168 au_set_gain(sc, &sc->sc_outports, newgain, balance); 9169 } 9170 } 9171 audio_exlock_mutex_exit(sc); 9172 } 9173 9174 static void 9175 audio_volume_toggle(device_t dv) 9176 { 9177 struct audio_softc *sc = device_private(dv); 9178 u_int gain, newgain; 9179 u_char balance; 9180 9181 if (audio_exlock_mutex_enter(sc) != 0) 9182 return; 9183 au_get_gain(sc, &sc->sc_outports, &gain, &balance); 9184 if (gain != 0) { 9185 sc->sc_lastgain = gain; 9186 newgain = 0; 9187 } else 9188 newgain = sc->sc_lastgain; 9189 au_set_gain(sc, &sc->sc_outports, newgain, balance); 9190 audio_exlock_mutex_exit(sc); 9191 } 9192 9193 /* 9194 * Must be called with sc_lock held. 9195 */ 9196 static int 9197 audio_query_devinfo(struct audio_softc *sc, mixer_devinfo_t *di) 9198 { 9199 9200 KASSERT(mutex_owned(sc->sc_lock)); 9201 9202 return sc->hw_if->query_devinfo(sc->hw_hdl, di); 9203 } 9204 9205 void 9206 audio_mixsample_to_linear(audio_filter_arg_t *arg) 9207 { 9208 const audio_format2_t *fmt; 9209 const aint2_t *m; 9210 uint8_t *p; 9211 u_int sample_count; 9212 bool swap; 9213 aint2_t v, xor; 9214 u_int i, bps; 9215 bool little; 9216 9217 DIAGNOSTIC_filter_arg(arg); 9218 KASSERT(audio_format2_is_linear(arg->dstfmt)); 9219 KASSERT(arg->srcfmt->channels == arg->dstfmt->channels); 9220 9221 fmt = arg->dstfmt; 9222 m = arg->src; 9223 p = arg->dst; 9224 sample_count = arg->count * fmt->channels; 9225 swap = arg->dstfmt->encoding == AUDIO_ENCODING_SLINEAR_OE; 9226 9227 #if BYTE_ORDER == LITTLE_ENDIAN 9228 little = !swap; 9229 #endif 9230 #if BYTE_ORDER == BIG_ENDIAN 9231 little = swap; 9232 #endif 9233 9234 bps = fmt->stride / NBBY; 9235 9236 xor = audio_format2_is_signed(fmt) 9237 ? 0 : 1 << (fmt->stride - 1); 9238 9239 for (i=0; i<sample_count; ++i) { 9240 v = *m++; 9241 9242 /* scale up to 32bit and then down to target size */ 9243 v <<= 32 - AUDIO_INTERNAL_BITS; 9244 v >>= (4 - bps) * NBBY; 9245 9246 /* signed -> unsigned */ 9247 v ^= xor; 9248 9249 if (little) { 9250 switch (bps) { 9251 case 4: *p++ = v; v >>= 8; /* FALLTHROUGH */ 9252 case 3: *p++ = v; v >>= 8; /* FALLTHROUGH */ 9253 case 2: *p++ = v; v >>= 8; /* FALLTHROUGH */ 9254 case 1: *p++ = v; /* FALLTHROUGH */ 9255 } 9256 } else { 9257 switch (bps) { 9258 case 4: *p++ = v >> 24; v <<= 8; /* FALLTHROUGH */ 9259 case 3: *p++ = v >> 24; v <<= 8; /* FALLTHROUGH */ 9260 case 2: *p++ = v >> 24; v <<= 8; /* FALLTHROUGH */ 9261 case 1: *p++ = v >> 24; /* FALLTHROUGH */ 9262 } 9263 } 9264 } 9265 } 9266 9267 9268 #endif /* NAUDIO > 0 */ 9269 9270 #if NAUDIO == 0 && (NMIDI > 0 || NMIDIBUS > 0) 9271 #include <sys/param.h> 9272 #include <sys/systm.h> 9273 #include <sys/device.h> 9274 #include <sys/audioio.h> 9275 #include <dev/audio/audio_if.h> 9276 #endif 9277 9278 #if NAUDIO > 0 || (NMIDI > 0 || NMIDIBUS > 0) 9279 int 9280 audioprint(void *aux, const char *pnp) 9281 { 9282 struct audio_attach_args *arg; 9283 const char *type; 9284 9285 if (pnp != NULL) { 9286 arg = aux; 9287 switch (arg->type) { 9288 case AUDIODEV_TYPE_AUDIO: 9289 type = "audio"; 9290 break; 9291 case AUDIODEV_TYPE_MIDI: 9292 type = "midi"; 9293 break; 9294 case AUDIODEV_TYPE_OPL: 9295 type = "opl"; 9296 break; 9297 case AUDIODEV_TYPE_MPU: 9298 type = "mpu"; 9299 break; 9300 case AUDIODEV_TYPE_AUX: 9301 type = "aux"; 9302 break; 9303 default: 9304 panic("audioprint: unknown type %d", arg->type); 9305 } 9306 aprint_normal("%s at %s", type, pnp); 9307 } 9308 return UNCONF; 9309 } 9310 9311 #endif /* NAUDIO > 0 || (NMIDI > 0 || NMIDIBUS > 0) */ 9312 9313 #ifdef _MODULE 9314 9315 devmajor_t audio_bmajor = -1, audio_cmajor = -1; 9316 9317 #include "ioconf.c" 9318 9319 #endif 9320 9321 MODULE(MODULE_CLASS_DRIVER, audio, NULL); 9322 9323 static int 9324 audio_modcmd(modcmd_t cmd, void *arg) 9325 { 9326 int error = 0; 9327 9328 switch (cmd) { 9329 case MODULE_CMD_INIT: 9330 /* XXX interrupt level? */ 9331 audio_psref_class = psref_class_create("audio", IPL_SOFTSERIAL); 9332 #ifdef _MODULE 9333 error = devsw_attach(audio_cd.cd_name, NULL, &audio_bmajor, 9334 &audio_cdevsw, &audio_cmajor); 9335 if (error) 9336 break; 9337 9338 error = config_init_component(cfdriver_ioconf_audio, 9339 cfattach_ioconf_audio, cfdata_ioconf_audio); 9340 if (error) { 9341 devsw_detach(NULL, &audio_cdevsw); 9342 } 9343 #endif 9344 break; 9345 case MODULE_CMD_FINI: 9346 #ifdef _MODULE 9347 error = config_fini_component(cfdriver_ioconf_audio, 9348 cfattach_ioconf_audio, cfdata_ioconf_audio); 9349 if (error == 0) 9350 devsw_detach(NULL, &audio_cdevsw); 9351 #endif 9352 if (error == 0) 9353 psref_class_destroy(audio_psref_class); 9354 break; 9355 default: 9356 error = ENOTTY; 9357 break; 9358 } 9359 9360 return error; 9361 } 9362
Indexes created Mon Sep 29 21:09:56 GMT 2025