audio.c revision 1.132
1 /* $NetBSD: audio.c,v 1.132 2022/04/23 11:30:57 isaki Exp $ */ 2 3 /*- 4 * Copyright (c) 2008 The NetBSD Foundation, Inc. 5 * All rights reserved. 6 * 7 * This code is derived from software contributed to The NetBSD Foundation 8 * by Andrew Doran. 9 * 10 * Redistribution and use in source and binary forms, with or without 11 * modification, are permitted provided that the following conditions 12 * are met: 13 * 1. Redistributions of source code must retain the above copyright 14 * notice, this list of conditions and the following disclaimer. 15 * 2. Redistributions in binary form must reproduce the above copyright 16 * notice, this list of conditions and the following disclaimer in the 17 * documentation and/or other materials provided with the distribution. 18 * 19 * THIS SOFTWARE IS PROVIDED BY THE NETBSD FOUNDATION, INC. AND CONTRIBUTORS 20 * ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED 21 * TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR 22 * PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE FOUNDATION OR CONTRIBUTORS 23 * BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR 24 * CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF 25 * SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS 26 * INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN 27 * CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) 28 * ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE 29 * POSSIBILITY OF SUCH DAMAGE. 30 */ 31 32 /* 33 * Copyright (c) 1991-1993 Regents of the University of California. 34 * All rights reserved. 35 * 36 * Redistribution and use in source and binary forms, with or without 37 * modification, are permitted provided that the following conditions 38 * are met: 39 * 1. Redistributions of source code must retain the above copyright 40 * notice, this list of conditions and the following disclaimer. 41 * 2. Redistributions in binary form must reproduce the above copyright 42 * notice, this list of conditions and the following disclaimer in the 43 * documentation and/or other materials provided with the distribution. 44 * 3. All advertising materials mentioning features or use of this software 45 * must display the following acknowledgement: 46 * This product includes software developed by the Computer Systems 47 * Engineering Group at Lawrence Berkeley Laboratory. 48 * 4. Neither the name of the University nor of the Laboratory may be used 49 * to endorse or promote products derived from this software without 50 * specific prior written permission. 51 * 52 * THIS SOFTWARE IS PROVIDED BY THE REGENTS AND CONTRIBUTORS ``AS IS'' AND 53 * ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE 54 * IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE 55 * ARE DISCLAIMED. IN NO EVENT SHALL THE REGENTS OR CONTRIBUTORS BE LIABLE 56 * FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL 57 * DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS 58 * OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) 59 * HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT 60 * LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY 61 * OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF 62 * SUCH DAMAGE. 63 */ 64 65 /* 66 * 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 1.132 2022/04/23 11:30:57 isaki 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... 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 void audio_printf(struct audio_softc *, const char *, ...) 565 __printflike(2, 3); 566 static int audio_exlock_mutex_enter(struct audio_softc *); 567 static void audio_exlock_mutex_exit(struct audio_softc *); 568 static int audio_exlock_enter(struct audio_softc *); 569 static void audio_exlock_exit(struct audio_softc *); 570 static struct audio_softc *audio_sc_acquire_fromfile(audio_file_t *, 571 struct psref *); 572 static void audio_sc_release(struct audio_softc *, struct psref *); 573 static int audio_track_waitio(struct audio_softc *, audio_track_t *); 574 575 static int audioclose(struct file *); 576 static int audioread(struct file *, off_t *, struct uio *, kauth_cred_t, int); 577 static int audiowrite(struct file *, off_t *, struct uio *, kauth_cred_t, int); 578 static int audioioctl(struct file *, u_long, void *); 579 static int audiopoll(struct file *, int); 580 static int audiokqfilter(struct file *, struct knote *); 581 static int audiommap(struct file *, off_t *, size_t, int, int *, int *, 582 struct uvm_object **, int *); 583 static int audiostat(struct file *, struct stat *); 584 585 static void filt_audiowrite_detach(struct knote *); 586 static int filt_audiowrite_event(struct knote *, long); 587 static void filt_audioread_detach(struct knote *); 588 static int filt_audioread_event(struct knote *, long); 589 590 static int audio_open(dev_t, struct audio_softc *, int, int, struct lwp *, 591 audio_file_t **); 592 static int audio_close(struct audio_softc *, audio_file_t *); 593 static void audio_unlink(struct audio_softc *, audio_file_t *); 594 static int audio_read(struct audio_softc *, struct uio *, int, audio_file_t *); 595 static int audio_write(struct audio_softc *, struct uio *, int, audio_file_t *); 596 static void audio_file_clear(struct audio_softc *, audio_file_t *); 597 static int audio_ioctl(dev_t, struct audio_softc *, u_long, void *, int, 598 struct lwp *, audio_file_t *); 599 static int audio_poll(struct audio_softc *, int, struct lwp *, audio_file_t *); 600 static int audio_kqfilter(struct audio_softc *, audio_file_t *, struct knote *); 601 static int audio_mmap(struct audio_softc *, off_t *, size_t, int, int *, int *, 602 struct uvm_object **, int *, audio_file_t *); 603 604 static int audioctl_open(dev_t, struct audio_softc *, int, int, struct lwp *); 605 606 static void audio_pintr(void *); 607 static void audio_rintr(void *); 608 609 static int audio_query_devinfo(struct audio_softc *, mixer_devinfo_t *); 610 611 static int audio_track_inputblk_as_usrbyte(const audio_track_t *, int); 612 static int audio_track_readablebytes(const audio_track_t *); 613 static int audio_file_setinfo(struct audio_softc *, audio_file_t *, 614 const struct audio_info *); 615 static int audio_track_setinfo_check(audio_track_t *, 616 audio_format2_t *, const struct audio_prinfo *); 617 static void audio_track_setinfo_water(audio_track_t *, 618 const struct audio_info *); 619 static int audio_hw_setinfo(struct audio_softc *, const struct audio_info *, 620 struct audio_info *); 621 static int audio_hw_set_format(struct audio_softc *, int, 622 const audio_format2_t *, const audio_format2_t *, 623 audio_filter_reg_t *, audio_filter_reg_t *); 624 static int audiogetinfo(struct audio_softc *, struct audio_info *, int, 625 audio_file_t *); 626 static bool audio_can_playback(struct audio_softc *); 627 static bool audio_can_capture(struct audio_softc *); 628 static int audio_check_params(audio_format2_t *); 629 static int audio_mixers_init(struct audio_softc *sc, int, 630 const audio_format2_t *, const audio_format2_t *, 631 const audio_filter_reg_t *, const audio_filter_reg_t *); 632 static int audio_select_freq(const struct audio_format *); 633 static int audio_hw_probe(struct audio_softc *, audio_format2_t *, int); 634 static int audio_hw_validate_format(struct audio_softc *, int, 635 const audio_format2_t *); 636 static int audio_mixers_set_format(struct audio_softc *, 637 const struct audio_info *); 638 static void audio_mixers_get_format(struct audio_softc *, struct audio_info *); 639 static int audio_sysctl_blk_ms(SYSCTLFN_PROTO); 640 static int audio_sysctl_multiuser(SYSCTLFN_PROTO); 641 #if defined(AUDIO_DEBUG) 642 static int audio_sysctl_debug(SYSCTLFN_PROTO); 643 static void audio_format2_tostr(char *, size_t, const audio_format2_t *); 644 static void audio_print_format2(const char *, const audio_format2_t *) __unused; 645 #endif 646 647 static void *audio_realloc(void *, size_t); 648 static int audio_realloc_usrbuf(audio_track_t *, int); 649 static void audio_free_usrbuf(audio_track_t *); 650 651 static audio_track_t *audio_track_create(struct audio_softc *, 652 audio_trackmixer_t *); 653 static void audio_track_destroy(audio_track_t *); 654 static audio_filter_t audio_track_get_codec(audio_track_t *, 655 const audio_format2_t *, const audio_format2_t *); 656 static int audio_track_set_format(audio_track_t *, audio_format2_t *); 657 static void audio_track_play(audio_track_t *); 658 static int audio_track_drain(struct audio_softc *, audio_track_t *); 659 static void audio_track_record(audio_track_t *); 660 static void audio_track_clear(struct audio_softc *, audio_track_t *); 661 662 static int audio_mixer_init(struct audio_softc *, int, 663 const audio_format2_t *, const audio_filter_reg_t *); 664 static void audio_mixer_destroy(struct audio_softc *, audio_trackmixer_t *); 665 static void audio_pmixer_start(struct audio_softc *, bool); 666 static void audio_pmixer_process(struct audio_softc *); 667 static void audio_pmixer_agc(audio_trackmixer_t *, int); 668 static int audio_pmixer_mix_track(audio_trackmixer_t *, audio_track_t *, int); 669 static void audio_pmixer_output(struct audio_softc *); 670 static int audio_pmixer_halt(struct audio_softc *); 671 static void audio_rmixer_start(struct audio_softc *); 672 static void audio_rmixer_process(struct audio_softc *); 673 static void audio_rmixer_input(struct audio_softc *); 674 static int audio_rmixer_halt(struct audio_softc *); 675 676 static void mixer_init(struct audio_softc *); 677 static int mixer_open(dev_t, struct audio_softc *, int, int, struct lwp *); 678 static int mixer_close(struct audio_softc *, audio_file_t *); 679 static int mixer_ioctl(struct audio_softc *, u_long, void *, int, struct lwp *); 680 static void mixer_async_add(struct audio_softc *, pid_t); 681 static void mixer_async_remove(struct audio_softc *, pid_t); 682 static void mixer_signal(struct audio_softc *); 683 684 static int au_portof(struct audio_softc *, char *, int); 685 686 static void au_setup_ports(struct audio_softc *, struct au_mixer_ports *, 687 mixer_devinfo_t *, const struct portname *); 688 static int au_set_lr_value(struct audio_softc *, mixer_ctrl_t *, int, int); 689 static int au_get_lr_value(struct audio_softc *, mixer_ctrl_t *, int *, int *); 690 static int au_set_gain(struct audio_softc *, struct au_mixer_ports *, int, int); 691 static void au_get_gain(struct audio_softc *, struct au_mixer_ports *, 692 u_int *, u_char *); 693 static int au_set_port(struct audio_softc *, struct au_mixer_ports *, u_int); 694 static int au_get_port(struct audio_softc *, struct au_mixer_ports *); 695 static int au_set_monitor_gain(struct audio_softc *, int); 696 static int au_get_monitor_gain(struct audio_softc *); 697 static int audio_get_port(struct audio_softc *, mixer_ctrl_t *); 698 static int audio_set_port(struct audio_softc *, mixer_ctrl_t *); 699 700 static __inline struct audio_params 701 format2_to_params(const audio_format2_t *f2) 702 { 703 audio_params_t p; 704 705 /* validbits/precision <-> precision/stride */ 706 p.sample_rate = f2->sample_rate; 707 p.channels = f2->channels; 708 p.encoding = f2->encoding; 709 p.validbits = f2->precision; 710 p.precision = f2->stride; 711 return p; 712 } 713 714 static __inline audio_format2_t 715 params_to_format2(const struct audio_params *p) 716 { 717 audio_format2_t f2; 718 719 /* precision/stride <-> validbits/precision */ 720 f2.sample_rate = p->sample_rate; 721 f2.channels = p->channels; 722 f2.encoding = p->encoding; 723 f2.precision = p->validbits; 724 f2.stride = p->precision; 725 return f2; 726 } 727 728 /* Return true if this track is a playback track. */ 729 static __inline bool 730 audio_track_is_playback(const audio_track_t *track) 731 { 732 733 return ((track->mode & AUMODE_PLAY) != 0); 734 } 735 736 #if 0 737 /* Return true if this track is a recording track. */ 738 static __inline bool 739 audio_track_is_record(const audio_track_t *track) 740 { 741 742 return ((track->mode & AUMODE_RECORD) != 0); 743 } 744 #endif 745 746 #if 0 /* XXX Not used yet */ 747 /* 748 * Convert 0..255 volume used in userland to internal presentation 0..256. 749 */ 750 static __inline u_int 751 audio_volume_to_inner(u_int v) 752 { 753 754 return v < 127 ? v : v + 1; 755 } 756 757 /* 758 * Convert 0..256 internal presentation to 0..255 volume used in userland. 759 */ 760 static __inline u_int 761 audio_volume_to_outer(u_int v) 762 { 763 764 return v < 127 ? v : v - 1; 765 } 766 #endif /* 0 */ 767 768 static dev_type_open(audioopen); 769 /* XXXMRG use more dev_type_xxx */ 770 771 static int 772 audiounit(dev_t dev) 773 { 774 775 return AUDIOUNIT(dev); 776 } 777 778 const struct cdevsw audio_cdevsw = { 779 .d_open = audioopen, 780 .d_close = noclose, 781 .d_read = noread, 782 .d_write = nowrite, 783 .d_ioctl = noioctl, 784 .d_stop = nostop, 785 .d_tty = notty, 786 .d_poll = nopoll, 787 .d_mmap = nommap, 788 .d_kqfilter = nokqfilter, 789 .d_discard = nodiscard, 790 .d_cfdriver = &audio_cd, 791 .d_devtounit = audiounit, 792 .d_flag = D_OTHER | D_MPSAFE 793 }; 794 795 const struct fileops audio_fileops = { 796 .fo_name = "audio", 797 .fo_read = audioread, 798 .fo_write = audiowrite, 799 .fo_ioctl = audioioctl, 800 .fo_fcntl = fnullop_fcntl, 801 .fo_stat = audiostat, 802 .fo_poll = audiopoll, 803 .fo_close = audioclose, 804 .fo_mmap = audiommap, 805 .fo_kqfilter = audiokqfilter, 806 .fo_restart = fnullop_restart 807 }; 808 809 /* The default audio mode: 8 kHz mono mu-law */ 810 static const struct audio_params audio_default = { 811 .sample_rate = 8000, 812 .encoding = AUDIO_ENCODING_ULAW, 813 .precision = 8, 814 .validbits = 8, 815 .channels = 1, 816 }; 817 818 static const char *encoding_names[] = { 819 "none", 820 AudioEmulaw, 821 AudioEalaw, 822 "pcm16", 823 "pcm8", 824 AudioEadpcm, 825 AudioEslinear_le, 826 AudioEslinear_be, 827 AudioEulinear_le, 828 AudioEulinear_be, 829 AudioEslinear, 830 AudioEulinear, 831 AudioEmpeg_l1_stream, 832 AudioEmpeg_l1_packets, 833 AudioEmpeg_l1_system, 834 AudioEmpeg_l2_stream, 835 AudioEmpeg_l2_packets, 836 AudioEmpeg_l2_system, 837 AudioEac3, 838 }; 839 840 /* 841 * Returns encoding name corresponding to AUDIO_ENCODING_*. 842 * Note that it may return a local buffer because it is mainly for debugging. 843 */ 844 const char * 845 audio_encoding_name(int encoding) 846 { 847 static char buf[16]; 848 849 if (0 <= encoding && encoding < __arraycount(encoding_names)) { 850 return encoding_names[encoding]; 851 } else { 852 snprintf(buf, sizeof(buf), "enc=%d", encoding); 853 return buf; 854 } 855 } 856 857 /* 858 * Supported encodings used by AUDIO_GETENC. 859 * index and flags are set by code. 860 * XXX is there any needs for SLINEAR_OE:>=16/ULINEAR_OE:>=16 ? 861 */ 862 static const audio_encoding_t audio_encodings[] = { 863 { 0, AudioEmulaw, AUDIO_ENCODING_ULAW, 8, 0 }, 864 { 0, AudioEalaw, AUDIO_ENCODING_ALAW, 8, 0 }, 865 { 0, AudioEslinear, AUDIO_ENCODING_SLINEAR, 8, 0 }, 866 { 0, AudioEulinear, AUDIO_ENCODING_ULINEAR, 8, 0 }, 867 { 0, AudioEslinear_le, AUDIO_ENCODING_SLINEAR_LE, 16, 0 }, 868 { 0, AudioEulinear_le, AUDIO_ENCODING_ULINEAR_LE, 16, 0 }, 869 { 0, AudioEslinear_be, AUDIO_ENCODING_SLINEAR_BE, 16, 0 }, 870 { 0, AudioEulinear_be, AUDIO_ENCODING_ULINEAR_BE, 16, 0 }, 871 #if defined(AUDIO_SUPPORT_LINEAR24) 872 { 0, AudioEslinear_le, AUDIO_ENCODING_SLINEAR_LE, 24, 0 }, 873 { 0, AudioEulinear_le, AUDIO_ENCODING_ULINEAR_LE, 24, 0 }, 874 { 0, AudioEslinear_be, AUDIO_ENCODING_SLINEAR_BE, 24, 0 }, 875 { 0, AudioEulinear_be, AUDIO_ENCODING_ULINEAR_BE, 24, 0 }, 876 #endif 877 { 0, AudioEslinear_le, AUDIO_ENCODING_SLINEAR_LE, 32, 0 }, 878 { 0, AudioEulinear_le, AUDIO_ENCODING_ULINEAR_LE, 32, 0 }, 879 { 0, AudioEslinear_be, AUDIO_ENCODING_SLINEAR_BE, 32, 0 }, 880 { 0, AudioEulinear_be, AUDIO_ENCODING_ULINEAR_BE, 32, 0 }, 881 }; 882 883 static const struct portname itable[] = { 884 { AudioNmicrophone, AUDIO_MICROPHONE }, 885 { AudioNline, AUDIO_LINE_IN }, 886 { AudioNcd, AUDIO_CD }, 887 { 0, 0 } 888 }; 889 static const struct portname otable[] = { 890 { AudioNspeaker, AUDIO_SPEAKER }, 891 { AudioNheadphone, AUDIO_HEADPHONE }, 892 { AudioNline, AUDIO_LINE_OUT }, 893 { 0, 0 } 894 }; 895 896 static struct psref_class *audio_psref_class __read_mostly; 897 898 CFATTACH_DECL3_NEW(audio, sizeof(struct audio_softc), 899 audiomatch, audioattach, audiodetach, audioactivate, audiorescan, 900 audiochilddet, DVF_DETACH_SHUTDOWN); 901 902 static int 903 audiomatch(device_t parent, cfdata_t match, void *aux) 904 { 905 struct audio_attach_args *sa; 906 907 sa = aux; 908 DPRINTF(1, "%s: type=%d sa=%p hw=%p\n", 909 __func__, sa->type, sa, sa->hwif); 910 return (sa->type == AUDIODEV_TYPE_AUDIO) ? 1 : 0; 911 } 912 913 static void 914 audioattach(device_t parent, device_t self, void *aux) 915 { 916 struct audio_softc *sc; 917 struct audio_attach_args *sa; 918 const struct audio_hw_if *hw_if; 919 audio_format2_t phwfmt; 920 audio_format2_t rhwfmt; 921 audio_filter_reg_t pfil; 922 audio_filter_reg_t rfil; 923 const struct sysctlnode *node; 924 void *hdlp; 925 bool has_playback; 926 bool has_capture; 927 bool has_indep; 928 bool has_fulldup; 929 int mode; 930 int error; 931 932 sc = device_private(self); 933 sc->sc_dev = self; 934 sa = (struct audio_attach_args *)aux; 935 hw_if = sa->hwif; 936 hdlp = sa->hdl; 937 938 if (hw_if == NULL) { 939 panic("audioattach: missing hw_if method"); 940 } 941 if (hw_if->get_locks == NULL || hw_if->get_props == NULL) { 942 aprint_error(": missing mandatory method\n"); 943 return; 944 } 945 946 hw_if->get_locks(hdlp, &sc->sc_intr_lock, &sc->sc_lock); 947 sc->sc_props = hw_if->get_props(hdlp); 948 949 has_playback = (sc->sc_props & AUDIO_PROP_PLAYBACK); 950 has_capture = (sc->sc_props & AUDIO_PROP_CAPTURE); 951 has_indep = (sc->sc_props & AUDIO_PROP_INDEPENDENT); 952 has_fulldup = (sc->sc_props & AUDIO_PROP_FULLDUPLEX); 953 954 #ifdef DIAGNOSTIC 955 if (hw_if->query_format == NULL || 956 hw_if->set_format == NULL || 957 hw_if->getdev == NULL || 958 hw_if->set_port == NULL || 959 hw_if->get_port == NULL || 960 hw_if->query_devinfo == NULL) { 961 aprint_error(": missing mandatory method\n"); 962 return; 963 } 964 if (has_playback) { 965 if ((hw_if->start_output == NULL && 966 hw_if->trigger_output == NULL) || 967 hw_if->halt_output == NULL) { 968 aprint_error(": missing playback method\n"); 969 } 970 } 971 if (has_capture) { 972 if ((hw_if->start_input == NULL && 973 hw_if->trigger_input == NULL) || 974 hw_if->halt_input == NULL) { 975 aprint_error(": missing capture method\n"); 976 } 977 } 978 #endif 979 980 sc->hw_if = hw_if; 981 sc->hw_hdl = hdlp; 982 sc->hw_dev = parent; 983 984 sc->sc_exlock = 1; 985 sc->sc_blk_ms = AUDIO_BLK_MS; 986 SLIST_INIT(&sc->sc_files); 987 cv_init(&sc->sc_exlockcv, "audiolk"); 988 sc->sc_am_capacity = 0; 989 sc->sc_am_used = 0; 990 sc->sc_am = NULL; 991 992 /* MMAP is now supported by upper layer. */ 993 sc->sc_props |= AUDIO_PROP_MMAP; 994 995 KASSERT(has_playback || has_capture); 996 /* Unidirectional device must have neither FULLDUP nor INDEPENDENT. */ 997 if (!has_playback || !has_capture) { 998 KASSERT(!has_indep); 999 KASSERT(!has_fulldup); 1000 } 1001 1002 mode = 0; 1003 if (has_playback) { 1004 aprint_normal(": playback"); 1005 mode |= AUMODE_PLAY; 1006 } 1007 if (has_capture) { 1008 aprint_normal("%c capture", has_playback ? ',' : ':'); 1009 mode |= AUMODE_RECORD; 1010 } 1011 if (has_playback && has_capture) { 1012 if (has_fulldup) 1013 aprint_normal(", full duplex"); 1014 else 1015 aprint_normal(", half duplex"); 1016 1017 if (has_indep) 1018 aprint_normal(", independent"); 1019 } 1020 1021 aprint_naive("\n"); 1022 aprint_normal("\n"); 1023 1024 /* probe hw params */ 1025 memset(&phwfmt, 0, sizeof(phwfmt)); 1026 memset(&rhwfmt, 0, sizeof(rhwfmt)); 1027 memset(&pfil, 0, sizeof(pfil)); 1028 memset(&rfil, 0, sizeof(rfil)); 1029 if (has_indep) { 1030 int perror, rerror; 1031 1032 /* On independent devices, probe separately. */ 1033 perror = audio_hw_probe(sc, &phwfmt, AUMODE_PLAY); 1034 rerror = audio_hw_probe(sc, &rhwfmt, AUMODE_RECORD); 1035 if (perror && rerror) { 1036 aprint_error_dev(self, 1037 "audio_hw_probe failed: perror=%d, rerror=%d\n", 1038 perror, rerror); 1039 goto bad; 1040 } 1041 if (perror) { 1042 mode &= ~AUMODE_PLAY; 1043 aprint_error_dev(self, "audio_hw_probe failed: " 1044 "errno=%d, playback disabled\n", perror); 1045 } 1046 if (rerror) { 1047 mode &= ~AUMODE_RECORD; 1048 aprint_error_dev(self, "audio_hw_probe failed: " 1049 "errno=%d, capture disabled\n", rerror); 1050 } 1051 } else { 1052 /* 1053 * On non independent devices or uni-directional devices, 1054 * probe once (simultaneously). 1055 */ 1056 audio_format2_t *fmt = has_playback ? &phwfmt : &rhwfmt; 1057 error = audio_hw_probe(sc, fmt, mode); 1058 if (error) { 1059 aprint_error_dev(self, 1060 "audio_hw_probe failed: errno=%d\n", error); 1061 goto bad; 1062 } 1063 if (has_playback && has_capture) 1064 rhwfmt = phwfmt; 1065 } 1066 1067 /* Init hardware. */ 1068 /* hw_probe() also validates [pr]hwfmt. */ 1069 error = audio_hw_set_format(sc, mode, &phwfmt, &rhwfmt, &pfil, &rfil); 1070 if (error) { 1071 aprint_error_dev(self, 1072 "audio_hw_set_format failed: errno=%d\n", error); 1073 goto bad; 1074 } 1075 1076 /* 1077 * Init track mixers. If at least one direction is available on 1078 * attach time, we assume a success. 1079 */ 1080 error = audio_mixers_init(sc, mode, &phwfmt, &rhwfmt, &pfil, &rfil); 1081 if (sc->sc_pmixer == NULL && sc->sc_rmixer == NULL) { 1082 aprint_error_dev(self, 1083 "audio_mixers_init failed: errno=%d\n", error); 1084 goto bad; 1085 } 1086 1087 sc->sc_psz = pserialize_create(); 1088 psref_target_init(&sc->sc_psref, audio_psref_class); 1089 1090 selinit(&sc->sc_wsel); 1091 selinit(&sc->sc_rsel); 1092 1093 /* Initial parameter of /dev/sound */ 1094 sc->sc_sound_pparams = params_to_format2(&audio_default); 1095 sc->sc_sound_rparams = params_to_format2(&audio_default); 1096 sc->sc_sound_ppause = false; 1097 sc->sc_sound_rpause = false; 1098 1099 /* XXX TODO: consider about sc_ai */ 1100 1101 mixer_init(sc); 1102 TRACE(2, "inputs ports=0x%x, input master=%d, " 1103 "output ports=0x%x, output master=%d", 1104 sc->sc_inports.allports, sc->sc_inports.master, 1105 sc->sc_outports.allports, sc->sc_outports.master); 1106 1107 sysctl_createv(&sc->sc_log, 0, NULL, &node, 1108 0, 1109 CTLTYPE_NODE, device_xname(sc->sc_dev), 1110 SYSCTL_DESCR("audio test"), 1111 NULL, 0, 1112 NULL, 0, 1113 CTL_HW, 1114 CTL_CREATE, CTL_EOL); 1115 1116 if (node != NULL) { 1117 sysctl_createv(&sc->sc_log, 0, NULL, NULL, 1118 CTLFLAG_READWRITE, 1119 CTLTYPE_INT, "blk_ms", 1120 SYSCTL_DESCR("blocksize in msec"), 1121 audio_sysctl_blk_ms, 0, (void *)sc, 0, 1122 CTL_HW, node->sysctl_num, CTL_CREATE, CTL_EOL); 1123 1124 sysctl_createv(&sc->sc_log, 0, NULL, NULL, 1125 CTLFLAG_READWRITE, 1126 CTLTYPE_BOOL, "multiuser", 1127 SYSCTL_DESCR("allow multiple user access"), 1128 audio_sysctl_multiuser, 0, (void *)sc, 0, 1129 CTL_HW, node->sysctl_num, CTL_CREATE, CTL_EOL); 1130 1131 #if defined(AUDIO_DEBUG) 1132 sysctl_createv(&sc->sc_log, 0, NULL, NULL, 1133 CTLFLAG_READWRITE, 1134 CTLTYPE_INT, "debug", 1135 SYSCTL_DESCR("debug level (0..4)"), 1136 audio_sysctl_debug, 0, (void *)sc, 0, 1137 CTL_HW, node->sysctl_num, CTL_CREATE, CTL_EOL); 1138 #endif 1139 } 1140 1141 #ifdef AUDIO_PM_IDLE 1142 callout_init(&sc->sc_idle_counter, 0); 1143 callout_setfunc(&sc->sc_idle_counter, audio_idle, self); 1144 #endif 1145 1146 if (!pmf_device_register(self, audio_suspend, audio_resume)) 1147 aprint_error_dev(self, "couldn't establish power handler\n"); 1148 #ifdef AUDIO_PM_IDLE 1149 if (!device_active_register(self, audio_activity)) 1150 aprint_error_dev(self, "couldn't register activity handler\n"); 1151 #endif 1152 1153 if (!pmf_event_register(self, PMFE_AUDIO_VOLUME_DOWN, 1154 audio_volume_down, true)) 1155 aprint_error_dev(self, "couldn't add volume down handler\n"); 1156 if (!pmf_event_register(self, PMFE_AUDIO_VOLUME_UP, 1157 audio_volume_up, true)) 1158 aprint_error_dev(self, "couldn't add volume up handler\n"); 1159 if (!pmf_event_register(self, PMFE_AUDIO_VOLUME_TOGGLE, 1160 audio_volume_toggle, true)) 1161 aprint_error_dev(self, "couldn't add volume toggle handler\n"); 1162 1163 #ifdef AUDIO_PM_IDLE 1164 callout_schedule(&sc->sc_idle_counter, audio_idle_timeout * hz); 1165 #endif 1166 1167 #if defined(AUDIO_DEBUG) 1168 audio_mlog_init(); 1169 #endif 1170 1171 audiorescan(self, NULL, NULL); 1172 sc->sc_exlock = 0; 1173 return; 1174 1175 bad: 1176 /* Clearing hw_if means that device is attached but disabled. */ 1177 sc->hw_if = NULL; 1178 sc->sc_exlock = 0; 1179 aprint_error_dev(sc->sc_dev, "disabled\n"); 1180 return; 1181 } 1182 1183 /* 1184 * Initialize hardware mixer. 1185 * This function is called from audioattach(). 1186 */ 1187 static void 1188 mixer_init(struct audio_softc *sc) 1189 { 1190 mixer_devinfo_t mi; 1191 int iclass, mclass, oclass, rclass; 1192 int record_master_found, record_source_found; 1193 1194 iclass = mclass = oclass = rclass = -1; 1195 sc->sc_inports.index = -1; 1196 sc->sc_inports.master = -1; 1197 sc->sc_inports.nports = 0; 1198 sc->sc_inports.isenum = false; 1199 sc->sc_inports.allports = 0; 1200 sc->sc_inports.isdual = false; 1201 sc->sc_inports.mixerout = -1; 1202 sc->sc_inports.cur_port = -1; 1203 sc->sc_outports.index = -1; 1204 sc->sc_outports.master = -1; 1205 sc->sc_outports.nports = 0; 1206 sc->sc_outports.isenum = false; 1207 sc->sc_outports.allports = 0; 1208 sc->sc_outports.isdual = false; 1209 sc->sc_outports.mixerout = -1; 1210 sc->sc_outports.cur_port = -1; 1211 sc->sc_monitor_port = -1; 1212 /* 1213 * Read through the underlying driver's list, picking out the class 1214 * names from the mixer descriptions. We'll need them to decode the 1215 * mixer descriptions on the next pass through the loop. 1216 */ 1217 mutex_enter(sc->sc_lock); 1218 for(mi.index = 0; ; mi.index++) { 1219 if (audio_query_devinfo(sc, &mi) != 0) 1220 break; 1221 /* 1222 * The type of AUDIO_MIXER_CLASS merely introduces a class. 1223 * All the other types describe an actual mixer. 1224 */ 1225 if (mi.type == AUDIO_MIXER_CLASS) { 1226 if (strcmp(mi.label.name, AudioCinputs) == 0) 1227 iclass = mi.mixer_class; 1228 if (strcmp(mi.label.name, AudioCmonitor) == 0) 1229 mclass = mi.mixer_class; 1230 if (strcmp(mi.label.name, AudioCoutputs) == 0) 1231 oclass = mi.mixer_class; 1232 if (strcmp(mi.label.name, AudioCrecord) == 0) 1233 rclass = mi.mixer_class; 1234 } 1235 } 1236 mutex_exit(sc->sc_lock); 1237 1238 /* Allocate save area. Ensure non-zero allocation. */ 1239 sc->sc_nmixer_states = mi.index; 1240 sc->sc_mixer_state = kmem_zalloc(sizeof(sc->sc_mixer_state[0]) * 1241 (sc->sc_nmixer_states + 1), KM_SLEEP); 1242 1243 /* 1244 * This is where we assign each control in the "audio" model, to the 1245 * underlying "mixer" control. We walk through the whole list once, 1246 * assigning likely candidates as we come across them. 1247 */ 1248 record_master_found = 0; 1249 record_source_found = 0; 1250 mutex_enter(sc->sc_lock); 1251 for(mi.index = 0; ; mi.index++) { 1252 if (audio_query_devinfo(sc, &mi) != 0) 1253 break; 1254 KASSERT(mi.index < sc->sc_nmixer_states); 1255 if (mi.type == AUDIO_MIXER_CLASS) 1256 continue; 1257 if (mi.mixer_class == iclass) { 1258 /* 1259 * AudioCinputs is only a fallback, when we don't 1260 * find what we're looking for in AudioCrecord, so 1261 * check the flags before accepting one of these. 1262 */ 1263 if (strcmp(mi.label.name, AudioNmaster) == 0 1264 && record_master_found == 0) 1265 sc->sc_inports.master = mi.index; 1266 if (strcmp(mi.label.name, AudioNsource) == 0 1267 && record_source_found == 0) { 1268 if (mi.type == AUDIO_MIXER_ENUM) { 1269 int i; 1270 for(i = 0; i < mi.un.e.num_mem; i++) 1271 if (strcmp(mi.un.e.member[i].label.name, 1272 AudioNmixerout) == 0) 1273 sc->sc_inports.mixerout = 1274 mi.un.e.member[i].ord; 1275 } 1276 au_setup_ports(sc, &sc->sc_inports, &mi, 1277 itable); 1278 } 1279 if (strcmp(mi.label.name, AudioNdac) == 0 && 1280 sc->sc_outports.master == -1) 1281 sc->sc_outports.master = mi.index; 1282 } else if (mi.mixer_class == mclass) { 1283 if (strcmp(mi.label.name, AudioNmonitor) == 0) 1284 sc->sc_monitor_port = mi.index; 1285 } else if (mi.mixer_class == oclass) { 1286 if (strcmp(mi.label.name, AudioNmaster) == 0) 1287 sc->sc_outports.master = mi.index; 1288 if (strcmp(mi.label.name, AudioNselect) == 0) 1289 au_setup_ports(sc, &sc->sc_outports, &mi, 1290 otable); 1291 } else if (mi.mixer_class == rclass) { 1292 /* 1293 * These are the preferred mixers for the audio record 1294 * controls, so set the flags here, but don't check. 1295 */ 1296 if (strcmp(mi.label.name, AudioNmaster) == 0) { 1297 sc->sc_inports.master = mi.index; 1298 record_master_found = 1; 1299 } 1300 #if 1 /* Deprecated. Use AudioNmaster. */ 1301 if (strcmp(mi.label.name, AudioNrecord) == 0) { 1302 sc->sc_inports.master = mi.index; 1303 record_master_found = 1; 1304 } 1305 if (strcmp(mi.label.name, AudioNvolume) == 0) { 1306 sc->sc_inports.master = mi.index; 1307 record_master_found = 1; 1308 } 1309 #endif 1310 if (strcmp(mi.label.name, AudioNsource) == 0) { 1311 if (mi.type == AUDIO_MIXER_ENUM) { 1312 int i; 1313 for(i = 0; i < mi.un.e.num_mem; i++) 1314 if (strcmp(mi.un.e.member[i].label.name, 1315 AudioNmixerout) == 0) 1316 sc->sc_inports.mixerout = 1317 mi.un.e.member[i].ord; 1318 } 1319 au_setup_ports(sc, &sc->sc_inports, &mi, 1320 itable); 1321 record_source_found = 1; 1322 } 1323 } 1324 } 1325 mutex_exit(sc->sc_lock); 1326 } 1327 1328 static int 1329 audioactivate(device_t self, enum devact act) 1330 { 1331 struct audio_softc *sc = device_private(self); 1332 1333 switch (act) { 1334 case DVACT_DEACTIVATE: 1335 mutex_enter(sc->sc_lock); 1336 sc->sc_dying = true; 1337 cv_broadcast(&sc->sc_exlockcv); 1338 mutex_exit(sc->sc_lock); 1339 return 0; 1340 default: 1341 return EOPNOTSUPP; 1342 } 1343 } 1344 1345 static int 1346 audiodetach(device_t self, int flags) 1347 { 1348 struct audio_softc *sc; 1349 struct audio_file *file; 1350 int maj, mn; 1351 int error; 1352 1353 sc = device_private(self); 1354 TRACE(2, "flags=%d", flags); 1355 1356 /* device is not initialized */ 1357 if (sc->hw_if == NULL) 1358 return 0; 1359 1360 /* Start draining existing accessors of the device. */ 1361 error = config_detach_children(self, flags); 1362 if (error) 1363 return error; 1364 1365 /* 1366 * Prevent new opens and wait for existing opens to complete. 1367 */ 1368 maj = cdevsw_lookup_major(&audio_cdevsw); 1369 mn = device_unit(self); 1370 vdevgone(maj, mn|SOUND_DEVICE, mn|SOUND_DEVICE, VCHR); 1371 vdevgone(maj, mn|AUDIO_DEVICE, mn|AUDIO_DEVICE, VCHR); 1372 vdevgone(maj, mn|AUDIOCTL_DEVICE, mn|AUDIOCTL_DEVICE, VCHR); 1373 vdevgone(maj, mn|MIXER_DEVICE, mn|MIXER_DEVICE, VCHR); 1374 1375 /* 1376 * This waits currently running sysctls to finish if exists. 1377 * After this, no more new sysctls will come. 1378 */ 1379 sysctl_teardown(&sc->sc_log); 1380 1381 mutex_enter(sc->sc_lock); 1382 sc->sc_dying = true; 1383 cv_broadcast(&sc->sc_exlockcv); 1384 if (sc->sc_pmixer) 1385 cv_broadcast(&sc->sc_pmixer->outcv); 1386 if (sc->sc_rmixer) 1387 cv_broadcast(&sc->sc_rmixer->outcv); 1388 1389 /* Prevent new users */ 1390 SLIST_FOREACH(file, &sc->sc_files, entry) { 1391 atomic_store_relaxed(&file->dying, true); 1392 } 1393 mutex_exit(sc->sc_lock); 1394 1395 /* 1396 * Wait for existing users to drain. 1397 * - pserialize_perform waits for all pserialize_read sections on 1398 * all CPUs; after this, no more new psref_acquire can happen. 1399 * - psref_target_destroy waits for all extant acquired psrefs to 1400 * be psref_released. 1401 */ 1402 pserialize_perform(sc->sc_psz); 1403 psref_target_destroy(&sc->sc_psref, audio_psref_class); 1404 1405 /* 1406 * We are now guaranteed that there are no calls to audio fileops 1407 * that hold sc, and any new calls with files that were for sc will 1408 * fail. Thus, we now have exclusive access to the softc. 1409 */ 1410 sc->sc_exlock = 1; 1411 1412 /* 1413 * Clean up all open instances. 1414 */ 1415 mutex_enter(sc->sc_lock); 1416 while ((file = SLIST_FIRST(&sc->sc_files)) != NULL) { 1417 mutex_enter(sc->sc_intr_lock); 1418 SLIST_REMOVE_HEAD(&sc->sc_files, entry); 1419 mutex_exit(sc->sc_intr_lock); 1420 if (file->ptrack || file->rtrack) { 1421 mutex_exit(sc->sc_lock); 1422 audio_unlink(sc, file); 1423 mutex_enter(sc->sc_lock); 1424 } 1425 } 1426 mutex_exit(sc->sc_lock); 1427 1428 pmf_event_deregister(self, PMFE_AUDIO_VOLUME_DOWN, 1429 audio_volume_down, true); 1430 pmf_event_deregister(self, PMFE_AUDIO_VOLUME_UP, 1431 audio_volume_up, true); 1432 pmf_event_deregister(self, PMFE_AUDIO_VOLUME_TOGGLE, 1433 audio_volume_toggle, true); 1434 1435 #ifdef AUDIO_PM_IDLE 1436 callout_halt(&sc->sc_idle_counter, sc->sc_lock); 1437 1438 device_active_deregister(self, audio_activity); 1439 #endif 1440 1441 pmf_device_deregister(self); 1442 1443 /* Free resources */ 1444 if (sc->sc_pmixer) { 1445 audio_mixer_destroy(sc, sc->sc_pmixer); 1446 kmem_free(sc->sc_pmixer, sizeof(*sc->sc_pmixer)); 1447 } 1448 if (sc->sc_rmixer) { 1449 audio_mixer_destroy(sc, sc->sc_rmixer); 1450 kmem_free(sc->sc_rmixer, sizeof(*sc->sc_rmixer)); 1451 } 1452 if (sc->sc_am) 1453 kern_free(sc->sc_am); 1454 1455 seldestroy(&sc->sc_wsel); 1456 seldestroy(&sc->sc_rsel); 1457 1458 #ifdef AUDIO_PM_IDLE 1459 callout_destroy(&sc->sc_idle_counter); 1460 #endif 1461 1462 cv_destroy(&sc->sc_exlockcv); 1463 1464 #if defined(AUDIO_DEBUG) 1465 audio_mlog_free(); 1466 #endif 1467 1468 return 0; 1469 } 1470 1471 static void 1472 audiochilddet(device_t self, device_t child) 1473 { 1474 1475 /* we hold no child references, so do nothing */ 1476 } 1477 1478 static int 1479 audiosearch(device_t parent, cfdata_t cf, const int *locs, void *aux) 1480 { 1481 1482 if (config_probe(parent, cf, aux)) 1483 config_attach(parent, cf, aux, NULL, 1484 CFARGS_NONE); 1485 1486 return 0; 1487 } 1488 1489 static int 1490 audiorescan(device_t self, const char *ifattr, const int *locators) 1491 { 1492 struct audio_softc *sc = device_private(self); 1493 1494 config_search(sc->sc_dev, NULL, 1495 CFARGS(.search = audiosearch)); 1496 1497 return 0; 1498 } 1499 1500 /* 1501 * Called from hardware driver. This is where the MI audio driver gets 1502 * probed/attached to the hardware driver. 1503 */ 1504 device_t 1505 audio_attach_mi(const struct audio_hw_if *ahwp, void *hdlp, device_t dev) 1506 { 1507 struct audio_attach_args arg; 1508 1509 #ifdef DIAGNOSTIC 1510 if (ahwp == NULL) { 1511 aprint_error("audio_attach_mi: NULL\n"); 1512 return 0; 1513 } 1514 #endif 1515 arg.type = AUDIODEV_TYPE_AUDIO; 1516 arg.hwif = ahwp; 1517 arg.hdl = hdlp; 1518 return config_found(dev, &arg, audioprint, 1519 CFARGS(.iattr = "audiobus")); 1520 } 1521 1522 /* 1523 * audio_printf() outputs fmt... with the audio device name and MD device 1524 * name prefixed. If the message is considered to be related to the MD 1525 * driver, use this one instead of device_printf(). 1526 */ 1527 static void 1528 audio_printf(struct audio_softc *sc, const char *fmt, ...) 1529 { 1530 va_list ap; 1531 1532 printf("%s(%s): ", device_xname(sc->sc_dev), device_xname(sc->hw_dev)); 1533 va_start(ap, fmt); 1534 vprintf(fmt, ap); 1535 va_end(ap); 1536 } 1537 1538 /* 1539 * Enter critical section and also keep sc_lock. 1540 * If successful, returns 0 with sc_lock held. Otherwise returns errno. 1541 * Must be called without sc_lock held. 1542 */ 1543 static int 1544 audio_exlock_mutex_enter(struct audio_softc *sc) 1545 { 1546 int error; 1547 1548 mutex_enter(sc->sc_lock); 1549 if (sc->sc_dying) { 1550 mutex_exit(sc->sc_lock); 1551 return EIO; 1552 } 1553 1554 while (__predict_false(sc->sc_exlock != 0)) { 1555 error = cv_wait_sig(&sc->sc_exlockcv, sc->sc_lock); 1556 if (sc->sc_dying) 1557 error = EIO; 1558 if (error) { 1559 mutex_exit(sc->sc_lock); 1560 return error; 1561 } 1562 } 1563 1564 /* Acquire */ 1565 sc->sc_exlock = 1; 1566 return 0; 1567 } 1568 1569 /* 1570 * Exit critical section and exit sc_lock. 1571 * Must be called with sc_lock held. 1572 */ 1573 static void 1574 audio_exlock_mutex_exit(struct audio_softc *sc) 1575 { 1576 1577 KASSERT(mutex_owned(sc->sc_lock)); 1578 1579 sc->sc_exlock = 0; 1580 cv_broadcast(&sc->sc_exlockcv); 1581 mutex_exit(sc->sc_lock); 1582 } 1583 1584 /* 1585 * Enter critical section. 1586 * If successful, it returns 0. Otherwise returns errno. 1587 * Must be called without sc_lock held. 1588 * This function returns without sc_lock held. 1589 */ 1590 static int 1591 audio_exlock_enter(struct audio_softc *sc) 1592 { 1593 int error; 1594 1595 error = audio_exlock_mutex_enter(sc); 1596 if (error) 1597 return error; 1598 mutex_exit(sc->sc_lock); 1599 return 0; 1600 } 1601 1602 /* 1603 * Exit critical section. 1604 * Must be called without sc_lock held. 1605 */ 1606 static void 1607 audio_exlock_exit(struct audio_softc *sc) 1608 { 1609 1610 mutex_enter(sc->sc_lock); 1611 audio_exlock_mutex_exit(sc); 1612 } 1613 1614 /* 1615 * Get sc from file, and increment reference counter for this sc. 1616 * This is intended to be used for methods other than open. 1617 * If successful, returns sc. Otherwise returns NULL. 1618 */ 1619 struct audio_softc * 1620 audio_sc_acquire_fromfile(audio_file_t *file, struct psref *refp) 1621 { 1622 int s; 1623 bool dying; 1624 1625 /* Block audiodetach while we acquire a reference */ 1626 s = pserialize_read_enter(); 1627 1628 /* If close or audiodetach already ran, tough -- no more audio */ 1629 dying = atomic_load_relaxed(&file->dying); 1630 if (dying) { 1631 pserialize_read_exit(s); 1632 return NULL; 1633 } 1634 1635 /* Acquire a reference */ 1636 psref_acquire(refp, &file->sc->sc_psref, audio_psref_class); 1637 1638 /* Now sc won't go away until we drop the reference count */ 1639 pserialize_read_exit(s); 1640 1641 return file->sc; 1642 } 1643 1644 /* 1645 * Decrement reference counter for this sc. 1646 */ 1647 void 1648 audio_sc_release(struct audio_softc *sc, struct psref *refp) 1649 { 1650 1651 psref_release(refp, &sc->sc_psref, audio_psref_class); 1652 } 1653 1654 /* 1655 * Wait for I/O to complete, releasing sc_lock. 1656 * Must be called with sc_lock held. 1657 */ 1658 static int 1659 audio_track_waitio(struct audio_softc *sc, audio_track_t *track) 1660 { 1661 int error; 1662 1663 KASSERT(track); 1664 KASSERT(mutex_owned(sc->sc_lock)); 1665 1666 /* Wait for pending I/O to complete. */ 1667 error = cv_timedwait_sig(&track->mixer->outcv, sc->sc_lock, 1668 mstohz(AUDIO_TIMEOUT)); 1669 if (sc->sc_suspending) { 1670 /* If it's about to suspend, ignore timeout error. */ 1671 if (error == EWOULDBLOCK) { 1672 TRACET(2, track, "timeout (suspending)"); 1673 return 0; 1674 } 1675 } 1676 if (sc->sc_dying) { 1677 error = EIO; 1678 } 1679 if (error) { 1680 TRACET(2, track, "cv_timedwait_sig failed %d", error); 1681 if (error == EWOULDBLOCK) 1682 audio_printf(sc, "device timeout\n"); 1683 } else { 1684 TRACET(3, track, "wakeup"); 1685 } 1686 return error; 1687 } 1688 1689 /* 1690 * Try to acquire track lock. 1691 * It doesn't block if the track lock is already acquired. 1692 * Returns true if the track lock was acquired, or false if the track 1693 * lock was already acquired. 1694 */ 1695 static __inline bool 1696 audio_track_lock_tryenter(audio_track_t *track) 1697 { 1698 1699 if (atomic_swap_uint(&track->lock, 1) != 0) 1700 return false; 1701 membar_acquire(); 1702 return true; 1703 } 1704 1705 /* 1706 * Acquire track lock. 1707 */ 1708 static __inline void 1709 audio_track_lock_enter(audio_track_t *track) 1710 { 1711 1712 /* Don't sleep here. */ 1713 while (audio_track_lock_tryenter(track) == false) 1714 SPINLOCK_BACKOFF_HOOK; 1715 } 1716 1717 /* 1718 * Release track lock. 1719 */ 1720 static __inline void 1721 audio_track_lock_exit(audio_track_t *track) 1722 { 1723 1724 atomic_store_release(&track->lock, 0); 1725 } 1726 1727 1728 static int 1729 audioopen(dev_t dev, int flags, int ifmt, struct lwp *l) 1730 { 1731 struct audio_softc *sc; 1732 int error; 1733 1734 /* 1735 * Find the device. Because we wired the cdevsw to the audio 1736 * autoconf instance, the system ensures it will not go away 1737 * until after we return. 1738 */ 1739 sc = device_lookup_private(&audio_cd, AUDIOUNIT(dev)); 1740 if (sc == NULL || sc->hw_if == NULL) 1741 return ENXIO; 1742 1743 error = audio_exlock_enter(sc); 1744 if (error) 1745 return error; 1746 1747 device_active(sc->sc_dev, DVA_SYSTEM); 1748 switch (AUDIODEV(dev)) { 1749 case SOUND_DEVICE: 1750 case AUDIO_DEVICE: 1751 error = audio_open(dev, sc, flags, ifmt, l, NULL); 1752 break; 1753 case AUDIOCTL_DEVICE: 1754 error = audioctl_open(dev, sc, flags, ifmt, l); 1755 break; 1756 case MIXER_DEVICE: 1757 error = mixer_open(dev, sc, flags, ifmt, l); 1758 break; 1759 default: 1760 error = ENXIO; 1761 break; 1762 } 1763 audio_exlock_exit(sc); 1764 1765 return error; 1766 } 1767 1768 static int 1769 audioclose(struct file *fp) 1770 { 1771 struct audio_softc *sc; 1772 struct psref sc_ref; 1773 audio_file_t *file; 1774 int bound; 1775 int error; 1776 dev_t dev; 1777 1778 KASSERT(fp->f_audioctx); 1779 file = fp->f_audioctx; 1780 dev = file->dev; 1781 error = 0; 1782 1783 /* 1784 * audioclose() must 1785 * - unplug track from the trackmixer (and unplug anything from softc), 1786 * if sc exists. 1787 * - free all memory objects, regardless of sc. 1788 */ 1789 1790 bound = curlwp_bind(); 1791 sc = audio_sc_acquire_fromfile(file, &sc_ref); 1792 if (sc) { 1793 switch (AUDIODEV(dev)) { 1794 case SOUND_DEVICE: 1795 case AUDIO_DEVICE: 1796 error = audio_close(sc, file); 1797 break; 1798 case AUDIOCTL_DEVICE: 1799 mutex_enter(sc->sc_lock); 1800 mutex_enter(sc->sc_intr_lock); 1801 SLIST_REMOVE(&sc->sc_files, file, audio_file, entry); 1802 mutex_exit(sc->sc_intr_lock); 1803 mutex_exit(sc->sc_lock); 1804 error = 0; 1805 break; 1806 case MIXER_DEVICE: 1807 mutex_enter(sc->sc_lock); 1808 mutex_enter(sc->sc_intr_lock); 1809 SLIST_REMOVE(&sc->sc_files, file, audio_file, entry); 1810 mutex_exit(sc->sc_intr_lock); 1811 mutex_exit(sc->sc_lock); 1812 error = mixer_close(sc, file); 1813 break; 1814 default: 1815 error = ENXIO; 1816 break; 1817 } 1818 1819 audio_sc_release(sc, &sc_ref); 1820 } 1821 curlwp_bindx(bound); 1822 1823 /* Free memory objects anyway */ 1824 TRACEF(2, file, "free memory"); 1825 if (file->ptrack) 1826 audio_track_destroy(file->ptrack); 1827 if (file->rtrack) 1828 audio_track_destroy(file->rtrack); 1829 kmem_free(file, sizeof(*file)); 1830 fp->f_audioctx = NULL; 1831 1832 return error; 1833 } 1834 1835 static int 1836 audioread(struct file *fp, off_t *offp, struct uio *uio, kauth_cred_t cred, 1837 int ioflag) 1838 { 1839 struct audio_softc *sc; 1840 struct psref sc_ref; 1841 audio_file_t *file; 1842 int bound; 1843 int error; 1844 dev_t dev; 1845 1846 KASSERT(fp->f_audioctx); 1847 file = fp->f_audioctx; 1848 dev = file->dev; 1849 1850 bound = curlwp_bind(); 1851 sc = audio_sc_acquire_fromfile(file, &sc_ref); 1852 if (sc == NULL) { 1853 error = EIO; 1854 goto done; 1855 } 1856 1857 if (fp->f_flag & O_NONBLOCK) 1858 ioflag |= IO_NDELAY; 1859 1860 switch (AUDIODEV(dev)) { 1861 case SOUND_DEVICE: 1862 case AUDIO_DEVICE: 1863 error = audio_read(sc, uio, ioflag, file); 1864 break; 1865 case AUDIOCTL_DEVICE: 1866 case MIXER_DEVICE: 1867 error = ENODEV; 1868 break; 1869 default: 1870 error = ENXIO; 1871 break; 1872 } 1873 1874 audio_sc_release(sc, &sc_ref); 1875 done: 1876 curlwp_bindx(bound); 1877 return error; 1878 } 1879 1880 static int 1881 audiowrite(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_write(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 audioioctl(struct file *fp, u_long cmd, void *addr) 1927 { 1928 struct audio_softc *sc; 1929 struct psref sc_ref; 1930 audio_file_t *file; 1931 struct lwp *l = curlwp; 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 switch (AUDIODEV(dev)) { 1948 case SOUND_DEVICE: 1949 case AUDIO_DEVICE: 1950 case AUDIOCTL_DEVICE: 1951 mutex_enter(sc->sc_lock); 1952 device_active(sc->sc_dev, DVA_SYSTEM); 1953 mutex_exit(sc->sc_lock); 1954 if (IOCGROUP(cmd) == IOCGROUP(AUDIO_MIXER_READ)) 1955 error = mixer_ioctl(sc, cmd, addr, fp->f_flag, l); 1956 else 1957 error = audio_ioctl(dev, sc, cmd, addr, fp->f_flag, l, 1958 file); 1959 break; 1960 case MIXER_DEVICE: 1961 error = mixer_ioctl(sc, cmd, addr, fp->f_flag, l); 1962 break; 1963 default: 1964 error = ENXIO; 1965 break; 1966 } 1967 1968 audio_sc_release(sc, &sc_ref); 1969 done: 1970 curlwp_bindx(bound); 1971 return error; 1972 } 1973 1974 static int 1975 audiostat(struct file *fp, struct stat *st) 1976 { 1977 struct audio_softc *sc; 1978 struct psref sc_ref; 1979 audio_file_t *file; 1980 int bound; 1981 int error; 1982 1983 KASSERT(fp->f_audioctx); 1984 file = fp->f_audioctx; 1985 1986 bound = curlwp_bind(); 1987 sc = audio_sc_acquire_fromfile(file, &sc_ref); 1988 if (sc == NULL) { 1989 error = EIO; 1990 goto done; 1991 } 1992 1993 error = 0; 1994 memset(st, 0, sizeof(*st)); 1995 1996 st->st_dev = file->dev; 1997 st->st_uid = kauth_cred_geteuid(fp->f_cred); 1998 st->st_gid = kauth_cred_getegid(fp->f_cred); 1999 st->st_mode = S_IFCHR; 2000 2001 audio_sc_release(sc, &sc_ref); 2002 done: 2003 curlwp_bindx(bound); 2004 return error; 2005 } 2006 2007 static int 2008 audiopoll(struct file *fp, int events) 2009 { 2010 struct audio_softc *sc; 2011 struct psref sc_ref; 2012 audio_file_t *file; 2013 struct lwp *l = curlwp; 2014 int bound; 2015 int revents; 2016 dev_t dev; 2017 2018 KASSERT(fp->f_audioctx); 2019 file = fp->f_audioctx; 2020 dev = file->dev; 2021 2022 bound = curlwp_bind(); 2023 sc = audio_sc_acquire_fromfile(file, &sc_ref); 2024 if (sc == NULL) { 2025 revents = POLLERR; 2026 goto done; 2027 } 2028 2029 switch (AUDIODEV(dev)) { 2030 case SOUND_DEVICE: 2031 case AUDIO_DEVICE: 2032 revents = audio_poll(sc, events, l, file); 2033 break; 2034 case AUDIOCTL_DEVICE: 2035 case MIXER_DEVICE: 2036 revents = 0; 2037 break; 2038 default: 2039 revents = POLLERR; 2040 break; 2041 } 2042 2043 audio_sc_release(sc, &sc_ref); 2044 done: 2045 curlwp_bindx(bound); 2046 return revents; 2047 } 2048 2049 static int 2050 audiokqfilter(struct file *fp, struct knote *kn) 2051 { 2052 struct audio_softc *sc; 2053 struct psref sc_ref; 2054 audio_file_t *file; 2055 dev_t dev; 2056 int bound; 2057 int error; 2058 2059 KASSERT(fp->f_audioctx); 2060 file = fp->f_audioctx; 2061 dev = file->dev; 2062 2063 bound = curlwp_bind(); 2064 sc = audio_sc_acquire_fromfile(file, &sc_ref); 2065 if (sc == NULL) { 2066 error = EIO; 2067 goto done; 2068 } 2069 2070 switch (AUDIODEV(dev)) { 2071 case SOUND_DEVICE: 2072 case AUDIO_DEVICE: 2073 error = audio_kqfilter(sc, file, kn); 2074 break; 2075 case AUDIOCTL_DEVICE: 2076 case MIXER_DEVICE: 2077 error = ENODEV; 2078 break; 2079 default: 2080 error = ENXIO; 2081 break; 2082 } 2083 2084 audio_sc_release(sc, &sc_ref); 2085 done: 2086 curlwp_bindx(bound); 2087 return error; 2088 } 2089 2090 static int 2091 audiommap(struct file *fp, off_t *offp, size_t len, int prot, int *flagsp, 2092 int *advicep, struct uvm_object **uobjp, int *maxprotp) 2093 { 2094 struct audio_softc *sc; 2095 struct psref sc_ref; 2096 audio_file_t *file; 2097 dev_t dev; 2098 int bound; 2099 int error; 2100 2101 KASSERT(fp->f_audioctx); 2102 file = fp->f_audioctx; 2103 dev = file->dev; 2104 2105 bound = curlwp_bind(); 2106 sc = audio_sc_acquire_fromfile(file, &sc_ref); 2107 if (sc == NULL) { 2108 error = EIO; 2109 goto done; 2110 } 2111 2112 mutex_enter(sc->sc_lock); 2113 device_active(sc->sc_dev, DVA_SYSTEM); /* XXXJDM */ 2114 mutex_exit(sc->sc_lock); 2115 2116 switch (AUDIODEV(dev)) { 2117 case SOUND_DEVICE: 2118 case AUDIO_DEVICE: 2119 error = audio_mmap(sc, offp, len, prot, flagsp, advicep, 2120 uobjp, maxprotp, file); 2121 break; 2122 case AUDIOCTL_DEVICE: 2123 case MIXER_DEVICE: 2124 default: 2125 error = ENOTSUP; 2126 break; 2127 } 2128 2129 audio_sc_release(sc, &sc_ref); 2130 done: 2131 curlwp_bindx(bound); 2132 return error; 2133 } 2134 2135 2136 /* Exported interfaces for audiobell. */ 2137 2138 /* 2139 * Open for audiobell. 2140 * It stores allocated file to *filep. 2141 * If successful returns 0, otherwise errno. 2142 */ 2143 int 2144 audiobellopen(dev_t dev, audio_file_t **filep) 2145 { 2146 device_t audiodev = NULL; 2147 struct audio_softc *sc; 2148 bool exlock = false; 2149 int error; 2150 2151 /* 2152 * Find the autoconf instance and make sure it doesn't go away 2153 * while we are opening it. 2154 */ 2155 audiodev = device_lookup_acquire(&audio_cd, AUDIOUNIT(dev)); 2156 if (audiodev == NULL) { 2157 error = ENXIO; 2158 goto out; 2159 } 2160 2161 /* If attach failed, it's hopeless -- give up. */ 2162 sc = device_private(audiodev); 2163 if (sc->hw_if == NULL) { 2164 error = ENXIO; 2165 goto out; 2166 } 2167 2168 /* Take the exclusive configuration lock. */ 2169 error = audio_exlock_enter(sc); 2170 if (error) 2171 goto out; 2172 exlock = true; 2173 2174 /* Open the audio device. */ 2175 device_active(sc->sc_dev, DVA_SYSTEM); 2176 error = audio_open(dev, sc, FWRITE, 0, curlwp, filep); 2177 2178 out: if (exlock) 2179 audio_exlock_exit(sc); 2180 if (audiodev) 2181 device_release(audiodev); 2182 return error; 2183 } 2184 2185 /* Close for audiobell */ 2186 int 2187 audiobellclose(audio_file_t *file) 2188 { 2189 struct audio_softc *sc; 2190 struct psref sc_ref; 2191 int bound; 2192 int error; 2193 2194 error = 0; 2195 /* 2196 * audiobellclose() must 2197 * - unplug track from the trackmixer if sc exist. 2198 * - free all memory objects, regardless of sc. 2199 */ 2200 bound = curlwp_bind(); 2201 sc = audio_sc_acquire_fromfile(file, &sc_ref); 2202 if (sc) { 2203 error = audio_close(sc, file); 2204 audio_sc_release(sc, &sc_ref); 2205 } 2206 curlwp_bindx(bound); 2207 2208 /* Free memory objects anyway */ 2209 KASSERT(file->ptrack); 2210 audio_track_destroy(file->ptrack); 2211 KASSERT(file->rtrack == NULL); 2212 kmem_free(file, sizeof(*file)); 2213 return error; 2214 } 2215 2216 /* Set sample rate for audiobell */ 2217 int 2218 audiobellsetrate(audio_file_t *file, u_int sample_rate) 2219 { 2220 struct audio_softc *sc; 2221 struct psref sc_ref; 2222 struct audio_info ai; 2223 int bound; 2224 int error; 2225 2226 bound = curlwp_bind(); 2227 sc = audio_sc_acquire_fromfile(file, &sc_ref); 2228 if (sc == NULL) { 2229 error = EIO; 2230 goto done1; 2231 } 2232 2233 AUDIO_INITINFO(&ai); 2234 ai.play.sample_rate = sample_rate; 2235 2236 error = audio_exlock_enter(sc); 2237 if (error) 2238 goto done2; 2239 error = audio_file_setinfo(sc, file, &ai); 2240 audio_exlock_exit(sc); 2241 2242 done2: 2243 audio_sc_release(sc, &sc_ref); 2244 done1: 2245 curlwp_bindx(bound); 2246 return error; 2247 } 2248 2249 /* Playback for audiobell */ 2250 int 2251 audiobellwrite(audio_file_t *file, struct uio *uio) 2252 { 2253 struct audio_softc *sc; 2254 struct psref sc_ref; 2255 int bound; 2256 int error; 2257 2258 bound = curlwp_bind(); 2259 sc = audio_sc_acquire_fromfile(file, &sc_ref); 2260 if (sc == NULL) { 2261 error = EIO; 2262 goto done; 2263 } 2264 2265 error = audio_write(sc, uio, 0, file); 2266 2267 audio_sc_release(sc, &sc_ref); 2268 done: 2269 curlwp_bindx(bound); 2270 return error; 2271 } 2272 2273 2274 /* 2275 * Audio driver 2276 */ 2277 2278 /* 2279 * Must be called with sc_exlock held and without sc_lock held. 2280 */ 2281 int 2282 audio_open(dev_t dev, struct audio_softc *sc, int flags, int ifmt, 2283 struct lwp *l, audio_file_t **bellfile) 2284 { 2285 struct audio_info ai; 2286 struct file *fp; 2287 audio_file_t *af; 2288 audio_ring_t *hwbuf; 2289 bool fullduplex; 2290 bool cred_held; 2291 bool hw_opened; 2292 bool rmixer_started; 2293 bool inserted; 2294 int fd; 2295 int error; 2296 2297 KASSERT(sc->sc_exlock); 2298 2299 TRACE(1, "%sdev=%s flags=0x%x po=%d ro=%d", 2300 (audiodebug >= 3) ? "start " : "", 2301 ISDEVSOUND(dev) ? "sound" : "audio", 2302 flags, sc->sc_popens, sc->sc_ropens); 2303 2304 fp = NULL; 2305 cred_held = false; 2306 hw_opened = false; 2307 rmixer_started = false; 2308 inserted = false; 2309 2310 af = kmem_zalloc(sizeof(*af), KM_SLEEP); 2311 af->sc = sc; 2312 af->dev = dev; 2313 if ((flags & FWRITE) != 0 && audio_can_playback(sc)) 2314 af->mode |= AUMODE_PLAY | AUMODE_PLAY_ALL; 2315 if ((flags & FREAD) != 0 && audio_can_capture(sc)) 2316 af->mode |= AUMODE_RECORD; 2317 if (af->mode == 0) { 2318 error = ENXIO; 2319 goto bad; 2320 } 2321 2322 fullduplex = (sc->sc_props & AUDIO_PROP_FULLDUPLEX); 2323 2324 /* 2325 * On half duplex hardware, 2326 * 1. if mode is (PLAY | REC), let mode PLAY. 2327 * 2. if mode is PLAY, let mode PLAY if no rec tracks, otherwise error. 2328 * 3. if mode is REC, let mode REC if no play tracks, otherwise error. 2329 */ 2330 if (fullduplex == false) { 2331 if ((af->mode & AUMODE_PLAY)) { 2332 if (sc->sc_ropens != 0) { 2333 TRACE(1, "record track already exists"); 2334 error = ENODEV; 2335 goto bad; 2336 } 2337 /* Play takes precedence */ 2338 af->mode &= ~AUMODE_RECORD; 2339 } 2340 if ((af->mode & AUMODE_RECORD)) { 2341 if (sc->sc_popens != 0) { 2342 TRACE(1, "play track already exists"); 2343 error = ENODEV; 2344 goto bad; 2345 } 2346 } 2347 } 2348 2349 /* Create tracks */ 2350 if ((af->mode & AUMODE_PLAY)) 2351 af->ptrack = audio_track_create(sc, sc->sc_pmixer); 2352 if ((af->mode & AUMODE_RECORD)) 2353 af->rtrack = audio_track_create(sc, sc->sc_rmixer); 2354 2355 /* Set parameters */ 2356 AUDIO_INITINFO(&ai); 2357 if (bellfile) { 2358 /* If audiobell, only sample_rate will be set later. */ 2359 ai.play.sample_rate = audio_default.sample_rate; 2360 ai.play.encoding = AUDIO_ENCODING_SLINEAR_NE; 2361 ai.play.channels = 1; 2362 ai.play.precision = 16; 2363 ai.play.pause = 0; 2364 } else if (ISDEVAUDIO(dev)) { 2365 /* If /dev/audio, initialize everytime. */ 2366 ai.play.sample_rate = audio_default.sample_rate; 2367 ai.play.encoding = audio_default.encoding; 2368 ai.play.channels = audio_default.channels; 2369 ai.play.precision = audio_default.precision; 2370 ai.play.pause = 0; 2371 ai.record.sample_rate = audio_default.sample_rate; 2372 ai.record.encoding = audio_default.encoding; 2373 ai.record.channels = audio_default.channels; 2374 ai.record.precision = audio_default.precision; 2375 ai.record.pause = 0; 2376 } else { 2377 /* If /dev/sound, take over the previous parameters. */ 2378 ai.play.sample_rate = sc->sc_sound_pparams.sample_rate; 2379 ai.play.encoding = sc->sc_sound_pparams.encoding; 2380 ai.play.channels = sc->sc_sound_pparams.channels; 2381 ai.play.precision = sc->sc_sound_pparams.precision; 2382 ai.play.pause = sc->sc_sound_ppause; 2383 ai.record.sample_rate = sc->sc_sound_rparams.sample_rate; 2384 ai.record.encoding = sc->sc_sound_rparams.encoding; 2385 ai.record.channels = sc->sc_sound_rparams.channels; 2386 ai.record.precision = sc->sc_sound_rparams.precision; 2387 ai.record.pause = sc->sc_sound_rpause; 2388 } 2389 error = audio_file_setinfo(sc, af, &ai); 2390 if (error) 2391 goto bad; 2392 2393 if (sc->sc_popens + sc->sc_ropens == 0) { 2394 /* First open */ 2395 2396 sc->sc_cred = kauth_cred_get(); 2397 kauth_cred_hold(sc->sc_cred); 2398 cred_held = true; 2399 2400 if (sc->hw_if->open) { 2401 int hwflags; 2402 2403 /* 2404 * Call hw_if->open() only at first open of 2405 * combination of playback and recording. 2406 * On full duplex hardware, the flags passed to 2407 * hw_if->open() is always (FREAD | FWRITE) 2408 * regardless of this open()'s flags. 2409 * see also dev/isa/aria.c 2410 * On half duplex hardware, the flags passed to 2411 * hw_if->open() is either FREAD or FWRITE. 2412 * see also arch/evbarm/mini2440/audio_mini2440.c 2413 */ 2414 if (fullduplex) { 2415 hwflags = FREAD | FWRITE; 2416 } else { 2417 /* Construct hwflags from af->mode. */ 2418 hwflags = 0; 2419 if ((af->mode & AUMODE_PLAY) != 0) 2420 hwflags |= FWRITE; 2421 if ((af->mode & AUMODE_RECORD) != 0) 2422 hwflags |= FREAD; 2423 } 2424 2425 mutex_enter(sc->sc_lock); 2426 mutex_enter(sc->sc_intr_lock); 2427 error = sc->hw_if->open(sc->hw_hdl, hwflags); 2428 mutex_exit(sc->sc_intr_lock); 2429 mutex_exit(sc->sc_lock); 2430 if (error) 2431 goto bad; 2432 } 2433 /* 2434 * Regardless of whether we called hw_if->open (whether 2435 * hw_if->open exists) or not, we move to the Opened phase 2436 * here. Therefore from this point, we have to call 2437 * hw_if->close (if exists) whenever abort. 2438 * Note that both of hw_if->{open,close} are optional. 2439 */ 2440 hw_opened = true; 2441 2442 /* 2443 * Set speaker mode when a half duplex. 2444 * XXX I'm not sure this is correct. 2445 */ 2446 if (1/*XXX*/) { 2447 if (sc->hw_if->speaker_ctl) { 2448 int on; 2449 if (af->ptrack) { 2450 on = 1; 2451 } else { 2452 on = 0; 2453 } 2454 mutex_enter(sc->sc_lock); 2455 mutex_enter(sc->sc_intr_lock); 2456 error = sc->hw_if->speaker_ctl(sc->hw_hdl, on); 2457 mutex_exit(sc->sc_intr_lock); 2458 mutex_exit(sc->sc_lock); 2459 if (error) 2460 goto bad; 2461 } 2462 } 2463 } else if (sc->sc_multiuser == false) { 2464 uid_t euid = kauth_cred_geteuid(kauth_cred_get()); 2465 if (euid != 0 && euid != kauth_cred_geteuid(sc->sc_cred)) { 2466 error = EPERM; 2467 goto bad; 2468 } 2469 } 2470 2471 /* Call init_output if this is the first playback open. */ 2472 if (af->ptrack && sc->sc_popens == 0) { 2473 if (sc->hw_if->init_output) { 2474 hwbuf = &sc->sc_pmixer->hwbuf; 2475 mutex_enter(sc->sc_lock); 2476 mutex_enter(sc->sc_intr_lock); 2477 error = sc->hw_if->init_output(sc->hw_hdl, 2478 hwbuf->mem, 2479 hwbuf->capacity * 2480 hwbuf->fmt.channels * hwbuf->fmt.stride / NBBY); 2481 mutex_exit(sc->sc_intr_lock); 2482 mutex_exit(sc->sc_lock); 2483 if (error) 2484 goto bad; 2485 } 2486 } 2487 /* 2488 * Call init_input and start rmixer, if this is the first recording 2489 * open. See pause consideration notes. 2490 */ 2491 if (af->rtrack && sc->sc_ropens == 0) { 2492 if (sc->hw_if->init_input) { 2493 hwbuf = &sc->sc_rmixer->hwbuf; 2494 mutex_enter(sc->sc_lock); 2495 mutex_enter(sc->sc_intr_lock); 2496 error = sc->hw_if->init_input(sc->hw_hdl, 2497 hwbuf->mem, 2498 hwbuf->capacity * 2499 hwbuf->fmt.channels * hwbuf->fmt.stride / NBBY); 2500 mutex_exit(sc->sc_intr_lock); 2501 mutex_exit(sc->sc_lock); 2502 if (error) 2503 goto bad; 2504 } 2505 2506 mutex_enter(sc->sc_lock); 2507 audio_rmixer_start(sc); 2508 mutex_exit(sc->sc_lock); 2509 rmixer_started = true; 2510 } 2511 2512 /* 2513 * This is the last sc_lock section in the function, so we have to 2514 * examine sc_dying again before starting the rest tasks. Because 2515 * audiodeatch() may have been invoked (and it would set sc_dying) 2516 * from the time audioopen() was executed until now. If it happens, 2517 * audiodetach() may already have set file->dying for all sc_files 2518 * that exist at that point, so that audioopen() must abort without 2519 * inserting af to sc_files, in order to keep consistency. 2520 */ 2521 mutex_enter(sc->sc_lock); 2522 if (sc->sc_dying) { 2523 mutex_exit(sc->sc_lock); 2524 error = ENXIO; 2525 goto bad; 2526 } 2527 2528 /* Count up finally */ 2529 if (af->ptrack) 2530 sc->sc_popens++; 2531 if (af->rtrack) 2532 sc->sc_ropens++; 2533 mutex_enter(sc->sc_intr_lock); 2534 SLIST_INSERT_HEAD(&sc->sc_files, af, entry); 2535 mutex_exit(sc->sc_intr_lock); 2536 mutex_exit(sc->sc_lock); 2537 inserted = true; 2538 2539 if (bellfile) { 2540 *bellfile = af; 2541 } else { 2542 error = fd_allocfile(&fp, &fd); 2543 if (error) 2544 goto bad; 2545 2546 error = fd_clone(fp, fd, flags, &audio_fileops, af); 2547 KASSERTMSG(error == EMOVEFD, "error=%d", error); 2548 } 2549 2550 /* Be nothing else after fd_clone */ 2551 2552 TRACEF(3, af, "done"); 2553 return error; 2554 2555 bad: 2556 if (inserted) { 2557 mutex_enter(sc->sc_lock); 2558 mutex_enter(sc->sc_intr_lock); 2559 SLIST_REMOVE(&sc->sc_files, af, audio_file, entry); 2560 mutex_exit(sc->sc_intr_lock); 2561 if (af->ptrack) 2562 sc->sc_popens--; 2563 if (af->rtrack) 2564 sc->sc_ropens--; 2565 mutex_exit(sc->sc_lock); 2566 } 2567 2568 if (rmixer_started) { 2569 mutex_enter(sc->sc_lock); 2570 audio_rmixer_halt(sc); 2571 mutex_exit(sc->sc_lock); 2572 } 2573 2574 if (hw_opened) { 2575 if (sc->hw_if->close) { 2576 mutex_enter(sc->sc_lock); 2577 mutex_enter(sc->sc_intr_lock); 2578 sc->hw_if->close(sc->hw_hdl); 2579 mutex_exit(sc->sc_intr_lock); 2580 mutex_exit(sc->sc_lock); 2581 } 2582 } 2583 if (cred_held) { 2584 kauth_cred_free(sc->sc_cred); 2585 } 2586 2587 /* 2588 * Since track here is not yet linked to sc_files, 2589 * you can call track_destroy() without sc_intr_lock. 2590 */ 2591 if (af->rtrack) { 2592 audio_track_destroy(af->rtrack); 2593 af->rtrack = NULL; 2594 } 2595 if (af->ptrack) { 2596 audio_track_destroy(af->ptrack); 2597 af->ptrack = NULL; 2598 } 2599 2600 kmem_free(af, sizeof(*af)); 2601 return error; 2602 } 2603 2604 /* 2605 * Must be called without sc_lock nor sc_exlock held. 2606 */ 2607 int 2608 audio_close(struct audio_softc *sc, audio_file_t *file) 2609 { 2610 int error; 2611 2612 /* 2613 * Drain first. 2614 * It must be done before unlinking(acquiring exlock). 2615 */ 2616 if (file->ptrack) { 2617 mutex_enter(sc->sc_lock); 2618 audio_track_drain(sc, file->ptrack); 2619 mutex_exit(sc->sc_lock); 2620 } 2621 2622 mutex_enter(sc->sc_lock); 2623 mutex_enter(sc->sc_intr_lock); 2624 SLIST_REMOVE(&sc->sc_files, file, audio_file, entry); 2625 mutex_exit(sc->sc_intr_lock); 2626 mutex_exit(sc->sc_lock); 2627 2628 error = audio_exlock_enter(sc); 2629 if (error) { 2630 /* 2631 * If EIO, this sc is about to detach. In this case, even if 2632 * we don't do subsequent _unlink(), audiodetach() will do it. 2633 */ 2634 if (error == EIO) 2635 return error; 2636 2637 /* XXX This should not happen but what should I do ? */ 2638 panic("%s: can't acquire exlock: errno=%d", __func__, error); 2639 } 2640 audio_unlink(sc, file); 2641 audio_exlock_exit(sc); 2642 2643 return 0; 2644 } 2645 2646 /* 2647 * Unlink this file, but not freeing memory here. 2648 * Must be called with sc_exlock held and without sc_lock held. 2649 */ 2650 static void 2651 audio_unlink(struct audio_softc *sc, audio_file_t *file) 2652 { 2653 kauth_cred_t cred = NULL; 2654 int error; 2655 2656 mutex_enter(sc->sc_lock); 2657 2658 TRACEF(1, file, "%spid=%d.%d po=%d ro=%d", 2659 (audiodebug >= 3) ? "start " : "", 2660 (int)curproc->p_pid, (int)curlwp->l_lid, 2661 sc->sc_popens, sc->sc_ropens); 2662 KASSERTMSG(sc->sc_popens + sc->sc_ropens > 0, 2663 "sc->sc_popens=%d, sc->sc_ropens=%d", 2664 sc->sc_popens, sc->sc_ropens); 2665 2666 device_active(sc->sc_dev, DVA_SYSTEM); 2667 2668 if (file->ptrack) { 2669 TRACET(3, file->ptrack, "dropframes=%" PRIu64, 2670 file->ptrack->dropframes); 2671 2672 KASSERT(sc->sc_popens > 0); 2673 sc->sc_popens--; 2674 2675 /* Call hw halt_output if this is the last playback track. */ 2676 if (sc->sc_popens == 0 && sc->sc_pbusy) { 2677 error = audio_pmixer_halt(sc); 2678 if (error) { 2679 audio_printf(sc, 2680 "halt_output failed: errno=%d (ignored)\n", 2681 error); 2682 } 2683 } 2684 2685 /* Restore mixing volume if all tracks are gone. */ 2686 if (sc->sc_popens == 0) { 2687 /* intr_lock is not necessary, but just manners. */ 2688 mutex_enter(sc->sc_intr_lock); 2689 sc->sc_pmixer->volume = 256; 2690 sc->sc_pmixer->voltimer = 0; 2691 mutex_exit(sc->sc_intr_lock); 2692 } 2693 } 2694 if (file->rtrack) { 2695 TRACET(3, file->rtrack, "dropframes=%" PRIu64, 2696 file->rtrack->dropframes); 2697 2698 KASSERT(sc->sc_ropens > 0); 2699 sc->sc_ropens--; 2700 2701 /* Call hw halt_input if this is the last recording track. */ 2702 if (sc->sc_ropens == 0 && sc->sc_rbusy) { 2703 error = audio_rmixer_halt(sc); 2704 if (error) { 2705 audio_printf(sc, 2706 "halt_input failed: errno=%d (ignored)\n", 2707 error); 2708 } 2709 } 2710 2711 } 2712 2713 /* Call hw close if this is the last track. */ 2714 if (sc->sc_popens + sc->sc_ropens == 0) { 2715 if (sc->hw_if->close) { 2716 TRACE(2, "hw_if close"); 2717 mutex_enter(sc->sc_intr_lock); 2718 sc->hw_if->close(sc->hw_hdl); 2719 mutex_exit(sc->sc_intr_lock); 2720 } 2721 cred = sc->sc_cred; 2722 sc->sc_cred = NULL; 2723 } 2724 2725 mutex_exit(sc->sc_lock); 2726 if (cred) 2727 kauth_cred_free(cred); 2728 2729 TRACE(3, "done"); 2730 } 2731 2732 /* 2733 * Must be called without sc_lock nor sc_exlock held. 2734 */ 2735 int 2736 audio_read(struct audio_softc *sc, struct uio *uio, int ioflag, 2737 audio_file_t *file) 2738 { 2739 audio_track_t *track; 2740 audio_ring_t *usrbuf; 2741 audio_ring_t *input; 2742 int error; 2743 2744 /* 2745 * On half-duplex hardware, O_RDWR is treated as O_WRONLY. 2746 * However read() system call itself can be called because it's 2747 * opened with O_RDWR. So in this case, deny this read(). 2748 */ 2749 track = file->rtrack; 2750 if (track == NULL) { 2751 return EBADF; 2752 } 2753 2754 /* I think it's better than EINVAL. */ 2755 if (track->mmapped) 2756 return EPERM; 2757 2758 TRACET(2, track, "resid=%zd ioflag=0x%x", uio->uio_resid, ioflag); 2759 2760 #ifdef AUDIO_PM_IDLE 2761 error = audio_exlock_mutex_enter(sc); 2762 if (error) 2763 return error; 2764 2765 if (device_is_active(&sc->sc_dev) || sc->sc_idle) 2766 device_active(&sc->sc_dev, DVA_SYSTEM); 2767 2768 /* In recording, unlike playback, read() never operates rmixer. */ 2769 2770 audio_exlock_mutex_exit(sc); 2771 #endif 2772 2773 usrbuf = &track->usrbuf; 2774 input = track->input; 2775 error = 0; 2776 2777 while (uio->uio_resid > 0 && error == 0) { 2778 int bytes; 2779 2780 TRACET(3, track, 2781 "while resid=%zd input=%d/%d/%d usrbuf=%d/%d/C%d", 2782 uio->uio_resid, 2783 input->head, input->used, input->capacity, 2784 usrbuf->head, usrbuf->used, usrbuf->capacity); 2785 2786 /* Wait when buffers are empty. */ 2787 mutex_enter(sc->sc_lock); 2788 for (;;) { 2789 bool empty; 2790 audio_track_lock_enter(track); 2791 empty = (input->used == 0 && usrbuf->used == 0); 2792 audio_track_lock_exit(track); 2793 if (!empty) 2794 break; 2795 2796 if ((ioflag & IO_NDELAY)) { 2797 mutex_exit(sc->sc_lock); 2798 return EWOULDBLOCK; 2799 } 2800 2801 TRACET(3, track, "sleep"); 2802 error = audio_track_waitio(sc, track); 2803 if (error) { 2804 mutex_exit(sc->sc_lock); 2805 return error; 2806 } 2807 } 2808 mutex_exit(sc->sc_lock); 2809 2810 audio_track_lock_enter(track); 2811 /* Convert one block if possible. */ 2812 if (usrbuf->used == 0 && input->used > 0) { 2813 audio_track_record(track); 2814 } 2815 2816 /* uiomove from usrbuf as many bytes as possible. */ 2817 bytes = uimin(usrbuf->used, uio->uio_resid); 2818 error = uiomove((uint8_t *)usrbuf->mem + usrbuf->head, bytes, 2819 uio); 2820 if (error) { 2821 audio_track_lock_exit(track); 2822 device_printf(sc->sc_dev, 2823 "%s: uiomove(%d) failed: errno=%d\n", 2824 __func__, bytes, error); 2825 goto abort; 2826 } 2827 auring_take(usrbuf, bytes); 2828 TRACET(3, track, "uiomove(len=%d) usrbuf=%d/%d/C%d", 2829 bytes, 2830 usrbuf->head, usrbuf->used, usrbuf->capacity); 2831 2832 audio_track_lock_exit(track); 2833 } 2834 2835 abort: 2836 return error; 2837 } 2838 2839 2840 /* 2841 * Clear file's playback and/or record track buffer immediately. 2842 */ 2843 static void 2844 audio_file_clear(struct audio_softc *sc, audio_file_t *file) 2845 { 2846 2847 if (file->ptrack) 2848 audio_track_clear(sc, file->ptrack); 2849 if (file->rtrack) 2850 audio_track_clear(sc, file->rtrack); 2851 } 2852 2853 /* 2854 * Must be called without sc_lock nor sc_exlock held. 2855 */ 2856 int 2857 audio_write(struct audio_softc *sc, struct uio *uio, int ioflag, 2858 audio_file_t *file) 2859 { 2860 audio_track_t *track; 2861 audio_ring_t *usrbuf; 2862 audio_ring_t *outbuf; 2863 int error; 2864 2865 track = file->ptrack; 2866 if (track == NULL) 2867 return EPERM; 2868 2869 /* I think it's better than EINVAL. */ 2870 if (track->mmapped) 2871 return EPERM; 2872 2873 TRACET(2, track, "%sresid=%zd pid=%d.%d ioflag=0x%x", 2874 audiodebug >= 3 ? "begin " : "", 2875 uio->uio_resid, (int)curproc->p_pid, (int)curlwp->l_lid, ioflag); 2876 2877 if (uio->uio_resid == 0) { 2878 track->eofcounter++; 2879 return 0; 2880 } 2881 2882 error = audio_exlock_mutex_enter(sc); 2883 if (error) 2884 return error; 2885 2886 #ifdef AUDIO_PM_IDLE 2887 if (device_is_active(&sc->sc_dev) || sc->sc_idle) 2888 device_active(&sc->sc_dev, DVA_SYSTEM); 2889 #endif 2890 2891 /* 2892 * The first write starts pmixer. 2893 */ 2894 if (sc->sc_pbusy == false) 2895 audio_pmixer_start(sc, false); 2896 audio_exlock_mutex_exit(sc); 2897 2898 usrbuf = &track->usrbuf; 2899 outbuf = &track->outbuf; 2900 track->pstate = AUDIO_STATE_RUNNING; 2901 error = 0; 2902 2903 while (uio->uio_resid > 0 && error == 0) { 2904 int bytes; 2905 2906 TRACET(3, track, "while resid=%zd usrbuf=%d/%d/H%d", 2907 uio->uio_resid, 2908 usrbuf->head, usrbuf->used, track->usrbuf_usedhigh); 2909 2910 /* Wait when buffers are full. */ 2911 mutex_enter(sc->sc_lock); 2912 for (;;) { 2913 bool full; 2914 audio_track_lock_enter(track); 2915 full = (usrbuf->used >= track->usrbuf_usedhigh && 2916 outbuf->used >= outbuf->capacity); 2917 audio_track_lock_exit(track); 2918 if (!full) 2919 break; 2920 2921 if ((ioflag & IO_NDELAY)) { 2922 error = EWOULDBLOCK; 2923 mutex_exit(sc->sc_lock); 2924 goto abort; 2925 } 2926 2927 TRACET(3, track, "sleep usrbuf=%d/H%d", 2928 usrbuf->used, track->usrbuf_usedhigh); 2929 error = audio_track_waitio(sc, track); 2930 if (error) { 2931 mutex_exit(sc->sc_lock); 2932 goto abort; 2933 } 2934 } 2935 mutex_exit(sc->sc_lock); 2936 2937 audio_track_lock_enter(track); 2938 2939 /* uiomove to usrbuf as many bytes as possible. */ 2940 bytes = uimin(track->usrbuf_usedhigh - usrbuf->used, 2941 uio->uio_resid); 2942 while (bytes > 0) { 2943 int tail = auring_tail(usrbuf); 2944 int len = uimin(bytes, usrbuf->capacity - tail); 2945 error = uiomove((uint8_t *)usrbuf->mem + tail, len, 2946 uio); 2947 if (error) { 2948 audio_track_lock_exit(track); 2949 device_printf(sc->sc_dev, 2950 "%s: uiomove(%d) failed: errno=%d\n", 2951 __func__, len, error); 2952 goto abort; 2953 } 2954 auring_push(usrbuf, len); 2955 TRACET(3, track, "uiomove(len=%d) usrbuf=%d/%d/C%d", 2956 len, 2957 usrbuf->head, usrbuf->used, usrbuf->capacity); 2958 bytes -= len; 2959 } 2960 2961 /* Convert them as many blocks as possible. */ 2962 while (usrbuf->used >= track->usrbuf_blksize && 2963 outbuf->used < outbuf->capacity) { 2964 audio_track_play(track); 2965 } 2966 2967 audio_track_lock_exit(track); 2968 } 2969 2970 abort: 2971 TRACET(3, track, "done error=%d", error); 2972 return error; 2973 } 2974 2975 /* 2976 * Must be called without sc_lock nor sc_exlock held. 2977 */ 2978 int 2979 audio_ioctl(dev_t dev, struct audio_softc *sc, u_long cmd, void *addr, int flag, 2980 struct lwp *l, audio_file_t *file) 2981 { 2982 struct audio_offset *ao; 2983 struct audio_info ai; 2984 audio_track_t *track; 2985 audio_encoding_t *ae; 2986 audio_format_query_t *query; 2987 u_int stamp; 2988 u_int offset; 2989 int val; 2990 int index; 2991 int error; 2992 2993 #if defined(AUDIO_DEBUG) 2994 const char *ioctlnames[] = { 2995 "AUDIO_GETINFO", /* 21 */ 2996 "AUDIO_SETINFO", /* 22 */ 2997 "AUDIO_DRAIN", /* 23 */ 2998 "AUDIO_FLUSH", /* 24 */ 2999 "AUDIO_WSEEK", /* 25 */ 3000 "AUDIO_RERROR", /* 26 */ 3001 "AUDIO_GETDEV", /* 27 */ 3002 "AUDIO_GETENC", /* 28 */ 3003 "AUDIO_GETFD", /* 29 */ 3004 "AUDIO_SETFD", /* 30 */ 3005 "AUDIO_PERROR", /* 31 */ 3006 "AUDIO_GETIOFFS", /* 32 */ 3007 "AUDIO_GETOOFFS", /* 33 */ 3008 "AUDIO_GETPROPS", /* 34 */ 3009 "AUDIO_GETBUFINFO", /* 35 */ 3010 "AUDIO_SETCHAN", /* 36 */ 3011 "AUDIO_GETCHAN", /* 37 */ 3012 "AUDIO_QUERYFORMAT", /* 38 */ 3013 "AUDIO_GETFORMAT", /* 39 */ 3014 "AUDIO_SETFORMAT", /* 40 */ 3015 }; 3016 char pre[64]; 3017 int nameidx = (cmd & 0xff); 3018 if (21 <= nameidx && nameidx <= 21 + __arraycount(ioctlnames)) { 3019 snprintf(pre, sizeof(pre), "pid=%d.%d %s", 3020 (int)curproc->p_pid, (int)l->l_lid, 3021 ioctlnames[nameidx - 21]); 3022 } else { 3023 snprintf(pre, sizeof(pre), "pid=%d.%d (%lu,'%c',%u)", 3024 (int)curproc->p_pid, (int)l->l_lid, 3025 IOCPARM_LEN(cmd), (char)IOCGROUP(cmd), nameidx); 3026 } 3027 #endif 3028 3029 error = 0; 3030 switch (cmd) { 3031 case FIONBIO: 3032 /* All handled in the upper FS layer. */ 3033 break; 3034 3035 case FIONREAD: 3036 /* Get the number of bytes that can be read. */ 3037 track = file->rtrack; 3038 if (track) { 3039 val = audio_track_readablebytes(track); 3040 *(int *)addr = val; 3041 TRACET(2, track, "pid=%d.%d FIONREAD bytes=%d", 3042 (int)curproc->p_pid, (int)l->l_lid, val); 3043 } else { 3044 TRACEF(2, file, "pid=%d.%d FIONREAD no track", 3045 (int)curproc->p_pid, (int)l->l_lid); 3046 } 3047 break; 3048 3049 case FIOASYNC: 3050 /* Set/Clear ASYNC I/O. */ 3051 if (*(int *)addr) { 3052 file->async_audio = curproc->p_pid; 3053 } else { 3054 file->async_audio = 0; 3055 } 3056 TRACEF(2, file, "pid=%d.%d FIOASYNC %s", 3057 (int)curproc->p_pid, (int)l->l_lid, 3058 file->async_audio ? "on" : "off"); 3059 break; 3060 3061 case AUDIO_FLUSH: 3062 /* XXX TODO: clear errors and restart? */ 3063 TRACEF(2, file, "%s", pre); 3064 audio_file_clear(sc, file); 3065 break; 3066 3067 case AUDIO_PERROR: 3068 case AUDIO_RERROR: 3069 /* 3070 * Number of dropped bytes during playback/record. We don't 3071 * know where or when they were dropped (including conversion 3072 * stage). Therefore, the number of accurate bytes or samples 3073 * is also unknown. 3074 */ 3075 track = (cmd == AUDIO_PERROR) ? file->ptrack : file->rtrack; 3076 if (track) { 3077 val = frametobyte(&track->usrbuf.fmt, 3078 track->dropframes); 3079 *(int *)addr = val; 3080 TRACET(2, track, "%s bytes=%d", pre, val); 3081 } else { 3082 TRACEF(2, file, "%s no track", pre); 3083 } 3084 break; 3085 3086 case AUDIO_GETIOFFS: 3087 ao = (struct audio_offset *)addr; 3088 track = file->rtrack; 3089 if (track == NULL) { 3090 ao->samples = 0; 3091 ao->deltablks = 0; 3092 ao->offset = 0; 3093 TRACEF(2, file, "%s no rtrack", pre); 3094 break; 3095 } 3096 mutex_enter(sc->sc_lock); 3097 mutex_enter(sc->sc_intr_lock); 3098 /* figure out where next transfer will start */ 3099 stamp = track->stamp; 3100 offset = auring_tail(track->input); 3101 mutex_exit(sc->sc_intr_lock); 3102 mutex_exit(sc->sc_lock); 3103 3104 /* samples will overflow soon but is as per spec. */ 3105 ao->samples = stamp * track->usrbuf_blksize; 3106 ao->deltablks = stamp - track->last_stamp; 3107 ao->offset = audio_track_inputblk_as_usrbyte(track, offset); 3108 TRACET(2, track, "%s samples=%u deltablks=%u offset=%u", 3109 pre, ao->samples, ao->deltablks, ao->offset); 3110 3111 track->last_stamp = stamp; 3112 break; 3113 3114 case AUDIO_GETOOFFS: 3115 ao = (struct audio_offset *)addr; 3116 track = file->ptrack; 3117 if (track == NULL) { 3118 ao->samples = 0; 3119 ao->deltablks = 0; 3120 ao->offset = 0; 3121 TRACEF(2, file, "%s no ptrack", pre); 3122 break; 3123 } 3124 mutex_enter(sc->sc_lock); 3125 mutex_enter(sc->sc_intr_lock); 3126 /* figure out where next transfer will start */ 3127 stamp = track->stamp; 3128 offset = track->usrbuf.head; 3129 mutex_exit(sc->sc_intr_lock); 3130 mutex_exit(sc->sc_lock); 3131 3132 /* samples will overflow soon but is as per spec. */ 3133 ao->samples = stamp * track->usrbuf_blksize; 3134 ao->deltablks = stamp - track->last_stamp; 3135 ao->offset = offset; 3136 TRACET(2, track, "%s samples=%u deltablks=%u offset=%u", 3137 pre, ao->samples, ao->deltablks, ao->offset); 3138 3139 track->last_stamp = stamp; 3140 break; 3141 3142 case AUDIO_WSEEK: 3143 track = file->ptrack; 3144 if (track) { 3145 val = track->usrbuf.used; 3146 *(u_long *)addr = val; 3147 TRACET(2, track, "%s bytes=%d", pre, val); 3148 } else { 3149 TRACEF(2, file, "%s no ptrack", pre); 3150 } 3151 break; 3152 3153 case AUDIO_SETINFO: 3154 TRACEF(2, file, "%s", pre); 3155 error = audio_exlock_enter(sc); 3156 if (error) 3157 break; 3158 error = audio_file_setinfo(sc, file, (struct audio_info *)addr); 3159 if (error) { 3160 audio_exlock_exit(sc); 3161 break; 3162 } 3163 /* XXX TODO: update last_ai if /dev/sound ? */ 3164 if (ISDEVSOUND(dev)) 3165 error = audiogetinfo(sc, &sc->sc_ai, 0, file); 3166 audio_exlock_exit(sc); 3167 break; 3168 3169 case AUDIO_GETINFO: 3170 TRACEF(2, file, "%s", pre); 3171 error = audio_exlock_enter(sc); 3172 if (error) 3173 break; 3174 error = audiogetinfo(sc, (struct audio_info *)addr, 1, file); 3175 audio_exlock_exit(sc); 3176 break; 3177 3178 case AUDIO_GETBUFINFO: 3179 TRACEF(2, file, "%s", pre); 3180 error = audio_exlock_enter(sc); 3181 if (error) 3182 break; 3183 error = audiogetinfo(sc, (struct audio_info *)addr, 0, file); 3184 audio_exlock_exit(sc); 3185 break; 3186 3187 case AUDIO_DRAIN: 3188 track = file->ptrack; 3189 if (track) { 3190 TRACET(2, track, "%s", pre); 3191 mutex_enter(sc->sc_lock); 3192 error = audio_track_drain(sc, track); 3193 mutex_exit(sc->sc_lock); 3194 } else { 3195 TRACEF(2, file, "%s no ptrack", pre); 3196 } 3197 break; 3198 3199 case AUDIO_GETDEV: 3200 TRACEF(2, file, "%s", pre); 3201 error = sc->hw_if->getdev(sc->hw_hdl, (audio_device_t *)addr); 3202 break; 3203 3204 case AUDIO_GETENC: 3205 ae = (audio_encoding_t *)addr; 3206 index = ae->index; 3207 TRACEF(2, file, "%s index=%d", pre, index); 3208 if (index < 0 || index >= __arraycount(audio_encodings)) { 3209 error = EINVAL; 3210 break; 3211 } 3212 *ae = audio_encodings[index]; 3213 ae->index = index; 3214 /* 3215 * EMULATED always. 3216 * EMULATED flag at that time used to mean that it could 3217 * not be passed directly to the hardware as-is. But 3218 * currently, all formats including hardware native is not 3219 * passed directly to the hardware. So I set EMULATED 3220 * flag for all formats. 3221 */ 3222 ae->flags = AUDIO_ENCODINGFLAG_EMULATED; 3223 break; 3224 3225 case AUDIO_GETFD: 3226 /* 3227 * Returns the current setting of full duplex mode. 3228 * If HW has full duplex mode and there are two mixers, 3229 * it is full duplex. Otherwise half duplex. 3230 */ 3231 error = audio_exlock_enter(sc); 3232 if (error) 3233 break; 3234 val = (sc->sc_props & AUDIO_PROP_FULLDUPLEX) 3235 && (sc->sc_pmixer && sc->sc_rmixer); 3236 audio_exlock_exit(sc); 3237 *(int *)addr = val; 3238 TRACEF(2, file, "%s fulldup=%d", pre, val); 3239 break; 3240 3241 case AUDIO_GETPROPS: 3242 val = sc->sc_props; 3243 *(int *)addr = val; 3244 #if defined(AUDIO_DEBUG) 3245 char pbuf[64]; 3246 snprintb(pbuf, sizeof(pbuf), "\x10" 3247 "\6CAPTURE" "\5PLAY" "\3INDEP" "\2MMAP" "\1FULLDUP", val); 3248 TRACEF(2, file, "%s %s", pre, pbuf); 3249 #endif 3250 break; 3251 3252 case AUDIO_QUERYFORMAT: 3253 query = (audio_format_query_t *)addr; 3254 TRACEF(2, file, "%s index=%u", pre, query->index); 3255 mutex_enter(sc->sc_lock); 3256 error = sc->hw_if->query_format(sc->hw_hdl, query); 3257 mutex_exit(sc->sc_lock); 3258 /* Hide internal information */ 3259 query->fmt.driver_data = NULL; 3260 break; 3261 3262 case AUDIO_GETFORMAT: 3263 TRACEF(2, file, "%s", pre); 3264 error = audio_exlock_enter(sc); 3265 if (error) 3266 break; 3267 audio_mixers_get_format(sc, (struct audio_info *)addr); 3268 audio_exlock_exit(sc); 3269 break; 3270 3271 case AUDIO_SETFORMAT: 3272 TRACEF(2, file, "%s", pre); 3273 error = audio_exlock_enter(sc); 3274 audio_mixers_get_format(sc, &ai); 3275 error = audio_mixers_set_format(sc, (struct audio_info *)addr); 3276 if (error) { 3277 /* Rollback */ 3278 audio_mixers_set_format(sc, &ai); 3279 } 3280 audio_exlock_exit(sc); 3281 break; 3282 3283 case AUDIO_SETFD: 3284 case AUDIO_SETCHAN: 3285 case AUDIO_GETCHAN: 3286 /* Obsoleted */ 3287 TRACEF(2, file, "%s", pre); 3288 break; 3289 3290 default: 3291 TRACEF(2, file, "%s", pre); 3292 if (sc->hw_if->dev_ioctl) { 3293 mutex_enter(sc->sc_lock); 3294 error = sc->hw_if->dev_ioctl(sc->hw_hdl, 3295 cmd, addr, flag, l); 3296 mutex_exit(sc->sc_lock); 3297 } else { 3298 error = EINVAL; 3299 } 3300 break; 3301 } 3302 3303 if (error) 3304 TRACEF(2, file, "%s error=%d", pre, error); 3305 return error; 3306 } 3307 3308 /* 3309 * Convert n [frames] of the input buffer to bytes in the usrbuf format. 3310 * n is in frames but should be a multiple of frame/block. Note that the 3311 * usrbuf's frame/block and the input buffer's frame/block may be different 3312 * (i.e., if frequencies are different). 3313 * 3314 * This function is for recording track only. 3315 */ 3316 static int 3317 audio_track_inputblk_as_usrbyte(const audio_track_t *track, int n) 3318 { 3319 int input_fpb; 3320 3321 /* 3322 * In the input buffer on recording track, these are the same. 3323 * input_fpb = frame_per_block(track->mixer, &track->input->fmt); 3324 */ 3325 input_fpb = track->mixer->frames_per_block; 3326 3327 return (n / input_fpb) * track->usrbuf_blksize; 3328 } 3329 3330 /* 3331 * Returns the number of bytes that can be read on recording buffer. 3332 */ 3333 static int 3334 audio_track_readablebytes(const audio_track_t *track) 3335 { 3336 int bytes; 3337 3338 KASSERT(track); 3339 KASSERT(track->mode == AUMODE_RECORD); 3340 3341 /* 3342 * For recording, track->input is the main block-unit buffer and 3343 * track->usrbuf holds less than one block of byte data ("fragment"). 3344 * Note that the input buffer is in frames and the usrbuf is in bytes. 3345 * 3346 * Actual total capacity of these two buffers is 3347 * input->capacity [frames] + usrbuf.capacity [bytes], 3348 * but only input->capacity is reported to userland as buffer_size. 3349 * So, even if the total used bytes exceed input->capacity, report it 3350 * as input->capacity for consistency. 3351 */ 3352 bytes = audio_track_inputblk_as_usrbyte(track, track->input->used); 3353 if (track->input->used < track->input->capacity) { 3354 bytes += track->usrbuf.used; 3355 } 3356 return bytes; 3357 } 3358 3359 /* 3360 * Must be called without sc_lock nor sc_exlock held. 3361 */ 3362 int 3363 audio_poll(struct audio_softc *sc, int events, struct lwp *l, 3364 audio_file_t *file) 3365 { 3366 audio_track_t *track; 3367 int revents; 3368 bool in_is_valid; 3369 bool out_is_valid; 3370 3371 #if defined(AUDIO_DEBUG) 3372 #define POLLEV_BITMAP "\177\020" \ 3373 "b\10WRBAND\0" \ 3374 "b\7RDBAND\0" "b\6RDNORM\0" "b\5NVAL\0" "b\4HUP\0" \ 3375 "b\3ERR\0" "b\2OUT\0" "b\1PRI\0" "b\0IN\0" 3376 char evbuf[64]; 3377 snprintb(evbuf, sizeof(evbuf), POLLEV_BITMAP, events); 3378 TRACEF(2, file, "pid=%d.%d events=%s", 3379 (int)curproc->p_pid, (int)l->l_lid, evbuf); 3380 #endif 3381 3382 revents = 0; 3383 in_is_valid = false; 3384 out_is_valid = false; 3385 if (events & (POLLIN | POLLRDNORM)) { 3386 track = file->rtrack; 3387 if (track) { 3388 int used; 3389 in_is_valid = true; 3390 used = audio_track_readablebytes(track); 3391 if (used > 0) 3392 revents |= events & (POLLIN | POLLRDNORM); 3393 } 3394 } 3395 if (events & (POLLOUT | POLLWRNORM)) { 3396 track = file->ptrack; 3397 if (track) { 3398 out_is_valid = true; 3399 if (track->usrbuf.used <= track->usrbuf_usedlow) 3400 revents |= events & (POLLOUT | POLLWRNORM); 3401 } 3402 } 3403 3404 if (revents == 0) { 3405 mutex_enter(sc->sc_lock); 3406 if (in_is_valid) { 3407 TRACEF(3, file, "selrecord rsel"); 3408 selrecord(l, &sc->sc_rsel); 3409 } 3410 if (out_is_valid) { 3411 TRACEF(3, file, "selrecord wsel"); 3412 selrecord(l, &sc->sc_wsel); 3413 } 3414 mutex_exit(sc->sc_lock); 3415 } 3416 3417 #if defined(AUDIO_DEBUG) 3418 snprintb(evbuf, sizeof(evbuf), POLLEV_BITMAP, revents); 3419 TRACEF(2, file, "revents=%s", evbuf); 3420 #endif 3421 return revents; 3422 } 3423 3424 static const struct filterops audioread_filtops = { 3425 .f_flags = FILTEROP_ISFD, 3426 .f_attach = NULL, 3427 .f_detach = filt_audioread_detach, 3428 .f_event = filt_audioread_event, 3429 }; 3430 3431 static void 3432 filt_audioread_detach(struct knote *kn) 3433 { 3434 struct audio_softc *sc; 3435 audio_file_t *file; 3436 3437 file = kn->kn_hook; 3438 sc = file->sc; 3439 TRACEF(3, file, "called"); 3440 3441 mutex_enter(sc->sc_lock); 3442 selremove_knote(&sc->sc_rsel, kn); 3443 mutex_exit(sc->sc_lock); 3444 } 3445 3446 static int 3447 filt_audioread_event(struct knote *kn, long hint) 3448 { 3449 audio_file_t *file; 3450 audio_track_t *track; 3451 3452 file = kn->kn_hook; 3453 track = file->rtrack; 3454 3455 /* 3456 * kn_data must contain the number of bytes can be read. 3457 * The return value indicates whether the event occurs or not. 3458 */ 3459 3460 if (track == NULL) { 3461 /* can not read with this descriptor. */ 3462 kn->kn_data = 0; 3463 return 0; 3464 } 3465 3466 kn->kn_data = audio_track_readablebytes(track); 3467 TRACEF(3, file, "data=%" PRId64, kn->kn_data); 3468 return kn->kn_data > 0; 3469 } 3470 3471 static const struct filterops audiowrite_filtops = { 3472 .f_flags = FILTEROP_ISFD, 3473 .f_attach = NULL, 3474 .f_detach = filt_audiowrite_detach, 3475 .f_event = filt_audiowrite_event, 3476 }; 3477 3478 static void 3479 filt_audiowrite_detach(struct knote *kn) 3480 { 3481 struct audio_softc *sc; 3482 audio_file_t *file; 3483 3484 file = kn->kn_hook; 3485 sc = file->sc; 3486 TRACEF(3, file, "called"); 3487 3488 mutex_enter(sc->sc_lock); 3489 selremove_knote(&sc->sc_wsel, kn); 3490 mutex_exit(sc->sc_lock); 3491 } 3492 3493 static int 3494 filt_audiowrite_event(struct knote *kn, long hint) 3495 { 3496 audio_file_t *file; 3497 audio_track_t *track; 3498 3499 file = kn->kn_hook; 3500 track = file->ptrack; 3501 3502 /* 3503 * kn_data must contain the number of bytes can be write. 3504 * The return value indicates whether the event occurs or not. 3505 */ 3506 3507 if (track == NULL) { 3508 /* can not write with this descriptor. */ 3509 kn->kn_data = 0; 3510 return 0; 3511 } 3512 3513 kn->kn_data = track->usrbuf_usedhigh - track->usrbuf.used; 3514 TRACEF(3, file, "data=%" PRId64, kn->kn_data); 3515 return (track->usrbuf.used < track->usrbuf_usedlow); 3516 } 3517 3518 /* 3519 * Must be called without sc_lock nor sc_exlock held. 3520 */ 3521 int 3522 audio_kqfilter(struct audio_softc *sc, audio_file_t *file, struct knote *kn) 3523 { 3524 struct selinfo *sip; 3525 3526 TRACEF(3, file, "kn=%p kn_filter=%x", kn, (int)kn->kn_filter); 3527 3528 switch (kn->kn_filter) { 3529 case EVFILT_READ: 3530 sip = &sc->sc_rsel; 3531 kn->kn_fop = &audioread_filtops; 3532 break; 3533 3534 case EVFILT_WRITE: 3535 sip = &sc->sc_wsel; 3536 kn->kn_fop = &audiowrite_filtops; 3537 break; 3538 3539 default: 3540 return EINVAL; 3541 } 3542 3543 kn->kn_hook = file; 3544 3545 mutex_enter(sc->sc_lock); 3546 selrecord_knote(sip, kn); 3547 mutex_exit(sc->sc_lock); 3548 3549 return 0; 3550 } 3551 3552 /* 3553 * Must be called without sc_lock nor sc_exlock held. 3554 */ 3555 int 3556 audio_mmap(struct audio_softc *sc, off_t *offp, size_t len, int prot, 3557 int *flagsp, int *advicep, struct uvm_object **uobjp, int *maxprotp, 3558 audio_file_t *file) 3559 { 3560 audio_track_t *track; 3561 vsize_t vsize; 3562 int error; 3563 3564 TRACEF(2, file, "off=%lld, prot=%d", (long long)(*offp), prot); 3565 3566 if (*offp < 0) 3567 return EINVAL; 3568 3569 #if 0 3570 /* XXX 3571 * The idea here was to use the protection to determine if 3572 * we are mapping the read or write buffer, but it fails. 3573 * The VM system is broken in (at least) two ways. 3574 * 1) If you map memory VM_PROT_WRITE you SIGSEGV 3575 * when writing to it, so VM_PROT_READ|VM_PROT_WRITE 3576 * has to be used for mmapping the play buffer. 3577 * 2) Even if calling mmap() with VM_PROT_READ|VM_PROT_WRITE 3578 * audio_mmap will get called at some point with VM_PROT_READ 3579 * only. 3580 * So, alas, we always map the play buffer for now. 3581 */ 3582 if (prot == (VM_PROT_READ|VM_PROT_WRITE) || 3583 prot == VM_PROT_WRITE) 3584 track = file->ptrack; 3585 else if (prot == VM_PROT_READ) 3586 track = file->rtrack; 3587 else 3588 return EINVAL; 3589 #else 3590 track = file->ptrack; 3591 #endif 3592 if (track == NULL) 3593 return EACCES; 3594 3595 vsize = roundup2(MAX(track->usrbuf.capacity, PAGE_SIZE), PAGE_SIZE); 3596 if (len > vsize) 3597 return EOVERFLOW; 3598 if (*offp > (uint)(vsize - len)) 3599 return EOVERFLOW; 3600 3601 /* XXX TODO: what happens when mmap twice. */ 3602 if (!track->mmapped) { 3603 track->mmapped = true; 3604 3605 if (!track->is_pause) { 3606 error = audio_exlock_mutex_enter(sc); 3607 if (error) 3608 return error; 3609 if (sc->sc_pbusy == false) 3610 audio_pmixer_start(sc, true); 3611 audio_exlock_mutex_exit(sc); 3612 } 3613 /* XXX mmapping record buffer is not supported */ 3614 } 3615 3616 /* get ringbuffer */ 3617 *uobjp = track->uobj; 3618 3619 /* Acquire a reference for the mmap. munmap will release. */ 3620 uao_reference(*uobjp); 3621 *maxprotp = prot; 3622 *advicep = UVM_ADV_RANDOM; 3623 *flagsp = MAP_SHARED; 3624 return 0; 3625 } 3626 3627 /* 3628 * /dev/audioctl has to be able to open at any time without interference 3629 * with any /dev/audio or /dev/sound. 3630 * Must be called with sc_exlock held and without sc_lock held. 3631 */ 3632 static int 3633 audioctl_open(dev_t dev, struct audio_softc *sc, int flags, int ifmt, 3634 struct lwp *l) 3635 { 3636 struct file *fp; 3637 audio_file_t *af; 3638 int fd; 3639 int error; 3640 3641 KASSERT(sc->sc_exlock); 3642 3643 TRACE(1, "called"); 3644 3645 error = fd_allocfile(&fp, &fd); 3646 if (error) 3647 return error; 3648 3649 af = kmem_zalloc(sizeof(*af), KM_SLEEP); 3650 af->sc = sc; 3651 af->dev = dev; 3652 3653 mutex_enter(sc->sc_lock); 3654 if (sc->sc_dying) { 3655 mutex_exit(sc->sc_lock); 3656 kmem_free(af, sizeof(*af)); 3657 fd_abort(curproc, fp, fd); 3658 return ENXIO; 3659 } 3660 mutex_enter(sc->sc_intr_lock); 3661 SLIST_INSERT_HEAD(&sc->sc_files, af, entry); 3662 mutex_exit(sc->sc_intr_lock); 3663 mutex_exit(sc->sc_lock); 3664 3665 error = fd_clone(fp, fd, flags, &audio_fileops, af); 3666 KASSERTMSG(error == EMOVEFD, "error=%d", error); 3667 3668 return error; 3669 } 3670 3671 /* 3672 * Free 'mem' if available, and initialize the pointer. 3673 * For this reason, this is implemented as macro. 3674 */ 3675 #define audio_free(mem) do { \ 3676 if (mem != NULL) { \ 3677 kern_free(mem); \ 3678 mem = NULL; \ 3679 } \ 3680 } while (0) 3681 3682 /* 3683 * (Re)allocate 'memblock' with specified 'bytes'. 3684 * bytes must not be 0. 3685 * This function never returns NULL. 3686 */ 3687 static void * 3688 audio_realloc(void *memblock, size_t bytes) 3689 { 3690 3691 KASSERT(bytes != 0); 3692 if (memblock) 3693 kern_free(memblock); 3694 return kern_malloc(bytes, M_WAITOK); 3695 } 3696 3697 /* 3698 * (Re)allocate usrbuf with 'newbufsize' bytes. 3699 * Use this function for usrbuf because only usrbuf can be mmapped. 3700 * If successful, it updates track->usrbuf.mem, track->usrbuf.capacity and 3701 * returns 0. Otherwise, it clears track->usrbuf.mem, track->usrbuf.capacity 3702 * and returns errno. 3703 * It must be called before updating usrbuf.capacity. 3704 */ 3705 static int 3706 audio_realloc_usrbuf(audio_track_t *track, int newbufsize) 3707 { 3708 struct audio_softc *sc; 3709 vaddr_t vstart; 3710 vsize_t oldvsize; 3711 vsize_t newvsize; 3712 int error; 3713 3714 KASSERT(newbufsize > 0); 3715 sc = track->mixer->sc; 3716 3717 /* Get a nonzero multiple of PAGE_SIZE */ 3718 newvsize = roundup2(MAX(newbufsize, PAGE_SIZE), PAGE_SIZE); 3719 3720 if (track->usrbuf.mem != NULL) { 3721 oldvsize = roundup2(MAX(track->usrbuf.capacity, PAGE_SIZE), 3722 PAGE_SIZE); 3723 if (oldvsize == newvsize) { 3724 track->usrbuf.capacity = newbufsize; 3725 return 0; 3726 } 3727 vstart = (vaddr_t)track->usrbuf.mem; 3728 uvm_unmap(kernel_map, vstart, vstart + oldvsize); 3729 /* uvm_unmap also detach uobj */ 3730 track->uobj = NULL; /* paranoia */ 3731 track->usrbuf.mem = NULL; 3732 } 3733 3734 /* Create a uvm anonymous object */ 3735 track->uobj = uao_create(newvsize, 0); 3736 3737 /* Map it into the kernel virtual address space */ 3738 vstart = 0; 3739 error = uvm_map(kernel_map, &vstart, newvsize, track->uobj, 0, 0, 3740 UVM_MAPFLAG(UVM_PROT_RW, UVM_PROT_RW, UVM_INH_NONE, 3741 UVM_ADV_RANDOM, 0)); 3742 if (error) { 3743 device_printf(sc->sc_dev, "uvm_map failed: errno=%d\n", error); 3744 uao_detach(track->uobj); /* release reference */ 3745 goto abort; 3746 } 3747 3748 error = uvm_map_pageable(kernel_map, vstart, vstart + newvsize, 3749 false, 0); 3750 if (error) { 3751 device_printf(sc->sc_dev, "uvm_map_pageable failed: errno=%d\n", 3752 error); 3753 uvm_unmap(kernel_map, vstart, vstart + newvsize); 3754 /* uvm_unmap also detach uobj */ 3755 goto abort; 3756 } 3757 3758 track->usrbuf.mem = (void *)vstart; 3759 track->usrbuf.capacity = newbufsize; 3760 memset(track->usrbuf.mem, 0, newvsize); 3761 return 0; 3762 3763 /* failure */ 3764 abort: 3765 track->uobj = NULL; /* paranoia */ 3766 track->usrbuf.mem = NULL; 3767 track->usrbuf.capacity = 0; 3768 return error; 3769 } 3770 3771 /* 3772 * Free usrbuf (if available). 3773 */ 3774 static void 3775 audio_free_usrbuf(audio_track_t *track) 3776 { 3777 vaddr_t vstart; 3778 vsize_t vsize; 3779 3780 vstart = (vaddr_t)track->usrbuf.mem; 3781 vsize = roundup2(MAX(track->usrbuf.capacity, PAGE_SIZE), PAGE_SIZE); 3782 if (track->usrbuf.mem != NULL) { 3783 /* 3784 * Unmap the kernel mapping. uvm_unmap releases the 3785 * reference to the uvm object, and this should be the 3786 * last virtual mapping of the uvm object, so no need 3787 * to explicitly release (`detach') the object. 3788 */ 3789 uvm_unmap(kernel_map, vstart, vstart + vsize); 3790 3791 track->uobj = NULL; 3792 track->usrbuf.mem = NULL; 3793 track->usrbuf.capacity = 0; 3794 } 3795 } 3796 3797 /* 3798 * This filter changes the volume for each channel. 3799 * arg->context points track->ch_volume[]. 3800 */ 3801 static void 3802 audio_track_chvol(audio_filter_arg_t *arg) 3803 { 3804 int16_t *ch_volume; 3805 const aint_t *s; 3806 aint_t *d; 3807 u_int i; 3808 u_int ch; 3809 u_int channels; 3810 3811 DIAGNOSTIC_filter_arg(arg); 3812 KASSERTMSG(arg->srcfmt->channels == arg->dstfmt->channels, 3813 "arg->srcfmt->channels=%d, arg->dstfmt->channels=%d", 3814 arg->srcfmt->channels, arg->dstfmt->channels); 3815 KASSERT(arg->context != NULL); 3816 KASSERTMSG(arg->srcfmt->channels <= AUDIO_MAX_CHANNELS, 3817 "arg->srcfmt->channels=%d", arg->srcfmt->channels); 3818 3819 s = arg->src; 3820 d = arg->dst; 3821 ch_volume = arg->context; 3822 3823 channels = arg->srcfmt->channels; 3824 for (i = 0; i < arg->count; i++) { 3825 for (ch = 0; ch < channels; ch++) { 3826 aint2_t val; 3827 val = *s++; 3828 val = AUDIO_SCALEDOWN(val * ch_volume[ch], 8); 3829 *d++ = (aint_t)val; 3830 } 3831 } 3832 } 3833 3834 /* 3835 * This filter performs conversion from stereo (or more channels) to mono. 3836 */ 3837 static void 3838 audio_track_chmix_mixLR(audio_filter_arg_t *arg) 3839 { 3840 const aint_t *s; 3841 aint_t *d; 3842 u_int i; 3843 3844 DIAGNOSTIC_filter_arg(arg); 3845 3846 s = arg->src; 3847 d = arg->dst; 3848 3849 for (i = 0; i < arg->count; i++) { 3850 *d++ = AUDIO_SCALEDOWN(s[0], 1) + AUDIO_SCALEDOWN(s[1], 1); 3851 s += arg->srcfmt->channels; 3852 } 3853 } 3854 3855 /* 3856 * This filter performs conversion from mono to stereo (or more channels). 3857 */ 3858 static void 3859 audio_track_chmix_dupLR(audio_filter_arg_t *arg) 3860 { 3861 const aint_t *s; 3862 aint_t *d; 3863 u_int i; 3864 u_int ch; 3865 u_int dstchannels; 3866 3867 DIAGNOSTIC_filter_arg(arg); 3868 3869 s = arg->src; 3870 d = arg->dst; 3871 dstchannels = arg->dstfmt->channels; 3872 3873 for (i = 0; i < arg->count; i++) { 3874 d[0] = s[0]; 3875 d[1] = s[0]; 3876 s++; 3877 d += dstchannels; 3878 } 3879 if (dstchannels > 2) { 3880 d = arg->dst; 3881 for (i = 0; i < arg->count; i++) { 3882 for (ch = 2; ch < dstchannels; ch++) { 3883 d[ch] = 0; 3884 } 3885 d += dstchannels; 3886 } 3887 } 3888 } 3889 3890 /* 3891 * This filter shrinks M channels into N channels. 3892 * Extra channels are discarded. 3893 */ 3894 static void 3895 audio_track_chmix_shrink(audio_filter_arg_t *arg) 3896 { 3897 const aint_t *s; 3898 aint_t *d; 3899 u_int i; 3900 u_int ch; 3901 3902 DIAGNOSTIC_filter_arg(arg); 3903 3904 s = arg->src; 3905 d = arg->dst; 3906 3907 for (i = 0; i < arg->count; i++) { 3908 for (ch = 0; ch < arg->dstfmt->channels; ch++) { 3909 *d++ = s[ch]; 3910 } 3911 s += arg->srcfmt->channels; 3912 } 3913 } 3914 3915 /* 3916 * This filter expands M channels into N channels. 3917 * Silence is inserted for missing channels. 3918 */ 3919 static void 3920 audio_track_chmix_expand(audio_filter_arg_t *arg) 3921 { 3922 const aint_t *s; 3923 aint_t *d; 3924 u_int i; 3925 u_int ch; 3926 u_int srcchannels; 3927 u_int dstchannels; 3928 3929 DIAGNOSTIC_filter_arg(arg); 3930 3931 s = arg->src; 3932 d = arg->dst; 3933 3934 srcchannels = arg->srcfmt->channels; 3935 dstchannels = arg->dstfmt->channels; 3936 for (i = 0; i < arg->count; i++) { 3937 for (ch = 0; ch < srcchannels; ch++) { 3938 *d++ = *s++; 3939 } 3940 for (; ch < dstchannels; ch++) { 3941 *d++ = 0; 3942 } 3943 } 3944 } 3945 3946 /* 3947 * This filter performs frequency conversion (up sampling). 3948 * It uses linear interpolation. 3949 */ 3950 static void 3951 audio_track_freq_up(audio_filter_arg_t *arg) 3952 { 3953 audio_track_t *track; 3954 audio_ring_t *src; 3955 audio_ring_t *dst; 3956 const aint_t *s; 3957 aint_t *d; 3958 aint_t prev[AUDIO_MAX_CHANNELS]; 3959 aint_t curr[AUDIO_MAX_CHANNELS]; 3960 aint_t grad[AUDIO_MAX_CHANNELS]; 3961 u_int i; 3962 u_int t; 3963 u_int step; 3964 u_int channels; 3965 u_int ch; 3966 int srcused; 3967 3968 track = arg->context; 3969 KASSERT(track); 3970 src = &track->freq.srcbuf; 3971 dst = track->freq.dst; 3972 DIAGNOSTIC_ring(dst); 3973 DIAGNOSTIC_ring(src); 3974 KASSERT(src->used > 0); 3975 KASSERTMSG(src->fmt.channels == dst->fmt.channels, 3976 "src->fmt.channels=%d dst->fmt.channels=%d", 3977 src->fmt.channels, dst->fmt.channels); 3978 KASSERTMSG(src->head % track->mixer->frames_per_block == 0, 3979 "src->head=%d track->mixer->frames_per_block=%d", 3980 src->head, track->mixer->frames_per_block); 3981 3982 s = arg->src; 3983 d = arg->dst; 3984 3985 /* 3986 * In order to facilitate interpolation for each block, slide (delay) 3987 * input by one sample. As a result, strictly speaking, the output 3988 * phase is delayed by 1/dstfreq. However, I believe there is no 3989 * observable impact. 3990 * 3991 * Example) 3992 * srcfreq:dstfreq = 1:3 3993 * 3994 * A - - 3995 * | 3996 * | 3997 * | B - - 3998 * +-----+-----> input timeframe 3999 * 0 1 4000 * 4001 * 0 1 4002 * +-----+-----> input timeframe 4003 * | A 4004 * | x x 4005 * | x x 4006 * x (B) 4007 * +-+-+-+-+-+-> output timeframe 4008 * 0 1 2 3 4 5 4009 */ 4010 4011 /* Last samples in previous block */ 4012 channels = src->fmt.channels; 4013 for (ch = 0; ch < channels; ch++) { 4014 prev[ch] = track->freq_prev[ch]; 4015 curr[ch] = track->freq_curr[ch]; 4016 grad[ch] = curr[ch] - prev[ch]; 4017 } 4018 4019 step = track->freq_step; 4020 t = track->freq_current; 4021 //#define FREQ_DEBUG 4022 #if defined(FREQ_DEBUG) 4023 #define PRINTF(fmt...) printf(fmt) 4024 #else 4025 #define PRINTF(fmt...) do { } while (0) 4026 #endif 4027 srcused = src->used; 4028 PRINTF("upstart step=%d leap=%d", step, track->freq_leap); 4029 PRINTF(" srcused=%d arg->count=%u", src->used, arg->count); 4030 PRINTF(" prev=%d curr=%d grad=%d", prev[0], curr[0], grad[0]); 4031 PRINTF(" t=%d\n", t); 4032 4033 for (i = 0; i < arg->count; i++) { 4034 PRINTF("i=%d t=%5d", i, t); 4035 if (t >= 65536) { 4036 for (ch = 0; ch < channels; ch++) { 4037 prev[ch] = curr[ch]; 4038 curr[ch] = *s++; 4039 grad[ch] = curr[ch] - prev[ch]; 4040 } 4041 PRINTF(" prev=%d s[%d]=%d", 4042 prev[0], src->used - srcused, curr[0]); 4043 4044 /* Update */ 4045 t -= 65536; 4046 srcused--; 4047 if (srcused < 0) { 4048 PRINTF(" break\n"); 4049 break; 4050 } 4051 } 4052 4053 for (ch = 0; ch < channels; ch++) { 4054 *d++ = prev[ch] + (aint2_t)grad[ch] * t / 65536; 4055 #if defined(FREQ_DEBUG) 4056 if (ch == 0) 4057 printf(" t=%5d *d=%d", t, d[-1]); 4058 #endif 4059 } 4060 t += step; 4061 4062 PRINTF("\n"); 4063 } 4064 PRINTF("end prev=%d curr=%d\n", prev[0], curr[0]); 4065 4066 auring_take(src, src->used); 4067 auring_push(dst, i); 4068 4069 /* Adjust */ 4070 t += track->freq_leap; 4071 4072 track->freq_current = t; 4073 for (ch = 0; ch < channels; ch++) { 4074 track->freq_prev[ch] = prev[ch]; 4075 track->freq_curr[ch] = curr[ch]; 4076 } 4077 } 4078 4079 /* 4080 * This filter performs frequency conversion (down sampling). 4081 * It uses simple thinning. 4082 */ 4083 static void 4084 audio_track_freq_down(audio_filter_arg_t *arg) 4085 { 4086 audio_track_t *track; 4087 audio_ring_t *src; 4088 audio_ring_t *dst; 4089 const aint_t *s0; 4090 aint_t *d; 4091 u_int i; 4092 u_int t; 4093 u_int step; 4094 u_int ch; 4095 u_int channels; 4096 4097 track = arg->context; 4098 KASSERT(track); 4099 src = &track->freq.srcbuf; 4100 dst = track->freq.dst; 4101 4102 DIAGNOSTIC_ring(dst); 4103 DIAGNOSTIC_ring(src); 4104 KASSERT(src->used > 0); 4105 KASSERTMSG(src->fmt.channels == dst->fmt.channels, 4106 "src->fmt.channels=%d dst->fmt.channels=%d", 4107 src->fmt.channels, dst->fmt.channels); 4108 KASSERTMSG(src->head % track->mixer->frames_per_block == 0, 4109 "src->head=%d track->mixer->frames_per_block=%d", 4110 src->head, track->mixer->frames_per_block); 4111 4112 s0 = arg->src; 4113 d = arg->dst; 4114 t = track->freq_current; 4115 step = track->freq_step; 4116 channels = dst->fmt.channels; 4117 PRINTF("downstart step=%d leap=%d", step, track->freq_leap); 4118 PRINTF(" srcused=%d arg->count=%u", src->used, arg->count); 4119 PRINTF(" t=%d\n", t); 4120 4121 for (i = 0; i < arg->count && t / 65536 < src->used; i++) { 4122 const aint_t *s; 4123 PRINTF("i=%4d t=%10d", i, t); 4124 s = s0 + (t / 65536) * channels; 4125 PRINTF(" s=%5ld", (s - s0) / channels); 4126 for (ch = 0; ch < channels; ch++) { 4127 if (ch == 0) PRINTF(" *s=%d", s[ch]); 4128 *d++ = s[ch]; 4129 } 4130 PRINTF("\n"); 4131 t += step; 4132 } 4133 t += track->freq_leap; 4134 PRINTF("end t=%d\n", t); 4135 auring_take(src, src->used); 4136 auring_push(dst, i); 4137 track->freq_current = t % 65536; 4138 } 4139 4140 /* 4141 * Creates track and returns it. 4142 * Must be called without sc_lock held. 4143 */ 4144 audio_track_t * 4145 audio_track_create(struct audio_softc *sc, audio_trackmixer_t *mixer) 4146 { 4147 audio_track_t *track; 4148 static int newid = 0; 4149 4150 track = kmem_zalloc(sizeof(*track), KM_SLEEP); 4151 4152 track->id = newid++; 4153 track->mixer = mixer; 4154 track->mode = mixer->mode; 4155 4156 /* Do TRACE after id is assigned. */ 4157 TRACET(3, track, "for %s", 4158 mixer->mode == AUMODE_PLAY ? "playback" : "recording"); 4159 4160 #if defined(AUDIO_SUPPORT_TRACK_VOLUME) 4161 track->volume = 256; 4162 #endif 4163 for (int i = 0; i < AUDIO_MAX_CHANNELS; i++) { 4164 track->ch_volume[i] = 256; 4165 } 4166 4167 return track; 4168 } 4169 4170 /* 4171 * Release all resources of the track and track itself. 4172 * track must not be NULL. Don't specify the track within the file 4173 * structure linked from sc->sc_files. 4174 */ 4175 static void 4176 audio_track_destroy(audio_track_t *track) 4177 { 4178 4179 KASSERT(track); 4180 4181 audio_free_usrbuf(track); 4182 audio_free(track->codec.srcbuf.mem); 4183 audio_free(track->chvol.srcbuf.mem); 4184 audio_free(track->chmix.srcbuf.mem); 4185 audio_free(track->freq.srcbuf.mem); 4186 audio_free(track->outbuf.mem); 4187 4188 kmem_free(track, sizeof(*track)); 4189 } 4190 4191 /* 4192 * It returns encoding conversion filter according to src and dst format. 4193 * If it is not a convertible pair, it returns NULL. Either src or dst 4194 * must be internal format. 4195 */ 4196 static audio_filter_t 4197 audio_track_get_codec(audio_track_t *track, const audio_format2_t *src, 4198 const audio_format2_t *dst) 4199 { 4200 4201 if (audio_format2_is_internal(src)) { 4202 if (dst->encoding == AUDIO_ENCODING_ULAW) { 4203 return audio_internal_to_mulaw; 4204 } else if (dst->encoding == AUDIO_ENCODING_ALAW) { 4205 return audio_internal_to_alaw; 4206 } else if (audio_format2_is_linear(dst)) { 4207 switch (dst->stride) { 4208 case 8: 4209 return audio_internal_to_linear8; 4210 case 16: 4211 return audio_internal_to_linear16; 4212 #if defined(AUDIO_SUPPORT_LINEAR24) 4213 case 24: 4214 return audio_internal_to_linear24; 4215 #endif 4216 case 32: 4217 return audio_internal_to_linear32; 4218 default: 4219 TRACET(1, track, "unsupported %s stride %d", 4220 "dst", dst->stride); 4221 goto abort; 4222 } 4223 } 4224 } else if (audio_format2_is_internal(dst)) { 4225 if (src->encoding == AUDIO_ENCODING_ULAW) { 4226 return audio_mulaw_to_internal; 4227 } else if (src->encoding == AUDIO_ENCODING_ALAW) { 4228 return audio_alaw_to_internal; 4229 } else if (audio_format2_is_linear(src)) { 4230 switch (src->stride) { 4231 case 8: 4232 return audio_linear8_to_internal; 4233 case 16: 4234 return audio_linear16_to_internal; 4235 #if defined(AUDIO_SUPPORT_LINEAR24) 4236 case 24: 4237 return audio_linear24_to_internal; 4238 #endif 4239 case 32: 4240 return audio_linear32_to_internal; 4241 default: 4242 TRACET(1, track, "unsupported %s stride %d", 4243 "src", src->stride); 4244 goto abort; 4245 } 4246 } 4247 } 4248 4249 TRACET(1, track, "unsupported encoding"); 4250 abort: 4251 #if defined(AUDIO_DEBUG) 4252 if (audiodebug >= 2) { 4253 char buf[100]; 4254 audio_format2_tostr(buf, sizeof(buf), src); 4255 TRACET(2, track, "src %s", buf); 4256 audio_format2_tostr(buf, sizeof(buf), dst); 4257 TRACET(2, track, "dst %s", buf); 4258 } 4259 #endif 4260 return NULL; 4261 } 4262 4263 /* 4264 * Initialize the codec stage of this track as necessary. 4265 * If successful, it initializes the codec stage as necessary, stores updated 4266 * last_dst in *last_dstp in any case, and returns 0. 4267 * Otherwise, it returns errno without modifying *last_dstp. 4268 */ 4269 static int 4270 audio_track_init_codec(audio_track_t *track, audio_ring_t **last_dstp) 4271 { 4272 audio_ring_t *last_dst; 4273 audio_ring_t *srcbuf; 4274 audio_format2_t *srcfmt; 4275 audio_format2_t *dstfmt; 4276 audio_filter_arg_t *arg; 4277 u_int len; 4278 int error; 4279 4280 KASSERT(track); 4281 4282 last_dst = *last_dstp; 4283 dstfmt = &last_dst->fmt; 4284 srcfmt = &track->inputfmt; 4285 srcbuf = &track->codec.srcbuf; 4286 error = 0; 4287 4288 if (srcfmt->encoding != dstfmt->encoding 4289 || srcfmt->precision != dstfmt->precision 4290 || srcfmt->stride != dstfmt->stride) { 4291 track->codec.dst = last_dst; 4292 4293 srcbuf->fmt = *dstfmt; 4294 srcbuf->fmt.encoding = srcfmt->encoding; 4295 srcbuf->fmt.precision = srcfmt->precision; 4296 srcbuf->fmt.stride = srcfmt->stride; 4297 4298 track->codec.filter = audio_track_get_codec(track, 4299 &srcbuf->fmt, dstfmt); 4300 if (track->codec.filter == NULL) { 4301 error = EINVAL; 4302 goto abort; 4303 } 4304 4305 srcbuf->head = 0; 4306 srcbuf->used = 0; 4307 srcbuf->capacity = frame_per_block(track->mixer, &srcbuf->fmt); 4308 len = auring_bytelen(srcbuf); 4309 srcbuf->mem = audio_realloc(srcbuf->mem, len); 4310 4311 arg = &track->codec.arg; 4312 arg->srcfmt = &srcbuf->fmt; 4313 arg->dstfmt = dstfmt; 4314 arg->context = NULL; 4315 4316 *last_dstp = srcbuf; 4317 return 0; 4318 } 4319 4320 abort: 4321 track->codec.filter = NULL; 4322 audio_free(srcbuf->mem); 4323 return error; 4324 } 4325 4326 /* 4327 * Initialize the chvol stage of this track as necessary. 4328 * If successful, it initializes the chvol stage as necessary, stores updated 4329 * last_dst in *last_dstp in any case, and returns 0. 4330 * Otherwise, it returns errno without modifying *last_dstp. 4331 */ 4332 static int 4333 audio_track_init_chvol(audio_track_t *track, audio_ring_t **last_dstp) 4334 { 4335 audio_ring_t *last_dst; 4336 audio_ring_t *srcbuf; 4337 audio_format2_t *srcfmt; 4338 audio_format2_t *dstfmt; 4339 audio_filter_arg_t *arg; 4340 u_int len; 4341 int error; 4342 4343 KASSERT(track); 4344 4345 last_dst = *last_dstp; 4346 dstfmt = &last_dst->fmt; 4347 srcfmt = &track->inputfmt; 4348 srcbuf = &track->chvol.srcbuf; 4349 error = 0; 4350 4351 /* Check whether channel volume conversion is necessary. */ 4352 bool use_chvol = false; 4353 for (int ch = 0; ch < srcfmt->channels; ch++) { 4354 if (track->ch_volume[ch] != 256) { 4355 use_chvol = true; 4356 break; 4357 } 4358 } 4359 4360 if (use_chvol == true) { 4361 track->chvol.dst = last_dst; 4362 track->chvol.filter = audio_track_chvol; 4363 4364 srcbuf->fmt = *dstfmt; 4365 /* no format conversion occurs */ 4366 4367 srcbuf->head = 0; 4368 srcbuf->used = 0; 4369 srcbuf->capacity = frame_per_block(track->mixer, &srcbuf->fmt); 4370 len = auring_bytelen(srcbuf); 4371 srcbuf->mem = audio_realloc(srcbuf->mem, len); 4372 4373 arg = &track->chvol.arg; 4374 arg->srcfmt = &srcbuf->fmt; 4375 arg->dstfmt = dstfmt; 4376 arg->context = track->ch_volume; 4377 4378 *last_dstp = srcbuf; 4379 return 0; 4380 } 4381 4382 track->chvol.filter = NULL; 4383 audio_free(srcbuf->mem); 4384 return error; 4385 } 4386 4387 /* 4388 * Initialize the chmix stage of this track as necessary. 4389 * If successful, it initializes the chmix stage as necessary, stores updated 4390 * last_dst in *last_dstp in any case, and returns 0. 4391 * Otherwise, it returns errno without modifying *last_dstp. 4392 */ 4393 static int 4394 audio_track_init_chmix(audio_track_t *track, audio_ring_t **last_dstp) 4395 { 4396 audio_ring_t *last_dst; 4397 audio_ring_t *srcbuf; 4398 audio_format2_t *srcfmt; 4399 audio_format2_t *dstfmt; 4400 audio_filter_arg_t *arg; 4401 u_int srcch; 4402 u_int dstch; 4403 u_int len; 4404 int error; 4405 4406 KASSERT(track); 4407 4408 last_dst = *last_dstp; 4409 dstfmt = &last_dst->fmt; 4410 srcfmt = &track->inputfmt; 4411 srcbuf = &track->chmix.srcbuf; 4412 error = 0; 4413 4414 srcch = srcfmt->channels; 4415 dstch = dstfmt->channels; 4416 if (srcch != dstch) { 4417 track->chmix.dst = last_dst; 4418 4419 if (srcch >= 2 && dstch == 1) { 4420 track->chmix.filter = audio_track_chmix_mixLR; 4421 } else if (srcch == 1 && dstch >= 2) { 4422 track->chmix.filter = audio_track_chmix_dupLR; 4423 } else if (srcch > dstch) { 4424 track->chmix.filter = audio_track_chmix_shrink; 4425 } else { 4426 track->chmix.filter = audio_track_chmix_expand; 4427 } 4428 4429 srcbuf->fmt = *dstfmt; 4430 srcbuf->fmt.channels = srcch; 4431 4432 srcbuf->head = 0; 4433 srcbuf->used = 0; 4434 /* XXX The buffer size should be able to calculate. */ 4435 srcbuf->capacity = frame_per_block(track->mixer, &srcbuf->fmt); 4436 len = auring_bytelen(srcbuf); 4437 srcbuf->mem = audio_realloc(srcbuf->mem, len); 4438 4439 arg = &track->chmix.arg; 4440 arg->srcfmt = &srcbuf->fmt; 4441 arg->dstfmt = dstfmt; 4442 arg->context = NULL; 4443 4444 *last_dstp = srcbuf; 4445 return 0; 4446 } 4447 4448 track->chmix.filter = NULL; 4449 audio_free(srcbuf->mem); 4450 return error; 4451 } 4452 4453 /* 4454 * Initialize the freq stage of this track as necessary. 4455 * If successful, it initializes the freq stage as necessary, stores updated 4456 * last_dst in *last_dstp in any case, and returns 0. 4457 * Otherwise, it returns errno without modifying *last_dstp. 4458 */ 4459 static int 4460 audio_track_init_freq(audio_track_t *track, audio_ring_t **last_dstp) 4461 { 4462 audio_ring_t *last_dst; 4463 audio_ring_t *srcbuf; 4464 audio_format2_t *srcfmt; 4465 audio_format2_t *dstfmt; 4466 audio_filter_arg_t *arg; 4467 uint32_t srcfreq; 4468 uint32_t dstfreq; 4469 u_int dst_capacity; 4470 u_int mod; 4471 u_int len; 4472 int error; 4473 4474 KASSERT(track); 4475 4476 last_dst = *last_dstp; 4477 dstfmt = &last_dst->fmt; 4478 srcfmt = &track->inputfmt; 4479 srcbuf = &track->freq.srcbuf; 4480 error = 0; 4481 4482 srcfreq = srcfmt->sample_rate; 4483 dstfreq = dstfmt->sample_rate; 4484 if (srcfreq != dstfreq) { 4485 track->freq.dst = last_dst; 4486 4487 memset(track->freq_prev, 0, sizeof(track->freq_prev)); 4488 memset(track->freq_curr, 0, sizeof(track->freq_curr)); 4489 4490 /* freq_step is the ratio of src/dst when let dst 65536. */ 4491 track->freq_step = (uint64_t)srcfreq * 65536 / dstfreq; 4492 4493 dst_capacity = frame_per_block(track->mixer, dstfmt); 4494 mod = (uint64_t)srcfreq * 65536 % dstfreq; 4495 track->freq_leap = (mod * dst_capacity + dstfreq / 2) / dstfreq; 4496 4497 if (track->freq_step < 65536) { 4498 track->freq.filter = audio_track_freq_up; 4499 /* In order to carry at the first time. */ 4500 track->freq_current = 65536; 4501 } else { 4502 track->freq.filter = audio_track_freq_down; 4503 track->freq_current = 0; 4504 } 4505 4506 srcbuf->fmt = *dstfmt; 4507 srcbuf->fmt.sample_rate = srcfreq; 4508 4509 srcbuf->head = 0; 4510 srcbuf->used = 0; 4511 srcbuf->capacity = frame_per_block(track->mixer, &srcbuf->fmt); 4512 len = auring_bytelen(srcbuf); 4513 srcbuf->mem = audio_realloc(srcbuf->mem, len); 4514 4515 arg = &track->freq.arg; 4516 arg->srcfmt = &srcbuf->fmt; 4517 arg->dstfmt = dstfmt;/*&last_dst->fmt;*/ 4518 arg->context = track; 4519 4520 *last_dstp = srcbuf; 4521 return 0; 4522 } 4523 4524 track->freq.filter = NULL; 4525 audio_free(srcbuf->mem); 4526 return error; 4527 } 4528 4529 /* 4530 * There are two unit of buffers; A block buffer and a byte buffer. Both use 4531 * audio_ring_t. Internally, audio data is always handled in block unit. 4532 * Converting format, sythesizing tracks, transferring from/to the hardware, 4533 * and etc. Only one exception is usrbuf. To transfer with userland, usrbuf 4534 * is buffered in byte unit. 4535 * For playing back, write(2) writes arbitrary length of data to usrbuf. 4536 * When one block is filled, it is sent to the next stage (converting and/or 4537 * synthesizing). 4538 * For recording, the rmixer writes one block length of data to input buffer 4539 * (the bottom stage buffer) each time. read(2) (converts one block if usrbuf 4540 * is empty and then) reads arbitrary length of data from usrbuf. 4541 * 4542 * The following charts show the data flow and buffer types for playback and 4543 * recording track. In this example, both have two conversion stages, codec 4544 * and freq. Every [**] represents a buffer described below. 4545 * 4546 * On playback track: 4547 * 4548 * write(2) 4549 * | 4550 * | uiomove 4551 * v 4552 * usrbuf [BB|BB ... BB|BB] .. Byte ring buffer 4553 * | 4554 * | memcpy one block 4555 * v 4556 * codec.srcbuf [FF] .. 1 block (ring) buffer 4557 * .dst ----+ 4558 * | 4559 * | convert 4560 * v 4561 * freq.srcbuf [FF] .. 1 block (ring) buffer 4562 * .dst ----+ 4563 * | 4564 * | convert 4565 * v 4566 * outbuf [FF|FF|FF|FF] .. NBLKOUT blocks ring buffer 4567 * | 4568 * v 4569 * pmixer 4570 * 4571 * There are three different types of buffers: 4572 * 4573 * [BB|BB ... BB|BB] usrbuf. Is the buffer closest to userland. Mandatory. 4574 * This is a byte buffer and its length is basically less 4575 * than or equal to 64KB or at least AUMINNOBLK blocks. 4576 * 4577 * [FF] Interim conversion stage's srcbuf if necessary. 4578 * This is one block (ring) buffer counted in frames. 4579 * 4580 * [FF|FF|FF|FF] outbuf. Is the buffer closest to pmixer. Mandatory. 4581 * This is NBLKOUT blocks ring buffer counted in frames. 4582 * 4583 * 4584 * On recording track: 4585 * 4586 * read(2) 4587 * ^ 4588 * | uiomove 4589 * | 4590 * usrbuf [BB] .. Byte (ring) buffer 4591 * ^ 4592 * | memcpy one block 4593 * | 4594 * outbuf [FF] .. 1 block (ring) buffer 4595 * ^ 4596 * | convert 4597 * | 4598 * codec.dst ----+ 4599 * .srcbuf [FF] .. 1 block (ring) buffer 4600 * ^ 4601 * | convert 4602 * | 4603 * freq.dst ----+ 4604 * .srcbuf [FF|FF ... FF|FF] .. NBLKIN blocks ring buffer 4605 * ^ 4606 * | 4607 * rmixer 4608 * 4609 * There are also three different types of buffers. 4610 * 4611 * [BB] usrbuf. Is the buffer closest to userland. Mandatory. 4612 * This is a byte buffer and its length is one block. 4613 * This buffer holds only "fragment". 4614 * 4615 * [FF] Interim conversion stage's srcbuf (or outbuf). 4616 * This is one block (ring) buffer counted in frames. 4617 * 4618 * [FF|FF ... FF|FF] The bottom conversion stage's srcbuf (or outbuf). 4619 * This is the buffer closest to rmixer, and mandatory. 4620 * This is NBLKIN blocks ring buffer counted in frames. 4621 * Also pointed by *input. 4622 */ 4623 4624 /* 4625 * Set the userland format of this track. 4626 * usrfmt argument should have been previously verified by 4627 * audio_track_setinfo_check(). 4628 * This function may release and reallocate all internal conversion buffers. 4629 * It returns 0 if successful. Otherwise it returns errno with clearing all 4630 * internal buffers. 4631 * It must be called without sc_intr_lock since uvm_* routines require non 4632 * intr_lock state. 4633 * It must be called with track lock held since it may release and reallocate 4634 * outbuf. 4635 */ 4636 static int 4637 audio_track_set_format(audio_track_t *track, audio_format2_t *usrfmt) 4638 { 4639 struct audio_softc *sc; 4640 audio_ring_t *last_dst; 4641 int is_playback; 4642 u_int newbufsize; 4643 u_int oldblksize; 4644 u_int len; 4645 int error; 4646 4647 KASSERT(track); 4648 sc = track->mixer->sc; 4649 4650 is_playback = audio_track_is_playback(track); 4651 4652 /* usrbuf is the closest buffer to the userland. */ 4653 track->usrbuf.fmt = *usrfmt; 4654 4655 /* 4656 * Usrbuf. 4657 * On the playback track, its capacity is less than or equal to 64KB 4658 * (for historical reason) and must be a multiple of a block 4659 * (constraint in this implementation). But at least AUMINNOBLK 4660 * blocks. 4661 * On the recording track, its capacity is one block. 4662 */ 4663 /* 4664 * For references, one block size (in 40msec) is: 4665 * 320 bytes = 204 blocks/64KB for mulaw/8kHz/1ch 4666 * 7680 bytes = 8 blocks/64KB for s16/48kHz/2ch 4667 * 30720 bytes = 90 KB/3blocks for s16/48kHz/8ch 4668 * 61440 bytes = 180 KB/3blocks for s16/96kHz/8ch 4669 * 245760 bytes = 720 KB/3blocks for s32/192kHz/8ch 4670 * 4671 * For example, 4672 * 1) If usrbuf_blksize = 7056 (s16/44.1k/2ch) and PAGE_SIZE = 8192, 4673 * newbufsize = rounddown(65536 / 7056) = 63504 4674 * newvsize = roundup2(63504, PAGE_SIZE) = 65536 4675 * Therefore it maps 8 * 8K pages and usrbuf->capacity = 63504. 4676 * 4677 * 2) If usrbuf_blksize = 7680 (s16/48k/2ch) and PAGE_SIZE = 4096, 4678 * newbufsize = rounddown(65536 / 7680) = 61440 4679 * newvsize = roundup2(61440, PAGE_SIZE) = 61440 (= 15 pages) 4680 * Therefore it maps 15 * 4K pages and usrbuf->capacity = 61440. 4681 */ 4682 oldblksize = track->usrbuf_blksize; 4683 track->usrbuf_blksize = frametobyte(&track->usrbuf.fmt, 4684 frame_per_block(track->mixer, &track->usrbuf.fmt)); 4685 track->usrbuf.head = 0; 4686 track->usrbuf.used = 0; 4687 if (is_playback) { 4688 if (track->usrbuf_blksize * AUMINNOBLK > 65536) 4689 newbufsize = track->usrbuf_blksize * AUMINNOBLK; 4690 else 4691 newbufsize = rounddown(65536, track->usrbuf_blksize); 4692 } else { 4693 newbufsize = track->usrbuf_blksize; 4694 } 4695 if (track->usrbuf_blksize != oldblksize) { 4696 error = audio_realloc_usrbuf(track, newbufsize); 4697 if (error) { 4698 device_printf(sc->sc_dev, "malloc usrbuf(%d) failed\n", 4699 newbufsize); 4700 goto error; 4701 } 4702 } 4703 4704 /* Recalc water mark. */ 4705 if (is_playback) { 4706 /* Set high at 100%, low at 75%. */ 4707 track->usrbuf_usedhigh = track->usrbuf.capacity; 4708 track->usrbuf_usedlow = track->usrbuf.capacity * 3 / 4; 4709 } else { 4710 /* Set high at 100%, low at 0%. (But not used) */ 4711 track->usrbuf_usedhigh = track->usrbuf.capacity; 4712 track->usrbuf_usedlow = 0; 4713 } 4714 4715 /* Stage buffer */ 4716 last_dst = &track->outbuf; 4717 if (is_playback) { 4718 /* On playback, initialize from the mixer side in order. */ 4719 track->inputfmt = *usrfmt; 4720 track->outbuf.fmt = track->mixer->track_fmt; 4721 4722 if ((error = audio_track_init_freq(track, &last_dst)) != 0) 4723 goto error; 4724 if ((error = audio_track_init_chmix(track, &last_dst)) != 0) 4725 goto error; 4726 if ((error = audio_track_init_chvol(track, &last_dst)) != 0) 4727 goto error; 4728 if ((error = audio_track_init_codec(track, &last_dst)) != 0) 4729 goto error; 4730 } else { 4731 /* On recording, initialize from userland side in order. */ 4732 track->inputfmt = track->mixer->track_fmt; 4733 track->outbuf.fmt = *usrfmt; 4734 4735 if ((error = audio_track_init_codec(track, &last_dst)) != 0) 4736 goto error; 4737 if ((error = audio_track_init_chvol(track, &last_dst)) != 0) 4738 goto error; 4739 if ((error = audio_track_init_chmix(track, &last_dst)) != 0) 4740 goto error; 4741 if ((error = audio_track_init_freq(track, &last_dst)) != 0) 4742 goto error; 4743 } 4744 #if 0 4745 /* debug */ 4746 if (track->freq.filter) { 4747 audio_print_format2("freq src", &track->freq.srcbuf.fmt); 4748 audio_print_format2("freq dst", &track->freq.dst->fmt); 4749 } 4750 if (track->chmix.filter) { 4751 audio_print_format2("chmix src", &track->chmix.srcbuf.fmt); 4752 audio_print_format2("chmix dst", &track->chmix.dst->fmt); 4753 } 4754 if (track->chvol.filter) { 4755 audio_print_format2("chvol src", &track->chvol.srcbuf.fmt); 4756 audio_print_format2("chvol dst", &track->chvol.dst->fmt); 4757 } 4758 if (track->codec.filter) { 4759 audio_print_format2("codec src", &track->codec.srcbuf.fmt); 4760 audio_print_format2("codec dst", &track->codec.dst->fmt); 4761 } 4762 #endif 4763 4764 /* Stage input buffer */ 4765 track->input = last_dst; 4766 4767 /* 4768 * Output buffer. 4769 * On the playback track, its capacity is NBLKOUT blocks. 4770 * On the recording track, its capacity is 1 block. 4771 */ 4772 track->outbuf.head = 0; 4773 track->outbuf.used = 0; 4774 track->outbuf.capacity = frame_per_block(track->mixer, 4775 &track->outbuf.fmt); 4776 if (is_playback) 4777 track->outbuf.capacity *= NBLKOUT; 4778 len = auring_bytelen(&track->outbuf); 4779 track->outbuf.mem = audio_realloc(track->outbuf.mem, len); 4780 4781 /* 4782 * On the recording track, expand the input stage buffer, which is 4783 * the closest buffer to rmixer, to NBLKOUT blocks. 4784 * Note that input buffer may point to outbuf. 4785 */ 4786 if (!is_playback) { 4787 int input_fpb; 4788 4789 input_fpb = frame_per_block(track->mixer, &track->input->fmt); 4790 track->input->capacity = input_fpb * NBLKIN; 4791 len = auring_bytelen(track->input); 4792 track->input->mem = audio_realloc(track->input->mem, len); 4793 } 4794 4795 #if defined(AUDIO_DEBUG) 4796 if (audiodebug >= 3) { 4797 struct audio_track_debugbuf m; 4798 4799 memset(&m, 0, sizeof(m)); 4800 snprintf(m.outbuf, sizeof(m.outbuf), " out=%d", 4801 track->outbuf.capacity * frametobyte(&track->outbuf.fmt,1)); 4802 if (track->freq.filter) 4803 snprintf(m.freq, sizeof(m.freq), " freq=%d", 4804 track->freq.srcbuf.capacity * 4805 frametobyte(&track->freq.srcbuf.fmt, 1)); 4806 if (track->chmix.filter) 4807 snprintf(m.chmix, sizeof(m.chmix), " chmix=%d", 4808 track->chmix.srcbuf.capacity * 4809 frametobyte(&track->chmix.srcbuf.fmt, 1)); 4810 if (track->chvol.filter) 4811 snprintf(m.chvol, sizeof(m.chvol), " chvol=%d", 4812 track->chvol.srcbuf.capacity * 4813 frametobyte(&track->chvol.srcbuf.fmt, 1)); 4814 if (track->codec.filter) 4815 snprintf(m.codec, sizeof(m.codec), " codec=%d", 4816 track->codec.srcbuf.capacity * 4817 frametobyte(&track->codec.srcbuf.fmt, 1)); 4818 snprintf(m.usrbuf, sizeof(m.usrbuf), 4819 " usr=%d", track->usrbuf.capacity); 4820 4821 if (is_playback) { 4822 TRACET(0, track, "bufsize%s%s%s%s%s%s", 4823 m.outbuf, m.freq, m.chmix, 4824 m.chvol, m.codec, m.usrbuf); 4825 } else { 4826 TRACET(0, track, "bufsize%s%s%s%s%s%s", 4827 m.freq, m.chmix, m.chvol, 4828 m.codec, m.outbuf, m.usrbuf); 4829 } 4830 } 4831 #endif 4832 return 0; 4833 4834 error: 4835 audio_free_usrbuf(track); 4836 audio_free(track->codec.srcbuf.mem); 4837 audio_free(track->chvol.srcbuf.mem); 4838 audio_free(track->chmix.srcbuf.mem); 4839 audio_free(track->freq.srcbuf.mem); 4840 audio_free(track->outbuf.mem); 4841 return error; 4842 } 4843 4844 /* 4845 * Fill silence frames (as the internal format) up to 1 block 4846 * if the ring is not empty and less than 1 block. 4847 * It returns the number of appended frames. 4848 */ 4849 static int 4850 audio_append_silence(audio_track_t *track, audio_ring_t *ring) 4851 { 4852 int fpb; 4853 int n; 4854 4855 KASSERT(track); 4856 KASSERT(audio_format2_is_internal(&ring->fmt)); 4857 4858 /* XXX is n correct? */ 4859 /* XXX memset uses frametobyte()? */ 4860 4861 if (ring->used == 0) 4862 return 0; 4863 4864 fpb = frame_per_block(track->mixer, &ring->fmt); 4865 if (ring->used >= fpb) 4866 return 0; 4867 4868 n = (ring->capacity - ring->used) % fpb; 4869 4870 KASSERTMSG(auring_get_contig_free(ring) >= n, 4871 "auring_get_contig_free(ring)=%d n=%d", 4872 auring_get_contig_free(ring), n); 4873 4874 memset(auring_tailptr_aint(ring), 0, 4875 n * ring->fmt.channels * sizeof(aint_t)); 4876 auring_push(ring, n); 4877 return n; 4878 } 4879 4880 /* 4881 * Execute the conversion stage. 4882 * It prepares arg from this stage and executes stage->filter. 4883 * It must be called only if stage->filter is not NULL. 4884 * 4885 * For stages other than frequency conversion, the function increments 4886 * src and dst counters here. For frequency conversion stage, on the 4887 * other hand, the function does not touch src and dst counters and 4888 * filter side has to increment them. 4889 */ 4890 static void 4891 audio_apply_stage(audio_track_t *track, audio_stage_t *stage, bool isfreq) 4892 { 4893 audio_filter_arg_t *arg; 4894 int srccount; 4895 int dstcount; 4896 int count; 4897 4898 KASSERT(track); 4899 KASSERT(stage->filter); 4900 4901 srccount = auring_get_contig_used(&stage->srcbuf); 4902 dstcount = auring_get_contig_free(stage->dst); 4903 4904 if (isfreq) { 4905 KASSERTMSG(srccount > 0, "freq but srccount=%d", srccount); 4906 count = uimin(dstcount, track->mixer->frames_per_block); 4907 } else { 4908 count = uimin(srccount, dstcount); 4909 } 4910 4911 if (count > 0) { 4912 arg = &stage->arg; 4913 arg->src = auring_headptr(&stage->srcbuf); 4914 arg->dst = auring_tailptr(stage->dst); 4915 arg->count = count; 4916 4917 stage->filter(arg); 4918 4919 if (!isfreq) { 4920 auring_take(&stage->srcbuf, count); 4921 auring_push(stage->dst, count); 4922 } 4923 } 4924 } 4925 4926 /* 4927 * Produce output buffer for playback from user input buffer. 4928 * It must be called only if usrbuf is not empty and outbuf is 4929 * available at least one free block. 4930 */ 4931 static void 4932 audio_track_play(audio_track_t *track) 4933 { 4934 audio_ring_t *usrbuf; 4935 audio_ring_t *input; 4936 int count; 4937 int framesize; 4938 int bytes; 4939 4940 KASSERT(track); 4941 KASSERT(track->lock); 4942 TRACET(4, track, "start pstate=%d", track->pstate); 4943 4944 /* At this point usrbuf must not be empty. */ 4945 KASSERT(track->usrbuf.used > 0); 4946 /* Also, outbuf must be available at least one block. */ 4947 count = auring_get_contig_free(&track->outbuf); 4948 KASSERTMSG(count >= frame_per_block(track->mixer, &track->outbuf.fmt), 4949 "count=%d fpb=%d", 4950 count, frame_per_block(track->mixer, &track->outbuf.fmt)); 4951 4952 usrbuf = &track->usrbuf; 4953 input = track->input; 4954 4955 /* 4956 * framesize is always 1 byte or more since all formats supported as 4957 * usrfmt(=input) have 8bit or more stride. 4958 */ 4959 framesize = frametobyte(&input->fmt, 1); 4960 KASSERT(framesize >= 1); 4961 4962 /* The next stage of usrbuf (=input) must be available. */ 4963 KASSERT(auring_get_contig_free(input) > 0); 4964 4965 /* 4966 * Copy usrbuf up to 1block to input buffer. 4967 * count is the number of frames to copy from usrbuf. 4968 * bytes is the number of bytes to copy from usrbuf. However it is 4969 * not copied less than one frame. 4970 */ 4971 count = uimin(usrbuf->used, track->usrbuf_blksize) / framesize; 4972 bytes = count * framesize; 4973 4974 if (usrbuf->head + bytes < usrbuf->capacity) { 4975 memcpy((uint8_t *)input->mem + auring_tail(input) * framesize, 4976 (uint8_t *)usrbuf->mem + usrbuf->head, 4977 bytes); 4978 auring_push(input, count); 4979 auring_take(usrbuf, bytes); 4980 } else { 4981 int bytes1; 4982 int bytes2; 4983 4984 bytes1 = auring_get_contig_used(usrbuf); 4985 KASSERTMSG(bytes1 % framesize == 0, 4986 "bytes1=%d framesize=%d", bytes1, framesize); 4987 memcpy((uint8_t *)input->mem + auring_tail(input) * framesize, 4988 (uint8_t *)usrbuf->mem + usrbuf->head, 4989 bytes1); 4990 auring_push(input, bytes1 / framesize); 4991 auring_take(usrbuf, bytes1); 4992 4993 bytes2 = bytes - bytes1; 4994 memcpy((uint8_t *)input->mem + auring_tail(input) * framesize, 4995 (uint8_t *)usrbuf->mem + usrbuf->head, 4996 bytes2); 4997 auring_push(input, bytes2 / framesize); 4998 auring_take(usrbuf, bytes2); 4999 } 5000 5001 /* Encoding conversion */ 5002 if (track->codec.filter) 5003 audio_apply_stage(track, &track->codec, false); 5004 5005 /* Channel volume */ 5006 if (track->chvol.filter) 5007 audio_apply_stage(track, &track->chvol, false); 5008 5009 /* Channel mix */ 5010 if (track->chmix.filter) 5011 audio_apply_stage(track, &track->chmix, false); 5012 5013 /* Frequency conversion */ 5014 /* 5015 * Since the frequency conversion needs correction for each block, 5016 * it rounds up to 1 block. 5017 */ 5018 if (track->freq.filter) { 5019 int n; 5020 n = audio_append_silence(track, &track->freq.srcbuf); 5021 if (n > 0) { 5022 TRACET(4, track, 5023 "freq.srcbuf add silence %d -> %d/%d/%d", 5024 n, 5025 track->freq.srcbuf.head, 5026 track->freq.srcbuf.used, 5027 track->freq.srcbuf.capacity); 5028 } 5029 if (track->freq.srcbuf.used > 0) { 5030 audio_apply_stage(track, &track->freq, true); 5031 } 5032 } 5033 5034 if (bytes < track->usrbuf_blksize) { 5035 /* 5036 * Clear all conversion buffer pointer if the conversion was 5037 * not exactly one block. These conversion stage buffers are 5038 * certainly circular buffers because of symmetry with the 5039 * previous and next stage buffer. However, since they are 5040 * treated as simple contiguous buffers in operation, so head 5041 * always should point 0. This may happen during drain-age. 5042 */ 5043 TRACET(4, track, "reset stage"); 5044 if (track->codec.filter) { 5045 KASSERT(track->codec.srcbuf.used == 0); 5046 track->codec.srcbuf.head = 0; 5047 } 5048 if (track->chvol.filter) { 5049 KASSERT(track->chvol.srcbuf.used == 0); 5050 track->chvol.srcbuf.head = 0; 5051 } 5052 if (track->chmix.filter) { 5053 KASSERT(track->chmix.srcbuf.used == 0); 5054 track->chmix.srcbuf.head = 0; 5055 } 5056 if (track->freq.filter) { 5057 KASSERT(track->freq.srcbuf.used == 0); 5058 track->freq.srcbuf.head = 0; 5059 } 5060 } 5061 5062 track->stamp++; 5063 5064 #if defined(AUDIO_DEBUG) 5065 if (audiodebug >= 3) { 5066 struct audio_track_debugbuf m; 5067 audio_track_bufstat(track, &m); 5068 TRACET(0, track, "end%s%s%s%s%s%s", 5069 m.outbuf, m.freq, m.chvol, m.chmix, m.codec, m.usrbuf); 5070 } 5071 #endif 5072 } 5073 5074 /* 5075 * Produce user output buffer for recording from input buffer. 5076 */ 5077 static void 5078 audio_track_record(audio_track_t *track) 5079 { 5080 audio_ring_t *outbuf; 5081 audio_ring_t *usrbuf; 5082 int count; 5083 int bytes; 5084 int framesize; 5085 5086 KASSERT(track); 5087 KASSERT(track->lock); 5088 5089 if (auring_get_contig_used(track->input) == 0) { 5090 TRACET(4, track, "input->used == 0"); 5091 return; 5092 } 5093 5094 /* Frequency conversion */ 5095 if (track->freq.filter) { 5096 if (track->freq.srcbuf.used > 0) { 5097 audio_apply_stage(track, &track->freq, true); 5098 /* XXX should input of freq be from beginning of buf? */ 5099 } 5100 } 5101 5102 /* Channel mix */ 5103 if (track->chmix.filter) 5104 audio_apply_stage(track, &track->chmix, false); 5105 5106 /* Channel volume */ 5107 if (track->chvol.filter) 5108 audio_apply_stage(track, &track->chvol, false); 5109 5110 /* Encoding conversion */ 5111 if (track->codec.filter) 5112 audio_apply_stage(track, &track->codec, false); 5113 5114 /* Copy outbuf to usrbuf */ 5115 outbuf = &track->outbuf; 5116 usrbuf = &track->usrbuf; 5117 /* usrbuf should be empty. */ 5118 KASSERT(usrbuf->used == 0); 5119 /* 5120 * framesize is always 1 byte or more since all formats supported 5121 * as usrfmt(=output) have 8bit or more stride. 5122 */ 5123 framesize = frametobyte(&outbuf->fmt, 1); 5124 KASSERT(framesize >= 1); 5125 /* 5126 * count is the number of frames to copy to usrbuf. 5127 * bytes is the number of bytes to copy to usrbuf. 5128 */ 5129 count = outbuf->used; 5130 count = uimin(count, track->usrbuf_blksize / framesize); 5131 bytes = count * framesize; 5132 if (auring_tail(usrbuf) + bytes < usrbuf->capacity) { 5133 memcpy((uint8_t *)usrbuf->mem + auring_tail(usrbuf), 5134 (uint8_t *)outbuf->mem + outbuf->head * framesize, 5135 bytes); 5136 auring_push(usrbuf, bytes); 5137 auring_take(outbuf, count); 5138 } else { 5139 int bytes1; 5140 int bytes2; 5141 5142 bytes1 = auring_get_contig_free(usrbuf); 5143 KASSERTMSG(bytes1 % framesize == 0, 5144 "bytes1=%d framesize=%d", bytes1, framesize); 5145 memcpy((uint8_t *)usrbuf->mem + auring_tail(usrbuf), 5146 (uint8_t *)outbuf->mem + outbuf->head * framesize, 5147 bytes1); 5148 auring_push(usrbuf, bytes1); 5149 auring_take(outbuf, bytes1 / framesize); 5150 5151 bytes2 = bytes - bytes1; 5152 memcpy((uint8_t *)usrbuf->mem + auring_tail(usrbuf), 5153 (uint8_t *)outbuf->mem + outbuf->head * framesize, 5154 bytes2); 5155 auring_push(usrbuf, bytes2); 5156 auring_take(outbuf, bytes2 / framesize); 5157 } 5158 5159 #if defined(AUDIO_DEBUG) 5160 if (audiodebug >= 3) { 5161 struct audio_track_debugbuf m; 5162 audio_track_bufstat(track, &m); 5163 TRACET(0, track, "end%s%s%s%s%s%s", 5164 m.freq, m.chvol, m.chmix, m.codec, m.outbuf, m.usrbuf); 5165 } 5166 #endif 5167 } 5168 5169 /* 5170 * Calculate blktime [msec] from mixer(.hwbuf.fmt). 5171 * Must be called with sc_exlock held. 5172 */ 5173 static u_int 5174 audio_mixer_calc_blktime(struct audio_softc *sc, audio_trackmixer_t *mixer) 5175 { 5176 audio_format2_t *fmt; 5177 u_int blktime; 5178 u_int frames_per_block; 5179 5180 KASSERT(sc->sc_exlock); 5181 5182 fmt = &mixer->hwbuf.fmt; 5183 blktime = sc->sc_blk_ms; 5184 5185 /* 5186 * If stride is not multiples of 8, special treatment is necessary. 5187 * For now, it is only x68k's vs(4), 4 bit/sample ADPCM. 5188 */ 5189 if (fmt->stride == 4) { 5190 frames_per_block = fmt->sample_rate * blktime / 1000; 5191 if ((frames_per_block & 1) != 0) 5192 blktime *= 2; 5193 } 5194 #ifdef DIAGNOSTIC 5195 else if (fmt->stride % NBBY != 0) { 5196 panic("unsupported HW stride %d", fmt->stride); 5197 } 5198 #endif 5199 5200 return blktime; 5201 } 5202 5203 /* 5204 * Initialize the mixer corresponding to the mode. 5205 * Set AUMODE_PLAY to the 'mode' for playback or AUMODE_RECORD for recording. 5206 * sc->sc_[pr]mixer (corresponding to the 'mode') must be zero-filled. 5207 * This function returns 0 on successful. Otherwise returns errno. 5208 * Must be called with sc_exlock held and without sc_lock held. 5209 */ 5210 static int 5211 audio_mixer_init(struct audio_softc *sc, int mode, 5212 const audio_format2_t *hwfmt, const audio_filter_reg_t *reg) 5213 { 5214 char codecbuf[64]; 5215 char blkdmsbuf[8]; 5216 audio_trackmixer_t *mixer; 5217 void (*softint_handler)(void *); 5218 int len; 5219 int blksize; 5220 int capacity; 5221 size_t bufsize; 5222 int hwblks; 5223 int blkms; 5224 int blkdms; 5225 int error; 5226 5227 KASSERT(hwfmt != NULL); 5228 KASSERT(reg != NULL); 5229 KASSERT(sc->sc_exlock); 5230 5231 error = 0; 5232 if (mode == AUMODE_PLAY) 5233 mixer = sc->sc_pmixer; 5234 else 5235 mixer = sc->sc_rmixer; 5236 5237 mixer->sc = sc; 5238 mixer->mode = mode; 5239 5240 mixer->hwbuf.fmt = *hwfmt; 5241 mixer->volume = 256; 5242 mixer->blktime_d = 1000; 5243 mixer->blktime_n = audio_mixer_calc_blktime(sc, mixer); 5244 sc->sc_blk_ms = mixer->blktime_n; 5245 hwblks = NBLKHW; 5246 5247 mixer->frames_per_block = frame_per_block(mixer, &mixer->hwbuf.fmt); 5248 blksize = frametobyte(&mixer->hwbuf.fmt, mixer->frames_per_block); 5249 if (sc->hw_if->round_blocksize) { 5250 int rounded; 5251 audio_params_t p = format2_to_params(&mixer->hwbuf.fmt); 5252 mutex_enter(sc->sc_lock); 5253 rounded = sc->hw_if->round_blocksize(sc->hw_hdl, blksize, 5254 mode, &p); 5255 mutex_exit(sc->sc_lock); 5256 TRACE(1, "round_blocksize %d -> %d", blksize, rounded); 5257 if (rounded != blksize) { 5258 if ((rounded * NBBY) % (mixer->hwbuf.fmt.stride * 5259 mixer->hwbuf.fmt.channels) != 0) { 5260 audio_printf(sc, 5261 "round_blocksize returned blocksize " 5262 "indivisible by framesize: " 5263 "blksize=%d rounded=%d " 5264 "stride=%ubit channels=%u\n", 5265 blksize, rounded, 5266 mixer->hwbuf.fmt.stride, 5267 mixer->hwbuf.fmt.channels); 5268 return EINVAL; 5269 } 5270 /* Recalculation */ 5271 blksize = rounded; 5272 mixer->frames_per_block = blksize * NBBY / 5273 (mixer->hwbuf.fmt.stride * 5274 mixer->hwbuf.fmt.channels); 5275 } 5276 } 5277 mixer->blktime_n = mixer->frames_per_block; 5278 mixer->blktime_d = mixer->hwbuf.fmt.sample_rate; 5279 5280 capacity = mixer->frames_per_block * hwblks; 5281 bufsize = frametobyte(&mixer->hwbuf.fmt, capacity); 5282 if (sc->hw_if->round_buffersize) { 5283 size_t rounded; 5284 mutex_enter(sc->sc_lock); 5285 rounded = sc->hw_if->round_buffersize(sc->hw_hdl, mode, 5286 bufsize); 5287 mutex_exit(sc->sc_lock); 5288 TRACE(1, "round_buffersize %zd -> %zd", bufsize, rounded); 5289 if (rounded < bufsize) { 5290 /* buffersize needs NBLKHW blocks at least. */ 5291 audio_printf(sc, 5292 "round_buffersize returned too small buffersize: " 5293 "buffersize=%zd blksize=%d\n", 5294 rounded, blksize); 5295 return EINVAL; 5296 } 5297 if (rounded % blksize != 0) { 5298 /* buffersize/blksize constraint mismatch? */ 5299 audio_printf(sc, 5300 "round_buffersize returned buffersize indivisible " 5301 "by blksize: buffersize=%zu blksize=%d\n", 5302 rounded, blksize); 5303 return EINVAL; 5304 } 5305 if (rounded != bufsize) { 5306 /* Recalculation */ 5307 bufsize = rounded; 5308 hwblks = bufsize / blksize; 5309 capacity = mixer->frames_per_block * hwblks; 5310 } 5311 } 5312 TRACE(1, "buffersize for %s = %zu", 5313 (mode == AUMODE_PLAY) ? "playback" : "recording", 5314 bufsize); 5315 mixer->hwbuf.capacity = capacity; 5316 5317 if (sc->hw_if->allocm) { 5318 /* sc_lock is not necessary for allocm */ 5319 mixer->hwbuf.mem = sc->hw_if->allocm(sc->hw_hdl, mode, bufsize); 5320 if (mixer->hwbuf.mem == NULL) { 5321 audio_printf(sc, "allocm(%zu) failed\n", bufsize); 5322 return ENOMEM; 5323 } 5324 } else { 5325 mixer->hwbuf.mem = kmem_alloc(bufsize, KM_SLEEP); 5326 } 5327 5328 /* From here, audio_mixer_destroy is necessary to exit. */ 5329 if (mode == AUMODE_PLAY) { 5330 cv_init(&mixer->outcv, "audiowr"); 5331 } else { 5332 cv_init(&mixer->outcv, "audiord"); 5333 } 5334 5335 if (mode == AUMODE_PLAY) { 5336 softint_handler = audio_softintr_wr; 5337 } else { 5338 softint_handler = audio_softintr_rd; 5339 } 5340 mixer->sih = softint_establish(SOFTINT_SERIAL | SOFTINT_MPSAFE, 5341 softint_handler, sc); 5342 if (mixer->sih == NULL) { 5343 device_printf(sc->sc_dev, "softint_establish failed\n"); 5344 goto abort; 5345 } 5346 5347 mixer->track_fmt.encoding = AUDIO_ENCODING_SLINEAR_NE; 5348 mixer->track_fmt.precision = AUDIO_INTERNAL_BITS; 5349 mixer->track_fmt.stride = AUDIO_INTERNAL_BITS; 5350 mixer->track_fmt.channels = mixer->hwbuf.fmt.channels; 5351 mixer->track_fmt.sample_rate = mixer->hwbuf.fmt.sample_rate; 5352 5353 if (mixer->hwbuf.fmt.encoding == AUDIO_ENCODING_SLINEAR_OE && 5354 mixer->hwbuf.fmt.precision == AUDIO_INTERNAL_BITS) { 5355 mixer->swap_endian = true; 5356 TRACE(1, "swap_endian"); 5357 } 5358 5359 if (mode == AUMODE_PLAY) { 5360 /* Mixing buffer */ 5361 mixer->mixfmt = mixer->track_fmt; 5362 mixer->mixfmt.precision *= 2; 5363 mixer->mixfmt.stride *= 2; 5364 /* XXX TODO: use some macros? */ 5365 len = mixer->frames_per_block * mixer->mixfmt.channels * 5366 mixer->mixfmt.stride / NBBY; 5367 mixer->mixsample = audio_realloc(mixer->mixsample, len); 5368 } else { 5369 /* No mixing buffer for recording */ 5370 } 5371 5372 if (reg->codec) { 5373 mixer->codec = reg->codec; 5374 mixer->codecarg.context = reg->context; 5375 if (mode == AUMODE_PLAY) { 5376 mixer->codecarg.srcfmt = &mixer->track_fmt; 5377 mixer->codecarg.dstfmt = &mixer->hwbuf.fmt; 5378 } else { 5379 mixer->codecarg.srcfmt = &mixer->hwbuf.fmt; 5380 mixer->codecarg.dstfmt = &mixer->track_fmt; 5381 } 5382 mixer->codecbuf.fmt = mixer->track_fmt; 5383 mixer->codecbuf.capacity = mixer->frames_per_block; 5384 len = auring_bytelen(&mixer->codecbuf); 5385 mixer->codecbuf.mem = audio_realloc(mixer->codecbuf.mem, len); 5386 } 5387 5388 /* Succeeded so display it. */ 5389 codecbuf[0] = '\0'; 5390 if (mixer->codec || mixer->swap_endian) { 5391 snprintf(codecbuf, sizeof(codecbuf), " %s %s:%d", 5392 (mode == AUMODE_PLAY) ? "->" : "<-", 5393 audio_encoding_name(mixer->hwbuf.fmt.encoding), 5394 mixer->hwbuf.fmt.precision); 5395 } 5396 blkms = mixer->blktime_n * 1000 / mixer->blktime_d; 5397 blkdms = (mixer->blktime_n * 10000 / mixer->blktime_d) % 10; 5398 blkdmsbuf[0] = '\0'; 5399 if (blkdms != 0) { 5400 snprintf(blkdmsbuf, sizeof(blkdmsbuf), ".%1d", blkdms); 5401 } 5402 aprint_normal_dev(sc->sc_dev, 5403 "%s:%d%s %dch %dHz, blk %d bytes (%d%sms) for %s\n", 5404 audio_encoding_name(mixer->track_fmt.encoding), 5405 mixer->track_fmt.precision, 5406 codecbuf, 5407 mixer->track_fmt.channels, 5408 mixer->track_fmt.sample_rate, 5409 blksize, 5410 blkms, blkdmsbuf, 5411 (mode == AUMODE_PLAY) ? "playback" : "recording"); 5412 5413 return 0; 5414 5415 abort: 5416 audio_mixer_destroy(sc, mixer); 5417 return error; 5418 } 5419 5420 /* 5421 * Releases all resources of 'mixer'. 5422 * Note that it does not release the memory area of 'mixer' itself. 5423 * Must be called with sc_exlock held and without sc_lock held. 5424 */ 5425 static void 5426 audio_mixer_destroy(struct audio_softc *sc, audio_trackmixer_t *mixer) 5427 { 5428 int bufsize; 5429 5430 KASSERT(sc->sc_exlock == 1); 5431 5432 bufsize = frametobyte(&mixer->hwbuf.fmt, mixer->hwbuf.capacity); 5433 5434 if (mixer->hwbuf.mem != NULL) { 5435 if (sc->hw_if->freem) { 5436 /* sc_lock is not necessary for freem */ 5437 sc->hw_if->freem(sc->hw_hdl, mixer->hwbuf.mem, bufsize); 5438 } else { 5439 kmem_free(mixer->hwbuf.mem, bufsize); 5440 } 5441 mixer->hwbuf.mem = NULL; 5442 } 5443 5444 audio_free(mixer->codecbuf.mem); 5445 audio_free(mixer->mixsample); 5446 5447 cv_destroy(&mixer->outcv); 5448 5449 if (mixer->sih) { 5450 softint_disestablish(mixer->sih); 5451 mixer->sih = NULL; 5452 } 5453 } 5454 5455 /* 5456 * Starts playback mixer. 5457 * Must be called only if sc_pbusy is false. 5458 * Must be called with sc_lock && sc_exlock held. 5459 * Must not be called from the interrupt context. 5460 */ 5461 static void 5462 audio_pmixer_start(struct audio_softc *sc, bool force) 5463 { 5464 audio_trackmixer_t *mixer; 5465 int minimum; 5466 5467 KASSERT(mutex_owned(sc->sc_lock)); 5468 KASSERT(sc->sc_exlock); 5469 KASSERT(sc->sc_pbusy == false); 5470 5471 mutex_enter(sc->sc_intr_lock); 5472 5473 mixer = sc->sc_pmixer; 5474 TRACE(2, "%smixseq=%d hwseq=%d hwbuf=%d/%d/%d%s", 5475 (audiodebug >= 3) ? "begin " : "", 5476 (int)mixer->mixseq, (int)mixer->hwseq, 5477 mixer->hwbuf.head, mixer->hwbuf.used, mixer->hwbuf.capacity, 5478 force ? " force" : ""); 5479 5480 /* Need two blocks to start normally. */ 5481 minimum = (force) ? 1 : 2; 5482 while (mixer->hwbuf.used < mixer->frames_per_block * minimum) { 5483 audio_pmixer_process(sc); 5484 } 5485 5486 /* Start output */ 5487 audio_pmixer_output(sc); 5488 sc->sc_pbusy = true; 5489 5490 TRACE(3, "end mixseq=%d hwseq=%d hwbuf=%d/%d/%d", 5491 (int)mixer->mixseq, (int)mixer->hwseq, 5492 mixer->hwbuf.head, mixer->hwbuf.used, mixer->hwbuf.capacity); 5493 5494 mutex_exit(sc->sc_intr_lock); 5495 } 5496 5497 /* 5498 * When playing back with MD filter: 5499 * 5500 * track track ... 5501 * v v 5502 * + mix (with aint2_t) 5503 * | master volume (with aint2_t) 5504 * v 5505 * mixsample [::::] wide-int 1 block (ring) buffer 5506 * | 5507 * | convert aint2_t -> aint_t 5508 * v 5509 * codecbuf [....] 1 block (ring) buffer 5510 * | 5511 * | convert to hw format 5512 * v 5513 * hwbuf [............] NBLKHW blocks ring buffer 5514 * 5515 * When playing back without MD filter: 5516 * 5517 * mixsample [::::] wide-int 1 block (ring) buffer 5518 * | 5519 * | convert aint2_t -> aint_t 5520 * | (with byte swap if necessary) 5521 * v 5522 * hwbuf [............] NBLKHW blocks ring buffer 5523 * 5524 * mixsample: slinear_NE, wide internal precision, HW ch, HW freq. 5525 * codecbuf: slinear_NE, internal precision, HW ch, HW freq. 5526 * hwbuf: HW encoding, HW precision, HW ch, HW freq. 5527 */ 5528 5529 /* 5530 * Performs track mixing and converts it to hwbuf. 5531 * Note that this function doesn't transfer hwbuf to hardware. 5532 * Must be called with sc_intr_lock held. 5533 */ 5534 static void 5535 audio_pmixer_process(struct audio_softc *sc) 5536 { 5537 audio_trackmixer_t *mixer; 5538 audio_file_t *f; 5539 int frame_count; 5540 int sample_count; 5541 int mixed; 5542 int i; 5543 aint2_t *m; 5544 aint_t *h; 5545 5546 mixer = sc->sc_pmixer; 5547 5548 frame_count = mixer->frames_per_block; 5549 KASSERTMSG(auring_get_contig_free(&mixer->hwbuf) >= frame_count, 5550 "auring_get_contig_free()=%d frame_count=%d", 5551 auring_get_contig_free(&mixer->hwbuf), frame_count); 5552 sample_count = frame_count * mixer->mixfmt.channels; 5553 5554 mixer->mixseq++; 5555 5556 /* Mix all tracks */ 5557 mixed = 0; 5558 SLIST_FOREACH(f, &sc->sc_files, entry) { 5559 audio_track_t *track = f->ptrack; 5560 5561 if (track == NULL) 5562 continue; 5563 5564 if (track->is_pause) { 5565 TRACET(4, track, "skip; paused"); 5566 continue; 5567 } 5568 5569 /* Skip if the track is used by process context. */ 5570 if (audio_track_lock_tryenter(track) == false) { 5571 TRACET(4, track, "skip; in use"); 5572 continue; 5573 } 5574 5575 /* Emulate mmap'ped track */ 5576 if (track->mmapped) { 5577 auring_push(&track->usrbuf, track->usrbuf_blksize); 5578 TRACET(4, track, "mmap; usr=%d/%d/C%d", 5579 track->usrbuf.head, 5580 track->usrbuf.used, 5581 track->usrbuf.capacity); 5582 } 5583 5584 if (track->outbuf.used < mixer->frames_per_block && 5585 track->usrbuf.used > 0) { 5586 TRACET(4, track, "process"); 5587 audio_track_play(track); 5588 } 5589 5590 if (track->outbuf.used > 0) { 5591 mixed = audio_pmixer_mix_track(mixer, track, mixed); 5592 } else { 5593 TRACET(4, track, "skip; empty"); 5594 } 5595 5596 audio_track_lock_exit(track); 5597 } 5598 5599 if (mixed == 0) { 5600 /* Silence */ 5601 memset(mixer->mixsample, 0, 5602 frametobyte(&mixer->mixfmt, frame_count)); 5603 } else { 5604 if (mixed > 1) { 5605 /* If there are multiple tracks, do auto gain control */ 5606 audio_pmixer_agc(mixer, sample_count); 5607 } 5608 5609 /* Apply master volume */ 5610 if (mixer->volume < 256) { 5611 m = mixer->mixsample; 5612 for (i = 0; i < sample_count; i++) { 5613 *m = AUDIO_SCALEDOWN(*m * mixer->volume, 8); 5614 m++; 5615 } 5616 5617 /* 5618 * Recover the volume gradually at the pace of 5619 * several times per second. If it's too fast, you 5620 * can recognize that the volume changes up and down 5621 * quickly and it's not so comfortable. 5622 */ 5623 mixer->voltimer += mixer->blktime_n; 5624 if (mixer->voltimer * 4 >= mixer->blktime_d) { 5625 mixer->volume++; 5626 mixer->voltimer = 0; 5627 #if defined(AUDIO_DEBUG_AGC) 5628 TRACE(1, "volume recover: %d", mixer->volume); 5629 #endif 5630 } 5631 } 5632 } 5633 5634 /* 5635 * The rest is the hardware part. 5636 */ 5637 5638 if (mixer->codec) { 5639 h = auring_tailptr_aint(&mixer->codecbuf); 5640 } else { 5641 h = auring_tailptr_aint(&mixer->hwbuf); 5642 } 5643 5644 m = mixer->mixsample; 5645 if (mixer->swap_endian) { 5646 for (i = 0; i < sample_count; i++) { 5647 *h++ = bswap16(*m++); 5648 } 5649 } else { 5650 for (i = 0; i < sample_count; i++) { 5651 *h++ = *m++; 5652 } 5653 } 5654 5655 /* Hardware driver's codec */ 5656 if (mixer->codec) { 5657 auring_push(&mixer->codecbuf, frame_count); 5658 mixer->codecarg.src = auring_headptr(&mixer->codecbuf); 5659 mixer->codecarg.dst = auring_tailptr(&mixer->hwbuf); 5660 mixer->codecarg.count = frame_count; 5661 mixer->codec(&mixer->codecarg); 5662 auring_take(&mixer->codecbuf, mixer->codecarg.count); 5663 } 5664 5665 auring_push(&mixer->hwbuf, frame_count); 5666 5667 TRACE(4, "done mixseq=%d hwbuf=%d/%d/%d%s", 5668 (int)mixer->mixseq, 5669 mixer->hwbuf.head, mixer->hwbuf.used, mixer->hwbuf.capacity, 5670 (mixed == 0) ? " silent" : ""); 5671 } 5672 5673 /* 5674 * Do auto gain control. 5675 * Must be called sc_intr_lock held. 5676 */ 5677 static void 5678 audio_pmixer_agc(audio_trackmixer_t *mixer, int sample_count) 5679 { 5680 struct audio_softc *sc __unused; 5681 aint2_t val; 5682 aint2_t maxval; 5683 aint2_t minval; 5684 aint2_t over_plus; 5685 aint2_t over_minus; 5686 aint2_t *m; 5687 int newvol; 5688 int i; 5689 5690 sc = mixer->sc; 5691 5692 /* Overflow detection */ 5693 maxval = AINT_T_MAX; 5694 minval = AINT_T_MIN; 5695 m = mixer->mixsample; 5696 for (i = 0; i < sample_count; i++) { 5697 val = *m++; 5698 if (val > maxval) 5699 maxval = val; 5700 else if (val < minval) 5701 minval = val; 5702 } 5703 5704 /* Absolute value of overflowed amount */ 5705 over_plus = maxval - AINT_T_MAX; 5706 over_minus = AINT_T_MIN - minval; 5707 5708 if (over_plus > 0 || over_minus > 0) { 5709 if (over_plus > over_minus) { 5710 newvol = (int)((aint2_t)AINT_T_MAX * 256 / maxval); 5711 } else { 5712 newvol = (int)((aint2_t)AINT_T_MIN * 256 / minval); 5713 } 5714 5715 /* 5716 * Change the volume only if new one is smaller. 5717 * Reset the timer even if the volume isn't changed. 5718 */ 5719 if (newvol <= mixer->volume) { 5720 mixer->volume = newvol; 5721 mixer->voltimer = 0; 5722 #if defined(AUDIO_DEBUG_AGC) 5723 TRACE(1, "auto volume adjust: %d", mixer->volume); 5724 #endif 5725 } 5726 } 5727 } 5728 5729 /* 5730 * Mix one track. 5731 * 'mixed' specifies the number of tracks mixed so far. 5732 * It returns the number of tracks mixed. In other words, it returns 5733 * mixed + 1 if this track is mixed. 5734 */ 5735 static int 5736 audio_pmixer_mix_track(audio_trackmixer_t *mixer, audio_track_t *track, 5737 int mixed) 5738 { 5739 int count; 5740 int sample_count; 5741 int remain; 5742 int i; 5743 const aint_t *s; 5744 aint2_t *d; 5745 5746 /* XXX TODO: Is this necessary for now? */ 5747 if (mixer->mixseq < track->seq) 5748 return mixed; 5749 5750 count = auring_get_contig_used(&track->outbuf); 5751 count = uimin(count, mixer->frames_per_block); 5752 5753 s = auring_headptr_aint(&track->outbuf); 5754 d = mixer->mixsample; 5755 5756 /* 5757 * Apply track volume with double-sized integer and perform 5758 * additive synthesis. 5759 * 5760 * XXX If you limit the track volume to 1.0 or less (<= 256), 5761 * it would be better to do this in the track conversion stage 5762 * rather than here. However, if you accept the volume to 5763 * be greater than 1.0 (> 256), it's better to do it here. 5764 * Because the operation here is done by double-sized integer. 5765 */ 5766 sample_count = count * mixer->mixfmt.channels; 5767 if (mixed == 0) { 5768 /* If this is the first track, assignment can be used. */ 5769 #if defined(AUDIO_SUPPORT_TRACK_VOLUME) 5770 if (track->volume != 256) { 5771 for (i = 0; i < sample_count; i++) { 5772 aint2_t v; 5773 v = *s++; 5774 *d++ = AUDIO_SCALEDOWN(v * track->volume, 8) 5775 } 5776 } else 5777 #endif 5778 { 5779 for (i = 0; i < sample_count; i++) { 5780 *d++ = ((aint2_t)*s++); 5781 } 5782 } 5783 /* Fill silence if the first track is not filled. */ 5784 for (; i < mixer->frames_per_block * mixer->mixfmt.channels; i++) 5785 *d++ = 0; 5786 } else { 5787 /* If this is the second or later, add it. */ 5788 #if defined(AUDIO_SUPPORT_TRACK_VOLUME) 5789 if (track->volume != 256) { 5790 for (i = 0; i < sample_count; i++) { 5791 aint2_t v; 5792 v = *s++; 5793 *d++ += AUDIO_SCALEDOWN(v * track->volume, 8); 5794 } 5795 } else 5796 #endif 5797 { 5798 for (i = 0; i < sample_count; i++) { 5799 *d++ += ((aint2_t)*s++); 5800 } 5801 } 5802 } 5803 5804 auring_take(&track->outbuf, count); 5805 /* 5806 * The counters have to align block even if outbuf is less than 5807 * one block. XXX Is this still necessary? 5808 */ 5809 remain = mixer->frames_per_block - count; 5810 if (__predict_false(remain != 0)) { 5811 auring_push(&track->outbuf, remain); 5812 auring_take(&track->outbuf, remain); 5813 } 5814 5815 /* 5816 * Update track sequence. 5817 * mixseq has previous value yet at this point. 5818 */ 5819 track->seq = mixer->mixseq + 1; 5820 5821 return mixed + 1; 5822 } 5823 5824 /* 5825 * Output one block from hwbuf to HW. 5826 * Must be called with sc_intr_lock held. 5827 */ 5828 static void 5829 audio_pmixer_output(struct audio_softc *sc) 5830 { 5831 audio_trackmixer_t *mixer; 5832 audio_params_t params; 5833 void *start; 5834 void *end; 5835 int blksize; 5836 int error; 5837 5838 mixer = sc->sc_pmixer; 5839 TRACE(4, "pbusy=%d hwbuf=%d/%d/%d", 5840 sc->sc_pbusy, 5841 mixer->hwbuf.head, mixer->hwbuf.used, mixer->hwbuf.capacity); 5842 KASSERTMSG(mixer->hwbuf.used >= mixer->frames_per_block, 5843 "mixer->hwbuf.used=%d mixer->frames_per_block=%d", 5844 mixer->hwbuf.used, mixer->frames_per_block); 5845 5846 blksize = frametobyte(&mixer->hwbuf.fmt, mixer->frames_per_block); 5847 5848 if (sc->hw_if->trigger_output) { 5849 /* trigger (at once) */ 5850 if (!sc->sc_pbusy) { 5851 start = mixer->hwbuf.mem; 5852 end = (uint8_t *)start + auring_bytelen(&mixer->hwbuf); 5853 params = format2_to_params(&mixer->hwbuf.fmt); 5854 5855 error = sc->hw_if->trigger_output(sc->hw_hdl, 5856 start, end, blksize, audio_pintr, sc, ¶ms); 5857 if (error) { 5858 audio_printf(sc, 5859 "trigger_output failed: errno=%d\n", 5860 error); 5861 return; 5862 } 5863 } 5864 } else { 5865 /* start (everytime) */ 5866 start = auring_headptr(&mixer->hwbuf); 5867 5868 error = sc->hw_if->start_output(sc->hw_hdl, 5869 start, blksize, audio_pintr, sc); 5870 if (error) { 5871 audio_printf(sc, 5872 "start_output failed: errno=%d\n", error); 5873 return; 5874 } 5875 } 5876 } 5877 5878 /* 5879 * This is an interrupt handler for playback. 5880 * It is called with sc_intr_lock held. 5881 * 5882 * It is usually called from hardware interrupt. However, note that 5883 * for some drivers (e.g. uaudio) it is called from software interrupt. 5884 */ 5885 static void 5886 audio_pintr(void *arg) 5887 { 5888 struct audio_softc *sc; 5889 audio_trackmixer_t *mixer; 5890 5891 sc = arg; 5892 KASSERT(mutex_owned(sc->sc_intr_lock)); 5893 5894 if (sc->sc_dying) 5895 return; 5896 if (sc->sc_pbusy == false) { 5897 #if defined(DIAGNOSTIC) 5898 audio_printf(sc, "DIAGNOSTIC: %s raised stray interrupt\n", 5899 device_xname(sc->hw_dev)); 5900 #endif 5901 return; 5902 } 5903 5904 mixer = sc->sc_pmixer; 5905 mixer->hw_complete_counter += mixer->frames_per_block; 5906 mixer->hwseq++; 5907 5908 auring_take(&mixer->hwbuf, mixer->frames_per_block); 5909 5910 TRACE(4, 5911 "HW_INT ++hwseq=%" PRIu64 " cmplcnt=%" PRIu64 " hwbuf=%d/%d/%d", 5912 mixer->hwseq, mixer->hw_complete_counter, 5913 mixer->hwbuf.head, mixer->hwbuf.used, mixer->hwbuf.capacity); 5914 5915 #if defined(AUDIO_HW_SINGLE_BUFFER) 5916 /* 5917 * Create a new block here and output it immediately. 5918 * It makes a latency lower but needs machine power. 5919 */ 5920 audio_pmixer_process(sc); 5921 audio_pmixer_output(sc); 5922 #else 5923 /* 5924 * It is called when block N output is done. 5925 * Output immediately block N+1 created by the last interrupt. 5926 * And then create block N+2 for the next interrupt. 5927 * This method makes playback robust even on slower machines. 5928 * Instead the latency is increased by one block. 5929 */ 5930 5931 /* At first, output ready block. */ 5932 if (mixer->hwbuf.used >= mixer->frames_per_block) { 5933 audio_pmixer_output(sc); 5934 } 5935 5936 bool later = false; 5937 5938 if (mixer->hwbuf.used < mixer->frames_per_block) { 5939 later = true; 5940 } 5941 5942 /* Then, process next block. */ 5943 audio_pmixer_process(sc); 5944 5945 if (later) { 5946 audio_pmixer_output(sc); 5947 } 5948 #endif 5949 5950 /* 5951 * When this interrupt is the real hardware interrupt, disabling 5952 * preemption here is not necessary. But some drivers (e.g. uaudio) 5953 * emulate it by software interrupt, so kpreempt_disable is necessary. 5954 */ 5955 kpreempt_disable(); 5956 softint_schedule(mixer->sih); 5957 kpreempt_enable(); 5958 } 5959 5960 /* 5961 * Starts record mixer. 5962 * Must be called only if sc_rbusy is false. 5963 * Must be called with sc_lock && sc_exlock held. 5964 * Must not be called from the interrupt context. 5965 */ 5966 static void 5967 audio_rmixer_start(struct audio_softc *sc) 5968 { 5969 5970 KASSERT(mutex_owned(sc->sc_lock)); 5971 KASSERT(sc->sc_exlock); 5972 KASSERT(sc->sc_rbusy == false); 5973 5974 mutex_enter(sc->sc_intr_lock); 5975 5976 TRACE(2, "%s", (audiodebug >= 3) ? "begin" : ""); 5977 audio_rmixer_input(sc); 5978 sc->sc_rbusy = true; 5979 TRACE(3, "end"); 5980 5981 mutex_exit(sc->sc_intr_lock); 5982 } 5983 5984 /* 5985 * When recording with MD filter: 5986 * 5987 * hwbuf [............] NBLKHW blocks ring buffer 5988 * | 5989 * | convert from hw format 5990 * v 5991 * codecbuf [....] 1 block (ring) buffer 5992 * | | 5993 * v v 5994 * track track ... 5995 * 5996 * When recording without MD filter: 5997 * 5998 * hwbuf [............] NBLKHW blocks ring buffer 5999 * | | 6000 * v v 6001 * track track ... 6002 * 6003 * hwbuf: HW encoding, HW precision, HW ch, HW freq. 6004 * codecbuf: slinear_NE, internal precision, HW ch, HW freq. 6005 */ 6006 6007 /* 6008 * Distribute a recorded block to all recording tracks. 6009 */ 6010 static void 6011 audio_rmixer_process(struct audio_softc *sc) 6012 { 6013 audio_trackmixer_t *mixer; 6014 audio_ring_t *mixersrc; 6015 audio_file_t *f; 6016 aint_t *p; 6017 int count; 6018 int bytes; 6019 int i; 6020 6021 mixer = sc->sc_rmixer; 6022 6023 /* 6024 * count is the number of frames to be retrieved this time. 6025 * count should be one block. 6026 */ 6027 count = auring_get_contig_used(&mixer->hwbuf); 6028 count = uimin(count, mixer->frames_per_block); 6029 if (count <= 0) { 6030 TRACE(4, "count %d: too short", count); 6031 return; 6032 } 6033 bytes = frametobyte(&mixer->track_fmt, count); 6034 6035 /* Hardware driver's codec */ 6036 if (mixer->codec) { 6037 mixer->codecarg.src = auring_headptr(&mixer->hwbuf); 6038 mixer->codecarg.dst = auring_tailptr(&mixer->codecbuf); 6039 mixer->codecarg.count = count; 6040 mixer->codec(&mixer->codecarg); 6041 auring_take(&mixer->hwbuf, mixer->codecarg.count); 6042 auring_push(&mixer->codecbuf, mixer->codecarg.count); 6043 mixersrc = &mixer->codecbuf; 6044 } else { 6045 mixersrc = &mixer->hwbuf; 6046 } 6047 6048 if (mixer->swap_endian) { 6049 /* inplace conversion */ 6050 p = auring_headptr_aint(mixersrc); 6051 for (i = 0; i < count * mixer->track_fmt.channels; i++, p++) { 6052 *p = bswap16(*p); 6053 } 6054 } 6055 6056 /* Distribute to all tracks. */ 6057 SLIST_FOREACH(f, &sc->sc_files, entry) { 6058 audio_track_t *track = f->rtrack; 6059 audio_ring_t *input; 6060 6061 if (track == NULL) 6062 continue; 6063 6064 if (track->is_pause) { 6065 TRACET(4, track, "skip; paused"); 6066 continue; 6067 } 6068 6069 if (audio_track_lock_tryenter(track) == false) { 6070 TRACET(4, track, "skip; in use"); 6071 continue; 6072 } 6073 6074 /* 6075 * If the track buffer has less than one block of free space, 6076 * make one block free. 6077 */ 6078 input = track->input; 6079 if (input->capacity - input->used < mixer->frames_per_block) { 6080 int drops = mixer->frames_per_block - 6081 (input->capacity - input->used); 6082 track->dropframes += drops; 6083 TRACET(4, track, "drop %d frames: inp=%d/%d/%d", 6084 drops, 6085 input->head, input->used, input->capacity); 6086 auring_take(input, drops); 6087 } 6088 6089 KASSERTMSG(auring_tail(input) % mixer->frames_per_block == 0, 6090 "inputtail=%d mixer->frames_per_block=%d", 6091 auring_tail(input), mixer->frames_per_block); 6092 memcpy(auring_tailptr_aint(input), 6093 auring_headptr_aint(mixersrc), 6094 bytes); 6095 auring_push(input, count); 6096 6097 track->stamp++; 6098 6099 audio_track_lock_exit(track); 6100 } 6101 6102 auring_take(mixersrc, count); 6103 } 6104 6105 /* 6106 * Input one block from HW to hwbuf. 6107 * Must be called with sc_intr_lock held. 6108 */ 6109 static void 6110 audio_rmixer_input(struct audio_softc *sc) 6111 { 6112 audio_trackmixer_t *mixer; 6113 audio_params_t params; 6114 void *start; 6115 void *end; 6116 int blksize; 6117 int error; 6118 6119 mixer = sc->sc_rmixer; 6120 blksize = frametobyte(&mixer->hwbuf.fmt, mixer->frames_per_block); 6121 6122 if (sc->hw_if->trigger_input) { 6123 /* trigger (at once) */ 6124 if (!sc->sc_rbusy) { 6125 start = mixer->hwbuf.mem; 6126 end = (uint8_t *)start + auring_bytelen(&mixer->hwbuf); 6127 params = format2_to_params(&mixer->hwbuf.fmt); 6128 6129 error = sc->hw_if->trigger_input(sc->hw_hdl, 6130 start, end, blksize, audio_rintr, sc, ¶ms); 6131 if (error) { 6132 audio_printf(sc, 6133 "trigger_input failed: errno=%d\n", 6134 error); 6135 return; 6136 } 6137 } 6138 } else { 6139 /* start (everytime) */ 6140 start = auring_tailptr(&mixer->hwbuf); 6141 6142 error = sc->hw_if->start_input(sc->hw_hdl, 6143 start, blksize, audio_rintr, sc); 6144 if (error) { 6145 audio_printf(sc, 6146 "start_input failed: errno=%d\n", error); 6147 return; 6148 } 6149 } 6150 } 6151 6152 /* 6153 * This is an interrupt handler for recording. 6154 * It is called with sc_intr_lock. 6155 * 6156 * It is usually called from hardware interrupt. However, note that 6157 * for some drivers (e.g. uaudio) it is called from software interrupt. 6158 */ 6159 static void 6160 audio_rintr(void *arg) 6161 { 6162 struct audio_softc *sc; 6163 audio_trackmixer_t *mixer; 6164 6165 sc = arg; 6166 KASSERT(mutex_owned(sc->sc_intr_lock)); 6167 6168 if (sc->sc_dying) 6169 return; 6170 if (sc->sc_rbusy == false) { 6171 #if defined(DIAGNOSTIC) 6172 audio_printf(sc, "DIAGNOSTIC: %s raised stray interrupt\n", 6173 device_xname(sc->hw_dev)); 6174 #endif 6175 return; 6176 } 6177 6178 mixer = sc->sc_rmixer; 6179 mixer->hw_complete_counter += mixer->frames_per_block; 6180 mixer->hwseq++; 6181 6182 auring_push(&mixer->hwbuf, mixer->frames_per_block); 6183 6184 TRACE(4, 6185 "HW_INT ++hwseq=%" PRIu64 " cmplcnt=%" PRIu64 " hwbuf=%d/%d/%d", 6186 mixer->hwseq, mixer->hw_complete_counter, 6187 mixer->hwbuf.head, mixer->hwbuf.used, mixer->hwbuf.capacity); 6188 6189 /* Distrubute recorded block */ 6190 audio_rmixer_process(sc); 6191 6192 /* Request next block */ 6193 audio_rmixer_input(sc); 6194 6195 /* 6196 * When this interrupt is the real hardware interrupt, disabling 6197 * preemption here is not necessary. But some drivers (e.g. uaudio) 6198 * emulate it by software interrupt, so kpreempt_disable is necessary. 6199 */ 6200 kpreempt_disable(); 6201 softint_schedule(mixer->sih); 6202 kpreempt_enable(); 6203 } 6204 6205 /* 6206 * Halts playback mixer. 6207 * This function also clears related parameters, so call this function 6208 * instead of calling halt_output directly. 6209 * Must be called only if sc_pbusy is true. 6210 * Must be called with sc_lock && sc_exlock held. 6211 */ 6212 static int 6213 audio_pmixer_halt(struct audio_softc *sc) 6214 { 6215 int error; 6216 6217 TRACE(2, "called"); 6218 KASSERT(mutex_owned(sc->sc_lock)); 6219 KASSERT(sc->sc_exlock); 6220 6221 mutex_enter(sc->sc_intr_lock); 6222 error = sc->hw_if->halt_output(sc->hw_hdl); 6223 6224 /* Halts anyway even if some error has occurred. */ 6225 sc->sc_pbusy = false; 6226 sc->sc_pmixer->hwbuf.head = 0; 6227 sc->sc_pmixer->hwbuf.used = 0; 6228 sc->sc_pmixer->mixseq = 0; 6229 sc->sc_pmixer->hwseq = 0; 6230 mutex_exit(sc->sc_intr_lock); 6231 6232 return error; 6233 } 6234 6235 /* 6236 * Halts recording mixer. 6237 * This function also clears related parameters, so call this function 6238 * instead of calling halt_input directly. 6239 * Must be called only if sc_rbusy is true. 6240 * Must be called with sc_lock && sc_exlock held. 6241 */ 6242 static int 6243 audio_rmixer_halt(struct audio_softc *sc) 6244 { 6245 int error; 6246 6247 TRACE(2, "called"); 6248 KASSERT(mutex_owned(sc->sc_lock)); 6249 KASSERT(sc->sc_exlock); 6250 6251 mutex_enter(sc->sc_intr_lock); 6252 error = sc->hw_if->halt_input(sc->hw_hdl); 6253 6254 /* Halts anyway even if some error has occurred. */ 6255 sc->sc_rbusy = false; 6256 sc->sc_rmixer->hwbuf.head = 0; 6257 sc->sc_rmixer->hwbuf.used = 0; 6258 sc->sc_rmixer->mixseq = 0; 6259 sc->sc_rmixer->hwseq = 0; 6260 mutex_exit(sc->sc_intr_lock); 6261 6262 return error; 6263 } 6264 6265 /* 6266 * Flush this track. 6267 * Halts all operations, clears all buffers, reset error counters. 6268 * XXX I'm not sure... 6269 */ 6270 static void 6271 audio_track_clear(struct audio_softc *sc, audio_track_t *track) 6272 { 6273 6274 KASSERT(track); 6275 TRACET(3, track, "clear"); 6276 6277 audio_track_lock_enter(track); 6278 6279 /* Clear all internal parameters. */ 6280 track->usrbuf.used = 0; 6281 track->usrbuf.head = 0; 6282 if (track->codec.filter) { 6283 track->codec.srcbuf.used = 0; 6284 track->codec.srcbuf.head = 0; 6285 } 6286 if (track->chvol.filter) { 6287 track->chvol.srcbuf.used = 0; 6288 track->chvol.srcbuf.head = 0; 6289 } 6290 if (track->chmix.filter) { 6291 track->chmix.srcbuf.used = 0; 6292 track->chmix.srcbuf.head = 0; 6293 } 6294 if (track->freq.filter) { 6295 track->freq.srcbuf.used = 0; 6296 track->freq.srcbuf.head = 0; 6297 if (track->freq_step < 65536) 6298 track->freq_current = 65536; 6299 else 6300 track->freq_current = 0; 6301 memset(track->freq_prev, 0, sizeof(track->freq_prev)); 6302 memset(track->freq_curr, 0, sizeof(track->freq_curr)); 6303 } 6304 /* Clear buffer, then operation halts naturally. */ 6305 track->outbuf.used = 0; 6306 6307 /* Clear counters. */ 6308 track->stamp = 0; 6309 track->last_stamp = 0; 6310 track->dropframes = 0; 6311 6312 audio_track_lock_exit(track); 6313 } 6314 6315 /* 6316 * Drain the track. 6317 * track must be present and for playback. 6318 * If successful, it returns 0. Otherwise returns errno. 6319 * Must be called with sc_lock held. 6320 */ 6321 static int 6322 audio_track_drain(struct audio_softc *sc, audio_track_t *track) 6323 { 6324 audio_trackmixer_t *mixer; 6325 int done; 6326 int error; 6327 6328 KASSERT(track); 6329 TRACET(3, track, "start"); 6330 mixer = track->mixer; 6331 KASSERT(mutex_owned(sc->sc_lock)); 6332 6333 /* Ignore them if pause. */ 6334 if (track->is_pause) { 6335 TRACET(3, track, "pause -> clear"); 6336 track->pstate = AUDIO_STATE_CLEAR; 6337 } 6338 /* Terminate early here if there is no data in the track. */ 6339 if (track->pstate == AUDIO_STATE_CLEAR) { 6340 TRACET(3, track, "no need to drain"); 6341 return 0; 6342 } 6343 track->pstate = AUDIO_STATE_DRAINING; 6344 6345 for (;;) { 6346 /* I want to display it before condition evaluation. */ 6347 TRACET(3, track, "pid=%d.%d trkseq=%d hwseq=%d out=%d/%d/%d", 6348 (int)curproc->p_pid, (int)curlwp->l_lid, 6349 (int)track->seq, (int)mixer->hwseq, 6350 track->outbuf.head, track->outbuf.used, 6351 track->outbuf.capacity); 6352 6353 /* Condition to terminate */ 6354 audio_track_lock_enter(track); 6355 done = (track->usrbuf.used < frametobyte(&track->inputfmt, 1) && 6356 track->outbuf.used == 0 && 6357 track->seq <= mixer->hwseq); 6358 audio_track_lock_exit(track); 6359 if (done) 6360 break; 6361 6362 TRACET(3, track, "sleep"); 6363 error = audio_track_waitio(sc, track); 6364 if (error) 6365 return error; 6366 6367 /* XXX call audio_track_play here ? */ 6368 } 6369 6370 track->pstate = AUDIO_STATE_CLEAR; 6371 TRACET(3, track, "done"); 6372 return 0; 6373 } 6374 6375 /* 6376 * Send signal to process. 6377 * This is intended to be called only from audio_softintr_{rd,wr}. 6378 * Must be called without sc_intr_lock held. 6379 */ 6380 static inline void 6381 audio_psignal(struct audio_softc *sc, pid_t pid, int signum) 6382 { 6383 proc_t *p; 6384 6385 KASSERT(pid != 0); 6386 6387 /* 6388 * psignal() must be called without spin lock held. 6389 */ 6390 6391 mutex_enter(&proc_lock); 6392 p = proc_find(pid); 6393 if (p) 6394 psignal(p, signum); 6395 mutex_exit(&proc_lock); 6396 } 6397 6398 /* 6399 * This is software interrupt handler for record. 6400 * It is called from recording hardware interrupt everytime. 6401 * It does: 6402 * - Deliver SIGIO for all async processes. 6403 * - Notify to audio_read() that data has arrived. 6404 * - selnotify() for select/poll-ing processes. 6405 */ 6406 /* 6407 * XXX If a process issues FIOASYNC between hardware interrupt and 6408 * software interrupt, (stray) SIGIO will be sent to the process 6409 * despite the fact that it has not receive recorded data yet. 6410 */ 6411 static void 6412 audio_softintr_rd(void *cookie) 6413 { 6414 struct audio_softc *sc = cookie; 6415 audio_file_t *f; 6416 pid_t pid; 6417 6418 mutex_enter(sc->sc_lock); 6419 6420 SLIST_FOREACH(f, &sc->sc_files, entry) { 6421 audio_track_t *track = f->rtrack; 6422 6423 if (track == NULL) 6424 continue; 6425 6426 TRACET(4, track, "broadcast; inp=%d/%d/%d", 6427 track->input->head, 6428 track->input->used, 6429 track->input->capacity); 6430 6431 pid = f->async_audio; 6432 if (pid != 0) { 6433 TRACEF(4, f, "sending SIGIO %d", pid); 6434 audio_psignal(sc, pid, SIGIO); 6435 } 6436 } 6437 6438 /* Notify that data has arrived. */ 6439 selnotify(&sc->sc_rsel, 0, NOTE_SUBMIT); 6440 cv_broadcast(&sc->sc_rmixer->outcv); 6441 6442 mutex_exit(sc->sc_lock); 6443 } 6444 6445 /* 6446 * This is software interrupt handler for playback. 6447 * It is called from playback hardware interrupt everytime. 6448 * It does: 6449 * - Deliver SIGIO for all async and writable (used < lowat) processes. 6450 * - Notify to audio_write() that outbuf block available. 6451 * - selnotify() for select/poll-ing processes if there are any writable 6452 * (used < lowat) processes. Checking each descriptor will be done by 6453 * filt_audiowrite_event(). 6454 */ 6455 static void 6456 audio_softintr_wr(void *cookie) 6457 { 6458 struct audio_softc *sc = cookie; 6459 audio_file_t *f; 6460 bool found; 6461 pid_t pid; 6462 6463 TRACE(4, "called"); 6464 found = false; 6465 6466 mutex_enter(sc->sc_lock); 6467 6468 SLIST_FOREACH(f, &sc->sc_files, entry) { 6469 audio_track_t *track = f->ptrack; 6470 6471 if (track == NULL) 6472 continue; 6473 6474 TRACET(4, track, "broadcast; trkseq=%d out=%d/%d/%d", 6475 (int)track->seq, 6476 track->outbuf.head, 6477 track->outbuf.used, 6478 track->outbuf.capacity); 6479 6480 /* 6481 * Send a signal if the process is async mode and 6482 * used is lower than lowat. 6483 */ 6484 if (track->usrbuf.used <= track->usrbuf_usedlow && 6485 !track->is_pause) { 6486 /* For selnotify */ 6487 found = true; 6488 /* For SIGIO */ 6489 pid = f->async_audio; 6490 if (pid != 0) { 6491 TRACEF(4, f, "sending SIGIO %d", pid); 6492 audio_psignal(sc, pid, SIGIO); 6493 } 6494 } 6495 } 6496 6497 /* 6498 * Notify for select/poll when someone become writable. 6499 * It needs sc_lock (and not sc_intr_lock). 6500 */ 6501 if (found) { 6502 TRACE(4, "selnotify"); 6503 selnotify(&sc->sc_wsel, 0, NOTE_SUBMIT); 6504 } 6505 6506 /* Notify to audio_write() that outbuf available. */ 6507 cv_broadcast(&sc->sc_pmixer->outcv); 6508 6509 mutex_exit(sc->sc_lock); 6510 } 6511 6512 /* 6513 * Check (and convert) the format *p came from userland. 6514 * If successful, it writes back the converted format to *p if necessary and 6515 * returns 0. Otherwise returns errno (*p may be changed even in this case). 6516 */ 6517 static int 6518 audio_check_params(audio_format2_t *p) 6519 { 6520 6521 /* 6522 * Convert obsolete AUDIO_ENCODING_PCM encodings. 6523 * 6524 * AUDIO_ENCODING_PCM16 == AUDIO_ENCODING_LINEAR 6525 * So, it's always signed, as in SunOS. 6526 * 6527 * AUDIO_ENCODING_PCM8 == AUDIO_ENCODING_LINEAR8 6528 * So, it's always unsigned, as in SunOS. 6529 */ 6530 if (p->encoding == AUDIO_ENCODING_PCM16) { 6531 p->encoding = AUDIO_ENCODING_SLINEAR; 6532 } else if (p->encoding == AUDIO_ENCODING_PCM8) { 6533 if (p->precision == 8) 6534 p->encoding = AUDIO_ENCODING_ULINEAR; 6535 else 6536 return EINVAL; 6537 } 6538 6539 /* 6540 * Convert obsoleted AUDIO_ENCODING_[SU]LINEAR without endianness 6541 * suffix. 6542 */ 6543 if (p->encoding == AUDIO_ENCODING_SLINEAR) 6544 p->encoding = AUDIO_ENCODING_SLINEAR_NE; 6545 if (p->encoding == AUDIO_ENCODING_ULINEAR) 6546 p->encoding = AUDIO_ENCODING_ULINEAR_NE; 6547 6548 switch (p->encoding) { 6549 case AUDIO_ENCODING_ULAW: 6550 case AUDIO_ENCODING_ALAW: 6551 if (p->precision != 8) 6552 return EINVAL; 6553 break; 6554 case AUDIO_ENCODING_ADPCM: 6555 if (p->precision != 4 && p->precision != 8) 6556 return EINVAL; 6557 break; 6558 case AUDIO_ENCODING_SLINEAR_LE: 6559 case AUDIO_ENCODING_SLINEAR_BE: 6560 case AUDIO_ENCODING_ULINEAR_LE: 6561 case AUDIO_ENCODING_ULINEAR_BE: 6562 if (p->precision != 8 && p->precision != 16 && 6563 p->precision != 24 && p->precision != 32) 6564 return EINVAL; 6565 6566 /* 8bit format does not have endianness. */ 6567 if (p->precision == 8) { 6568 if (p->encoding == AUDIO_ENCODING_SLINEAR_OE) 6569 p->encoding = AUDIO_ENCODING_SLINEAR_NE; 6570 if (p->encoding == AUDIO_ENCODING_ULINEAR_OE) 6571 p->encoding = AUDIO_ENCODING_ULINEAR_NE; 6572 } 6573 6574 if (p->precision > p->stride) 6575 return EINVAL; 6576 break; 6577 case AUDIO_ENCODING_MPEG_L1_STREAM: 6578 case AUDIO_ENCODING_MPEG_L1_PACKETS: 6579 case AUDIO_ENCODING_MPEG_L1_SYSTEM: 6580 case AUDIO_ENCODING_MPEG_L2_STREAM: 6581 case AUDIO_ENCODING_MPEG_L2_PACKETS: 6582 case AUDIO_ENCODING_MPEG_L2_SYSTEM: 6583 case AUDIO_ENCODING_AC3: 6584 break; 6585 default: 6586 return EINVAL; 6587 } 6588 6589 /* sanity check # of channels*/ 6590 if (p->channels < 1 || p->channels > AUDIO_MAX_CHANNELS) 6591 return EINVAL; 6592 6593 return 0; 6594 } 6595 6596 /* 6597 * Initialize playback and record mixers. 6598 * mode (AUMODE_{PLAY,RECORD}) indicates the mixer to be initialized. 6599 * phwfmt and rhwfmt indicate the hardware format. pfil and rfil indicate 6600 * the filter registration information. These four must not be NULL. 6601 * If successful returns 0. Otherwise returns errno. 6602 * Must be called with sc_exlock held and without sc_lock held. 6603 * Must not be called if there are any tracks. 6604 * Caller should check that the initialization succeed by whether 6605 * sc_[pr]mixer is not NULL. 6606 */ 6607 static int 6608 audio_mixers_init(struct audio_softc *sc, int mode, 6609 const audio_format2_t *phwfmt, const audio_format2_t *rhwfmt, 6610 const audio_filter_reg_t *pfil, const audio_filter_reg_t *rfil) 6611 { 6612 int error; 6613 6614 KASSERT(phwfmt != NULL); 6615 KASSERT(rhwfmt != NULL); 6616 KASSERT(pfil != NULL); 6617 KASSERT(rfil != NULL); 6618 KASSERT(sc->sc_exlock); 6619 6620 if ((mode & AUMODE_PLAY)) { 6621 if (sc->sc_pmixer == NULL) { 6622 sc->sc_pmixer = kmem_zalloc(sizeof(*sc->sc_pmixer), 6623 KM_SLEEP); 6624 } else { 6625 /* destroy() doesn't free memory. */ 6626 audio_mixer_destroy(sc, sc->sc_pmixer); 6627 memset(sc->sc_pmixer, 0, sizeof(*sc->sc_pmixer)); 6628 } 6629 error = audio_mixer_init(sc, AUMODE_PLAY, phwfmt, pfil); 6630 if (error) { 6631 /* audio_mixer_init already displayed error code */ 6632 audio_printf(sc, "configuring playback mode failed\n"); 6633 kmem_free(sc->sc_pmixer, sizeof(*sc->sc_pmixer)); 6634 sc->sc_pmixer = NULL; 6635 return error; 6636 } 6637 } 6638 if ((mode & AUMODE_RECORD)) { 6639 if (sc->sc_rmixer == NULL) { 6640 sc->sc_rmixer = kmem_zalloc(sizeof(*sc->sc_rmixer), 6641 KM_SLEEP); 6642 } else { 6643 /* destroy() doesn't free memory. */ 6644 audio_mixer_destroy(sc, sc->sc_rmixer); 6645 memset(sc->sc_rmixer, 0, sizeof(*sc->sc_rmixer)); 6646 } 6647 error = audio_mixer_init(sc, AUMODE_RECORD, rhwfmt, rfil); 6648 if (error) { 6649 /* audio_mixer_init already displayed error code */ 6650 audio_printf(sc, "configuring record mode failed\n"); 6651 kmem_free(sc->sc_rmixer, sizeof(*sc->sc_rmixer)); 6652 sc->sc_rmixer = NULL; 6653 return error; 6654 } 6655 } 6656 6657 return 0; 6658 } 6659 6660 /* 6661 * Select a frequency. 6662 * Prioritize 48kHz and 44.1kHz. Otherwise choose the highest one. 6663 * XXX Better algorithm? 6664 */ 6665 static int 6666 audio_select_freq(const struct audio_format *fmt) 6667 { 6668 int freq; 6669 int high; 6670 int low; 6671 int j; 6672 6673 if (fmt->frequency_type == 0) { 6674 low = fmt->frequency[0]; 6675 high = fmt->frequency[1]; 6676 freq = 48000; 6677 if (low <= freq && freq <= high) { 6678 return freq; 6679 } 6680 freq = 44100; 6681 if (low <= freq && freq <= high) { 6682 return freq; 6683 } 6684 return high; 6685 } else { 6686 for (j = 0; j < fmt->frequency_type; j++) { 6687 if (fmt->frequency[j] == 48000) { 6688 return fmt->frequency[j]; 6689 } 6690 } 6691 high = 0; 6692 for (j = 0; j < fmt->frequency_type; j++) { 6693 if (fmt->frequency[j] == 44100) { 6694 return fmt->frequency[j]; 6695 } 6696 if (fmt->frequency[j] > high) { 6697 high = fmt->frequency[j]; 6698 } 6699 } 6700 return high; 6701 } 6702 } 6703 6704 /* 6705 * Choose the most preferred hardware format. 6706 * If successful, it will store the chosen format into *cand and return 0. 6707 * Otherwise, return errno. 6708 * Must be called without sc_lock held. 6709 */ 6710 static int 6711 audio_hw_probe(struct audio_softc *sc, audio_format2_t *cand, int mode) 6712 { 6713 audio_format_query_t query; 6714 int cand_score; 6715 int score; 6716 int i; 6717 int error; 6718 6719 /* 6720 * Score each formats and choose the highest one. 6721 * 6722 * +---- priority(0-3) 6723 * |+--- encoding/precision 6724 * ||+-- channels 6725 * score = 0x000000PEC 6726 */ 6727 6728 cand_score = 0; 6729 for (i = 0; ; i++) { 6730 memset(&query, 0, sizeof(query)); 6731 query.index = i; 6732 6733 mutex_enter(sc->sc_lock); 6734 error = sc->hw_if->query_format(sc->hw_hdl, &query); 6735 mutex_exit(sc->sc_lock); 6736 if (error == EINVAL) 6737 break; 6738 if (error) 6739 return error; 6740 6741 #if defined(AUDIO_DEBUG) 6742 DPRINTF(1, "fmt[%d] %c%c pri=%d %s,%d/%dbit,%dch,", i, 6743 (query.fmt.mode & AUMODE_PLAY) ? 'P' : '-', 6744 (query.fmt.mode & AUMODE_RECORD) ? 'R' : '-', 6745 query.fmt.priority, 6746 audio_encoding_name(query.fmt.encoding), 6747 query.fmt.validbits, 6748 query.fmt.precision, 6749 query.fmt.channels); 6750 if (query.fmt.frequency_type == 0) { 6751 DPRINTF(1, "{%d-%d", 6752 query.fmt.frequency[0], query.fmt.frequency[1]); 6753 } else { 6754 int j; 6755 for (j = 0; j < query.fmt.frequency_type; j++) { 6756 DPRINTF(1, "%c%d", 6757 (j == 0) ? '{' : ',', 6758 query.fmt.frequency[j]); 6759 } 6760 } 6761 DPRINTF(1, "}\n"); 6762 #endif 6763 6764 if ((query.fmt.mode & mode) == 0) { 6765 DPRINTF(1, "fmt[%d] skip; mode not match %d\n", i, 6766 mode); 6767 continue; 6768 } 6769 6770 if (query.fmt.priority < 0) { 6771 DPRINTF(1, "fmt[%d] skip; unsupported encoding\n", i); 6772 continue; 6773 } 6774 6775 /* Score */ 6776 score = (query.fmt.priority & 3) * 0x100; 6777 if (query.fmt.encoding == AUDIO_ENCODING_SLINEAR_NE && 6778 query.fmt.validbits == AUDIO_INTERNAL_BITS && 6779 query.fmt.precision == AUDIO_INTERNAL_BITS) { 6780 score += 0x20; 6781 } else if (query.fmt.encoding == AUDIO_ENCODING_SLINEAR_OE && 6782 query.fmt.validbits == AUDIO_INTERNAL_BITS && 6783 query.fmt.precision == AUDIO_INTERNAL_BITS) { 6784 score += 0x10; 6785 } 6786 6787 /* Do not prefer surround formats */ 6788 if (query.fmt.channels <= 2) 6789 score += query.fmt.channels; 6790 6791 if (score < cand_score) { 6792 DPRINTF(1, "fmt[%d] skip; score 0x%x < 0x%x\n", i, 6793 score, cand_score); 6794 continue; 6795 } 6796 6797 /* Update candidate */ 6798 cand_score = score; 6799 cand->encoding = query.fmt.encoding; 6800 cand->precision = query.fmt.validbits; 6801 cand->stride = query.fmt.precision; 6802 cand->channels = query.fmt.channels; 6803 cand->sample_rate = audio_select_freq(&query.fmt); 6804 DPRINTF(1, "fmt[%d] candidate (score=0x%x)" 6805 " pri=%d %s,%d/%d,%dch,%dHz\n", i, 6806 cand_score, query.fmt.priority, 6807 audio_encoding_name(query.fmt.encoding), 6808 cand->precision, cand->stride, 6809 cand->channels, cand->sample_rate); 6810 } 6811 6812 if (cand_score == 0) { 6813 DPRINTF(1, "%s no fmt\n", __func__); 6814 return ENXIO; 6815 } 6816 DPRINTF(1, "%s selected: %s,%d/%d,%dch,%dHz\n", __func__, 6817 audio_encoding_name(cand->encoding), 6818 cand->precision, cand->stride, cand->channels, cand->sample_rate); 6819 return 0; 6820 } 6821 6822 /* 6823 * Validate fmt with query_format. 6824 * If fmt is included in the result of query_format, returns 0. 6825 * Otherwise returns EINVAL. 6826 * Must be called without sc_lock held. 6827 */ 6828 static int 6829 audio_hw_validate_format(struct audio_softc *sc, int mode, 6830 const audio_format2_t *fmt) 6831 { 6832 audio_format_query_t query; 6833 struct audio_format *q; 6834 int index; 6835 int error; 6836 int j; 6837 6838 for (index = 0; ; index++) { 6839 query.index = index; 6840 mutex_enter(sc->sc_lock); 6841 error = sc->hw_if->query_format(sc->hw_hdl, &query); 6842 mutex_exit(sc->sc_lock); 6843 if (error == EINVAL) 6844 break; 6845 if (error) 6846 return error; 6847 6848 q = &query.fmt; 6849 /* 6850 * Note that fmt is audio_format2_t (precision/stride) but 6851 * q is audio_format_t (validbits/precision). 6852 */ 6853 if ((q->mode & mode) == 0) { 6854 continue; 6855 } 6856 if (fmt->encoding != q->encoding) { 6857 continue; 6858 } 6859 if (fmt->precision != q->validbits) { 6860 continue; 6861 } 6862 if (fmt->stride != q->precision) { 6863 continue; 6864 } 6865 if (fmt->channels != q->channels) { 6866 continue; 6867 } 6868 if (q->frequency_type == 0) { 6869 if (fmt->sample_rate < q->frequency[0] || 6870 fmt->sample_rate > q->frequency[1]) { 6871 continue; 6872 } 6873 } else { 6874 for (j = 0; j < q->frequency_type; j++) { 6875 if (fmt->sample_rate == q->frequency[j]) 6876 break; 6877 } 6878 if (j == query.fmt.frequency_type) { 6879 continue; 6880 } 6881 } 6882 6883 /* Matched. */ 6884 return 0; 6885 } 6886 6887 return EINVAL; 6888 } 6889 6890 /* 6891 * Set track mixer's format depending on ai->mode. 6892 * If AUMODE_PLAY is set in ai->mode, it set up the playback mixer 6893 * with ai.play.*. 6894 * If AUMODE_RECORD is set in ai->mode, it set up the recording mixer 6895 * with ai.record.*. 6896 * All other fields in ai are ignored. 6897 * If successful returns 0. Otherwise returns errno. 6898 * This function does not roll back even if it fails. 6899 * Must be called with sc_exlock held and without sc_lock held. 6900 */ 6901 static int 6902 audio_mixers_set_format(struct audio_softc *sc, const struct audio_info *ai) 6903 { 6904 audio_format2_t phwfmt; 6905 audio_format2_t rhwfmt; 6906 audio_filter_reg_t pfil; 6907 audio_filter_reg_t rfil; 6908 int mode; 6909 int error; 6910 6911 KASSERT(sc->sc_exlock); 6912 6913 /* 6914 * Even when setting either one of playback and recording, 6915 * both must be halted. 6916 */ 6917 if (sc->sc_popens + sc->sc_ropens > 0) 6918 return EBUSY; 6919 6920 if (!SPECIFIED(ai->mode) || ai->mode == 0) 6921 return ENOTTY; 6922 6923 mode = ai->mode; 6924 if ((mode & AUMODE_PLAY)) { 6925 phwfmt.encoding = ai->play.encoding; 6926 phwfmt.precision = ai->play.precision; 6927 phwfmt.stride = ai->play.precision; 6928 phwfmt.channels = ai->play.channels; 6929 phwfmt.sample_rate = ai->play.sample_rate; 6930 } 6931 if ((mode & AUMODE_RECORD)) { 6932 rhwfmt.encoding = ai->record.encoding; 6933 rhwfmt.precision = ai->record.precision; 6934 rhwfmt.stride = ai->record.precision; 6935 rhwfmt.channels = ai->record.channels; 6936 rhwfmt.sample_rate = ai->record.sample_rate; 6937 } 6938 6939 /* On non-independent devices, use the same format for both. */ 6940 if ((sc->sc_props & AUDIO_PROP_INDEPENDENT) == 0) { 6941 if (mode == AUMODE_RECORD) { 6942 phwfmt = rhwfmt; 6943 } else { 6944 rhwfmt = phwfmt; 6945 } 6946 mode = AUMODE_PLAY | AUMODE_RECORD; 6947 } 6948 6949 /* Then, unset the direction not exist on the hardware. */ 6950 if ((sc->sc_props & AUDIO_PROP_PLAYBACK) == 0) 6951 mode &= ~AUMODE_PLAY; 6952 if ((sc->sc_props & AUDIO_PROP_CAPTURE) == 0) 6953 mode &= ~AUMODE_RECORD; 6954 6955 /* debug */ 6956 if ((mode & AUMODE_PLAY)) { 6957 TRACE(1, "play=%s/%d/%d/%dch/%dHz", 6958 audio_encoding_name(phwfmt.encoding), 6959 phwfmt.precision, 6960 phwfmt.stride, 6961 phwfmt.channels, 6962 phwfmt.sample_rate); 6963 } 6964 if ((mode & AUMODE_RECORD)) { 6965 TRACE(1, "rec =%s/%d/%d/%dch/%dHz", 6966 audio_encoding_name(rhwfmt.encoding), 6967 rhwfmt.precision, 6968 rhwfmt.stride, 6969 rhwfmt.channels, 6970 rhwfmt.sample_rate); 6971 } 6972 6973 /* Check the format */ 6974 if ((mode & AUMODE_PLAY)) { 6975 if (audio_hw_validate_format(sc, AUMODE_PLAY, &phwfmt)) { 6976 TRACE(1, "invalid format"); 6977 return EINVAL; 6978 } 6979 } 6980 if ((mode & AUMODE_RECORD)) { 6981 if (audio_hw_validate_format(sc, AUMODE_RECORD, &rhwfmt)) { 6982 TRACE(1, "invalid format"); 6983 return EINVAL; 6984 } 6985 } 6986 6987 /* Configure the mixers. */ 6988 memset(&pfil, 0, sizeof(pfil)); 6989 memset(&rfil, 0, sizeof(rfil)); 6990 error = audio_hw_set_format(sc, mode, &phwfmt, &rhwfmt, &pfil, &rfil); 6991 if (error) 6992 return error; 6993 6994 error = audio_mixers_init(sc, mode, &phwfmt, &rhwfmt, &pfil, &rfil); 6995 if (error) 6996 return error; 6997 6998 /* 6999 * Reinitialize the sticky parameters for /dev/sound. 7000 * If the number of the hardware channels becomes less than the number 7001 * of channels that sticky parameters remember, subsequent /dev/sound 7002 * open will fail. To prevent this, reinitialize the sticky 7003 * parameters whenever the hardware format is changed. 7004 */ 7005 sc->sc_sound_pparams = params_to_format2(&audio_default); 7006 sc->sc_sound_rparams = params_to_format2(&audio_default); 7007 sc->sc_sound_ppause = false; 7008 sc->sc_sound_rpause = false; 7009 7010 return 0; 7011 } 7012 7013 /* 7014 * Store current mixers format into *ai. 7015 * Must be called with sc_exlock held. 7016 */ 7017 static void 7018 audio_mixers_get_format(struct audio_softc *sc, struct audio_info *ai) 7019 { 7020 7021 KASSERT(sc->sc_exlock); 7022 7023 /* 7024 * There is no stride information in audio_info but it doesn't matter. 7025 * trackmixer always treats stride and precision as the same. 7026 */ 7027 AUDIO_INITINFO(ai); 7028 ai->mode = 0; 7029 if (sc->sc_pmixer) { 7030 audio_format2_t *fmt = &sc->sc_pmixer->track_fmt; 7031 ai->play.encoding = fmt->encoding; 7032 ai->play.precision = fmt->precision; 7033 ai->play.channels = fmt->channels; 7034 ai->play.sample_rate = fmt->sample_rate; 7035 ai->mode |= AUMODE_PLAY; 7036 } 7037 if (sc->sc_rmixer) { 7038 audio_format2_t *fmt = &sc->sc_rmixer->track_fmt; 7039 ai->record.encoding = fmt->encoding; 7040 ai->record.precision = fmt->precision; 7041 ai->record.channels = fmt->channels; 7042 ai->record.sample_rate = fmt->sample_rate; 7043 ai->mode |= AUMODE_RECORD; 7044 } 7045 } 7046 7047 /* 7048 * audio_info details: 7049 * 7050 * ai.{play,record}.sample_rate (R/W) 7051 * ai.{play,record}.encoding (R/W) 7052 * ai.{play,record}.precision (R/W) 7053 * ai.{play,record}.channels (R/W) 7054 * These specify the playback or recording format. 7055 * Ignore members within an inactive track. 7056 * 7057 * ai.mode (R/W) 7058 * It specifies the playback or recording mode, AUMODE_*. 7059 * Currently, a mode change operation by ai.mode after opening is 7060 * prohibited. In addition, AUMODE_PLAY_ALL no longer makes sense. 7061 * However, it's possible to get or to set for backward compatibility. 7062 * 7063 * ai.{hiwat,lowat} (R/W) 7064 * These specify the high water mark and low water mark for playback 7065 * track. The unit is block. 7066 * 7067 * ai.{play,record}.gain (R/W) 7068 * It specifies the HW mixer volume in 0-255. 7069 * It is historical reason that the gain is connected to HW mixer. 7070 * 7071 * ai.{play,record}.balance (R/W) 7072 * It specifies the left-right balance of HW mixer in 0-64. 7073 * 32 means the center. 7074 * It is historical reason that the balance is connected to HW mixer. 7075 * 7076 * ai.{play,record}.port (R/W) 7077 * It specifies the input/output port of HW mixer. 7078 * 7079 * ai.monitor_gain (R/W) 7080 * It specifies the recording monitor gain(?) of HW mixer. 7081 * 7082 * ai.{play,record}.pause (R/W) 7083 * Non-zero means the track is paused. 7084 * 7085 * ai.play.seek (R/-) 7086 * It indicates the number of bytes written but not processed. 7087 * ai.record.seek (R/-) 7088 * It indicates the number of bytes to be able to read. 7089 * 7090 * ai.{play,record}.avail_ports (R/-) 7091 * Mixer info. 7092 * 7093 * ai.{play,record}.buffer_size (R/-) 7094 * It indicates the buffer size in bytes. Internally it means usrbuf. 7095 * 7096 * ai.{play,record}.samples (R/-) 7097 * It indicates the total number of bytes played or recorded. 7098 * 7099 * ai.{play,record}.eof (R/-) 7100 * It indicates the number of times reached EOF(?). 7101 * 7102 * ai.{play,record}.error (R/-) 7103 * Non-zero indicates overflow/underflow has occurred. 7104 * 7105 * ai.{play,record}.waiting (R/-) 7106 * Non-zero indicates that other process waits to open. 7107 * It will never happen anymore. 7108 * 7109 * ai.{play,record}.open (R/-) 7110 * Non-zero indicates the direction is opened by this process(?). 7111 * XXX Is this better to indicate that "the device is opened by 7112 * at least one process"? 7113 * 7114 * ai.{play,record}.active (R/-) 7115 * Non-zero indicates that I/O is currently active. 7116 * 7117 * ai.blocksize (R/-) 7118 * It indicates the block size in bytes. 7119 * XXX The blocksize of playback and recording may be different. 7120 */ 7121 7122 /* 7123 * Pause consideration: 7124 * 7125 * Pausing/unpausing never affect [pr]mixer. This single rule makes 7126 * operation simple. Note that playback and recording are asymmetric. 7127 * 7128 * For playback, 7129 * 1. Any playback open doesn't start pmixer regardless of initial pause 7130 * state of this track. 7131 * 2. The first write access among playback tracks only starts pmixer 7132 * regardless of this track's pause state. 7133 * 3. Even a pause of the last playback track doesn't stop pmixer. 7134 * 4. The last close of all playback tracks only stops pmixer. 7135 * 7136 * For recording, 7137 * 1. The first recording open only starts rmixer regardless of initial 7138 * pause state of this track. 7139 * 2. Even a pause of the last track doesn't stop rmixer. 7140 * 3. The last close of all recording tracks only stops rmixer. 7141 */ 7142 7143 /* 7144 * Set both track's parameters within a file depending on ai. 7145 * Update sc_sound_[pr]* if set. 7146 * Must be called with sc_exlock held and without sc_lock held. 7147 */ 7148 static int 7149 audio_file_setinfo(struct audio_softc *sc, audio_file_t *file, 7150 const struct audio_info *ai) 7151 { 7152 const struct audio_prinfo *pi; 7153 const struct audio_prinfo *ri; 7154 audio_track_t *ptrack; 7155 audio_track_t *rtrack; 7156 audio_format2_t pfmt; 7157 audio_format2_t rfmt; 7158 int pchanges; 7159 int rchanges; 7160 int mode; 7161 struct audio_info saved_ai; 7162 audio_format2_t saved_pfmt; 7163 audio_format2_t saved_rfmt; 7164 int error; 7165 7166 KASSERT(sc->sc_exlock); 7167 7168 pi = &ai->play; 7169 ri = &ai->record; 7170 pchanges = 0; 7171 rchanges = 0; 7172 7173 ptrack = file->ptrack; 7174 rtrack = file->rtrack; 7175 7176 #if defined(AUDIO_DEBUG) 7177 if (audiodebug >= 2) { 7178 char buf[256]; 7179 char p[64]; 7180 int buflen; 7181 int plen; 7182 #define SPRINTF(var, fmt...) do { \ 7183 var##len += snprintf(var + var##len, sizeof(var) - var##len, fmt); \ 7184 } while (0) 7185 7186 buflen = 0; 7187 plen = 0; 7188 if (SPECIFIED(pi->encoding)) 7189 SPRINTF(p, "/%s", audio_encoding_name(pi->encoding)); 7190 if (SPECIFIED(pi->precision)) 7191 SPRINTF(p, "/%dbit", pi->precision); 7192 if (SPECIFIED(pi->channels)) 7193 SPRINTF(p, "/%dch", pi->channels); 7194 if (SPECIFIED(pi->sample_rate)) 7195 SPRINTF(p, "/%dHz", pi->sample_rate); 7196 if (plen > 0) 7197 SPRINTF(buf, ",play.param=%s", p + 1); 7198 7199 plen = 0; 7200 if (SPECIFIED(ri->encoding)) 7201 SPRINTF(p, "/%s", audio_encoding_name(ri->encoding)); 7202 if (SPECIFIED(ri->precision)) 7203 SPRINTF(p, "/%dbit", ri->precision); 7204 if (SPECIFIED(ri->channels)) 7205 SPRINTF(p, "/%dch", ri->channels); 7206 if (SPECIFIED(ri->sample_rate)) 7207 SPRINTF(p, "/%dHz", ri->sample_rate); 7208 if (plen > 0) 7209 SPRINTF(buf, ",record.param=%s", p + 1); 7210 7211 if (SPECIFIED(ai->mode)) 7212 SPRINTF(buf, ",mode=%d", ai->mode); 7213 if (SPECIFIED(ai->hiwat)) 7214 SPRINTF(buf, ",hiwat=%d", ai->hiwat); 7215 if (SPECIFIED(ai->lowat)) 7216 SPRINTF(buf, ",lowat=%d", ai->lowat); 7217 if (SPECIFIED(ai->play.gain)) 7218 SPRINTF(buf, ",play.gain=%d", ai->play.gain); 7219 if (SPECIFIED(ai->record.gain)) 7220 SPRINTF(buf, ",record.gain=%d", ai->record.gain); 7221 if (SPECIFIED_CH(ai->play.balance)) 7222 SPRINTF(buf, ",play.balance=%d", ai->play.balance); 7223 if (SPECIFIED_CH(ai->record.balance)) 7224 SPRINTF(buf, ",record.balance=%d", ai->record.balance); 7225 if (SPECIFIED(ai->play.port)) 7226 SPRINTF(buf, ",play.port=%d", ai->play.port); 7227 if (SPECIFIED(ai->record.port)) 7228 SPRINTF(buf, ",record.port=%d", ai->record.port); 7229 if (SPECIFIED(ai->monitor_gain)) 7230 SPRINTF(buf, ",monitor_gain=%d", ai->monitor_gain); 7231 if (SPECIFIED_CH(ai->play.pause)) 7232 SPRINTF(buf, ",play.pause=%d", ai->play.pause); 7233 if (SPECIFIED_CH(ai->record.pause)) 7234 SPRINTF(buf, ",record.pause=%d", ai->record.pause); 7235 7236 if (buflen > 0) 7237 TRACE(2, "specified %s", buf + 1); 7238 } 7239 #endif 7240 7241 AUDIO_INITINFO(&saved_ai); 7242 /* XXX shut up gcc */ 7243 memset(&saved_pfmt, 0, sizeof(saved_pfmt)); 7244 memset(&saved_rfmt, 0, sizeof(saved_rfmt)); 7245 7246 /* 7247 * Set default value and save current parameters. 7248 * For backward compatibility, use sticky parameters for nonexistent 7249 * track. 7250 */ 7251 if (ptrack) { 7252 pfmt = ptrack->usrbuf.fmt; 7253 saved_pfmt = ptrack->usrbuf.fmt; 7254 saved_ai.play.pause = ptrack->is_pause; 7255 } else { 7256 pfmt = sc->sc_sound_pparams; 7257 } 7258 if (rtrack) { 7259 rfmt = rtrack->usrbuf.fmt; 7260 saved_rfmt = rtrack->usrbuf.fmt; 7261 saved_ai.record.pause = rtrack->is_pause; 7262 } else { 7263 rfmt = sc->sc_sound_rparams; 7264 } 7265 saved_ai.mode = file->mode; 7266 7267 /* 7268 * Overwrite if specified. 7269 */ 7270 mode = file->mode; 7271 if (SPECIFIED(ai->mode)) { 7272 /* 7273 * Setting ai->mode no longer does anything because it's 7274 * prohibited to change playback/recording mode after open 7275 * and AUMODE_PLAY_ALL is obsoleted. However, it still 7276 * keeps the state of AUMODE_PLAY_ALL itself for backward 7277 * compatibility. 7278 * In the internal, only file->mode has the state of 7279 * AUMODE_PLAY_ALL flag and track->mode in both track does 7280 * not have. 7281 */ 7282 if ((file->mode & AUMODE_PLAY)) { 7283 mode = (file->mode & (AUMODE_PLAY | AUMODE_RECORD)) 7284 | (ai->mode & AUMODE_PLAY_ALL); 7285 } 7286 } 7287 7288 pchanges = audio_track_setinfo_check(ptrack, &pfmt, pi); 7289 if (pchanges == -1) { 7290 #if defined(AUDIO_DEBUG) 7291 TRACEF(1, file, "check play.params failed: " 7292 "%s %ubit %uch %uHz", 7293 audio_encoding_name(pi->encoding), 7294 pi->precision, 7295 pi->channels, 7296 pi->sample_rate); 7297 #endif 7298 return EINVAL; 7299 } 7300 7301 rchanges = audio_track_setinfo_check(rtrack, &rfmt, ri); 7302 if (rchanges == -1) { 7303 #if defined(AUDIO_DEBUG) 7304 TRACEF(1, file, "check record.params failed: " 7305 "%s %ubit %uch %uHz", 7306 audio_encoding_name(ri->encoding), 7307 ri->precision, 7308 ri->channels, 7309 ri->sample_rate); 7310 #endif 7311 return EINVAL; 7312 } 7313 7314 if (SPECIFIED(ai->mode)) { 7315 pchanges = 1; 7316 rchanges = 1; 7317 } 7318 7319 /* 7320 * Even when setting either one of playback and recording, 7321 * both track must be halted. 7322 */ 7323 if (pchanges || rchanges) { 7324 audio_file_clear(sc, file); 7325 #if defined(AUDIO_DEBUG) 7326 char nbuf[16]; 7327 char fmtbuf[64]; 7328 if (pchanges) { 7329 if (ptrack) { 7330 snprintf(nbuf, sizeof(nbuf), "%d", ptrack->id); 7331 } else { 7332 snprintf(nbuf, sizeof(nbuf), "-"); 7333 } 7334 audio_format2_tostr(fmtbuf, sizeof(fmtbuf), &pfmt); 7335 DPRINTF(1, "audio track#%s play mode: %s\n", 7336 nbuf, fmtbuf); 7337 } 7338 if (rchanges) { 7339 if (rtrack) { 7340 snprintf(nbuf, sizeof(nbuf), "%d", rtrack->id); 7341 } else { 7342 snprintf(nbuf, sizeof(nbuf), "-"); 7343 } 7344 audio_format2_tostr(fmtbuf, sizeof(fmtbuf), &rfmt); 7345 DPRINTF(1, "audio track#%s rec mode: %s\n", 7346 nbuf, fmtbuf); 7347 } 7348 #endif 7349 } 7350 7351 /* Set mixer parameters */ 7352 mutex_enter(sc->sc_lock); 7353 error = audio_hw_setinfo(sc, ai, &saved_ai); 7354 mutex_exit(sc->sc_lock); 7355 if (error) 7356 goto abort1; 7357 7358 /* 7359 * Set to track and update sticky parameters. 7360 */ 7361 error = 0; 7362 file->mode = mode; 7363 7364 if (SPECIFIED_CH(pi->pause)) { 7365 if (ptrack) 7366 ptrack->is_pause = pi->pause; 7367 sc->sc_sound_ppause = pi->pause; 7368 } 7369 if (pchanges) { 7370 if (ptrack) { 7371 audio_track_lock_enter(ptrack); 7372 error = audio_track_set_format(ptrack, &pfmt); 7373 audio_track_lock_exit(ptrack); 7374 if (error) { 7375 TRACET(1, ptrack, "set play.params failed"); 7376 goto abort2; 7377 } 7378 } 7379 sc->sc_sound_pparams = pfmt; 7380 } 7381 /* Change water marks after initializing the buffers. */ 7382 if (SPECIFIED(ai->hiwat) || SPECIFIED(ai->lowat)) { 7383 if (ptrack) 7384 audio_track_setinfo_water(ptrack, ai); 7385 } 7386 7387 if (SPECIFIED_CH(ri->pause)) { 7388 if (rtrack) 7389 rtrack->is_pause = ri->pause; 7390 sc->sc_sound_rpause = ri->pause; 7391 } 7392 if (rchanges) { 7393 if (rtrack) { 7394 audio_track_lock_enter(rtrack); 7395 error = audio_track_set_format(rtrack, &rfmt); 7396 audio_track_lock_exit(rtrack); 7397 if (error) { 7398 TRACET(1, rtrack, "set record.params failed"); 7399 goto abort3; 7400 } 7401 } 7402 sc->sc_sound_rparams = rfmt; 7403 } 7404 7405 return 0; 7406 7407 /* Rollback */ 7408 abort3: 7409 if (error != ENOMEM) { 7410 rtrack->is_pause = saved_ai.record.pause; 7411 audio_track_lock_enter(rtrack); 7412 audio_track_set_format(rtrack, &saved_rfmt); 7413 audio_track_lock_exit(rtrack); 7414 } 7415 sc->sc_sound_rpause = saved_ai.record.pause; 7416 sc->sc_sound_rparams = saved_rfmt; 7417 abort2: 7418 if (ptrack && error != ENOMEM) { 7419 ptrack->is_pause = saved_ai.play.pause; 7420 audio_track_lock_enter(ptrack); 7421 audio_track_set_format(ptrack, &saved_pfmt); 7422 audio_track_lock_exit(ptrack); 7423 } 7424 sc->sc_sound_ppause = saved_ai.play.pause; 7425 sc->sc_sound_pparams = saved_pfmt; 7426 file->mode = saved_ai.mode; 7427 abort1: 7428 mutex_enter(sc->sc_lock); 7429 audio_hw_setinfo(sc, &saved_ai, NULL); 7430 mutex_exit(sc->sc_lock); 7431 7432 return error; 7433 } 7434 7435 /* 7436 * Write SPECIFIED() parameters within info back to fmt. 7437 * Note that track can be NULL here. 7438 * Return value of 1 indicates that fmt is modified. 7439 * Return value of 0 indicates that fmt is not modified. 7440 * Return value of -1 indicates that error EINVAL has occurred. 7441 */ 7442 static int 7443 audio_track_setinfo_check(audio_track_t *track, 7444 audio_format2_t *fmt, const struct audio_prinfo *info) 7445 { 7446 const audio_format2_t *hwfmt; 7447 int changes; 7448 7449 changes = 0; 7450 if (SPECIFIED(info->sample_rate)) { 7451 if (info->sample_rate < AUDIO_MIN_FREQUENCY) 7452 return -1; 7453 if (info->sample_rate > AUDIO_MAX_FREQUENCY) 7454 return -1; 7455 fmt->sample_rate = info->sample_rate; 7456 changes = 1; 7457 } 7458 if (SPECIFIED(info->encoding)) { 7459 fmt->encoding = info->encoding; 7460 changes = 1; 7461 } 7462 if (SPECIFIED(info->precision)) { 7463 fmt->precision = info->precision; 7464 /* we don't have API to specify stride */ 7465 fmt->stride = info->precision; 7466 changes = 1; 7467 } 7468 if (SPECIFIED(info->channels)) { 7469 /* 7470 * We can convert between monaural and stereo each other. 7471 * We can reduce than the number of channels that the hardware 7472 * supports. 7473 */ 7474 if (info->channels > 2) { 7475 if (track) { 7476 hwfmt = &track->mixer->hwbuf.fmt; 7477 if (info->channels > hwfmt->channels) 7478 return -1; 7479 } else { 7480 /* 7481 * This should never happen. 7482 * If track == NULL, channels should be <= 2. 7483 */ 7484 return -1; 7485 } 7486 } 7487 fmt->channels = info->channels; 7488 changes = 1; 7489 } 7490 7491 if (changes) { 7492 if (audio_check_params(fmt) != 0) 7493 return -1; 7494 } 7495 7496 return changes; 7497 } 7498 7499 /* 7500 * Change water marks for playback track if specified. 7501 */ 7502 static void 7503 audio_track_setinfo_water(audio_track_t *track, const struct audio_info *ai) 7504 { 7505 u_int blks; 7506 u_int maxblks; 7507 u_int blksize; 7508 7509 KASSERT(audio_track_is_playback(track)); 7510 7511 blksize = track->usrbuf_blksize; 7512 maxblks = track->usrbuf.capacity / blksize; 7513 7514 if (SPECIFIED(ai->hiwat)) { 7515 blks = ai->hiwat; 7516 if (blks > maxblks) 7517 blks = maxblks; 7518 if (blks < 2) 7519 blks = 2; 7520 track->usrbuf_usedhigh = blks * blksize; 7521 } 7522 if (SPECIFIED(ai->lowat)) { 7523 blks = ai->lowat; 7524 if (blks > maxblks - 1) 7525 blks = maxblks - 1; 7526 track->usrbuf_usedlow = blks * blksize; 7527 } 7528 if (SPECIFIED(ai->hiwat) || SPECIFIED(ai->lowat)) { 7529 if (track->usrbuf_usedlow > track->usrbuf_usedhigh - blksize) { 7530 track->usrbuf_usedlow = track->usrbuf_usedhigh - 7531 blksize; 7532 } 7533 } 7534 } 7535 7536 /* 7537 * Set hardware part of *newai. 7538 * The parameters handled here are *.port, *.gain, *.balance and monitor_gain. 7539 * If oldai is specified, previous parameters are stored. 7540 * This function itself does not roll back if error occurred. 7541 * Must be called with sc_lock && sc_exlock held. 7542 */ 7543 static int 7544 audio_hw_setinfo(struct audio_softc *sc, const struct audio_info *newai, 7545 struct audio_info *oldai) 7546 { 7547 const struct audio_prinfo *newpi; 7548 const struct audio_prinfo *newri; 7549 struct audio_prinfo *oldpi; 7550 struct audio_prinfo *oldri; 7551 u_int pgain; 7552 u_int rgain; 7553 u_char pbalance; 7554 u_char rbalance; 7555 int error; 7556 7557 KASSERT(mutex_owned(sc->sc_lock)); 7558 KASSERT(sc->sc_exlock); 7559 7560 /* XXX shut up gcc */ 7561 oldpi = NULL; 7562 oldri = NULL; 7563 7564 newpi = &newai->play; 7565 newri = &newai->record; 7566 if (oldai) { 7567 oldpi = &oldai->play; 7568 oldri = &oldai->record; 7569 } 7570 error = 0; 7571 7572 /* 7573 * It looks like unnecessary to halt HW mixers to set HW mixers. 7574 * mixer_ioctl(MIXER_WRITE) also doesn't halt. 7575 */ 7576 7577 if (SPECIFIED(newpi->port)) { 7578 if (oldai) 7579 oldpi->port = au_get_port(sc, &sc->sc_outports); 7580 error = au_set_port(sc, &sc->sc_outports, newpi->port); 7581 if (error) { 7582 audio_printf(sc, 7583 "setting play.port=%d failed: errno=%d\n", 7584 newpi->port, error); 7585 goto abort; 7586 } 7587 } 7588 if (SPECIFIED(newri->port)) { 7589 if (oldai) 7590 oldri->port = au_get_port(sc, &sc->sc_inports); 7591 error = au_set_port(sc, &sc->sc_inports, newri->port); 7592 if (error) { 7593 audio_printf(sc, 7594 "setting record.port=%d failed: errno=%d\n", 7595 newri->port, error); 7596 goto abort; 7597 } 7598 } 7599 7600 /* play.{gain,balance} */ 7601 if (SPECIFIED(newpi->gain) || SPECIFIED_CH(newpi->balance)) { 7602 au_get_gain(sc, &sc->sc_outports, &pgain, &pbalance); 7603 if (oldai) { 7604 oldpi->gain = pgain; 7605 oldpi->balance = pbalance; 7606 } 7607 7608 if (SPECIFIED(newpi->gain)) 7609 pgain = newpi->gain; 7610 if (SPECIFIED_CH(newpi->balance)) 7611 pbalance = newpi->balance; 7612 error = au_set_gain(sc, &sc->sc_outports, pgain, pbalance); 7613 if (error) { 7614 audio_printf(sc, 7615 "setting play.gain=%d/balance=%d failed: " 7616 "errno=%d\n", 7617 pgain, pbalance, error); 7618 goto abort; 7619 } 7620 } 7621 7622 /* record.{gain,balance} */ 7623 if (SPECIFIED(newri->gain) || SPECIFIED_CH(newri->balance)) { 7624 au_get_gain(sc, &sc->sc_inports, &rgain, &rbalance); 7625 if (oldai) { 7626 oldri->gain = rgain; 7627 oldri->balance = rbalance; 7628 } 7629 7630 if (SPECIFIED(newri->gain)) 7631 rgain = newri->gain; 7632 if (SPECIFIED_CH(newri->balance)) 7633 rbalance = newri->balance; 7634 error = au_set_gain(sc, &sc->sc_inports, rgain, rbalance); 7635 if (error) { 7636 audio_printf(sc, 7637 "setting record.gain=%d/balance=%d failed: " 7638 "errno=%d\n", 7639 rgain, rbalance, error); 7640 goto abort; 7641 } 7642 } 7643 7644 if (SPECIFIED(newai->monitor_gain) && sc->sc_monitor_port != -1) { 7645 if (oldai) 7646 oldai->monitor_gain = au_get_monitor_gain(sc); 7647 error = au_set_monitor_gain(sc, newai->monitor_gain); 7648 if (error) { 7649 audio_printf(sc, 7650 "setting monitor_gain=%d failed: errno=%d\n", 7651 newai->monitor_gain, error); 7652 goto abort; 7653 } 7654 } 7655 7656 /* XXX TODO */ 7657 /* sc->sc_ai = *ai; */ 7658 7659 error = 0; 7660 abort: 7661 return error; 7662 } 7663 7664 /* 7665 * Setup the hardware with mixer format phwfmt, rhwfmt. 7666 * The arguments have following restrictions: 7667 * - setmode is the direction you want to set, AUMODE_PLAY or AUMODE_RECORD, 7668 * or both. 7669 * - phwfmt and rhwfmt must not be NULL regardless of setmode. 7670 * - On non-independent devices, phwfmt and rhwfmt must have the same 7671 * parameters. 7672 * - pfil and rfil must be zero-filled. 7673 * If successful, 7674 * - pfil, rfil will be filled with filter information specified by the 7675 * hardware driver if necessary. 7676 * and then returns 0. Otherwise returns errno. 7677 * Must be called without sc_lock held. 7678 */ 7679 static int 7680 audio_hw_set_format(struct audio_softc *sc, int setmode, 7681 const audio_format2_t *phwfmt, const audio_format2_t *rhwfmt, 7682 audio_filter_reg_t *pfil, audio_filter_reg_t *rfil) 7683 { 7684 audio_params_t pp, rp; 7685 int error; 7686 7687 KASSERT(phwfmt != NULL); 7688 KASSERT(rhwfmt != NULL); 7689 7690 pp = format2_to_params(phwfmt); 7691 rp = format2_to_params(rhwfmt); 7692 7693 mutex_enter(sc->sc_lock); 7694 error = sc->hw_if->set_format(sc->hw_hdl, setmode, 7695 &pp, &rp, pfil, rfil); 7696 if (error) { 7697 mutex_exit(sc->sc_lock); 7698 audio_printf(sc, "set_format failed: errno=%d\n", error); 7699 return error; 7700 } 7701 7702 if (sc->hw_if->commit_settings) { 7703 error = sc->hw_if->commit_settings(sc->hw_hdl); 7704 if (error) { 7705 mutex_exit(sc->sc_lock); 7706 audio_printf(sc, 7707 "commit_settings failed: errno=%d\n", error); 7708 return error; 7709 } 7710 } 7711 mutex_exit(sc->sc_lock); 7712 7713 return 0; 7714 } 7715 7716 /* 7717 * Fill audio_info structure. If need_mixerinfo is true, it will also 7718 * fill the hardware mixer information. 7719 * Must be called with sc_exlock held and without sc_lock held. 7720 */ 7721 static int 7722 audiogetinfo(struct audio_softc *sc, struct audio_info *ai, int need_mixerinfo, 7723 audio_file_t *file) 7724 { 7725 struct audio_prinfo *ri, *pi; 7726 audio_track_t *track; 7727 audio_track_t *ptrack; 7728 audio_track_t *rtrack; 7729 int gain; 7730 7731 KASSERT(sc->sc_exlock); 7732 7733 ri = &ai->record; 7734 pi = &ai->play; 7735 ptrack = file->ptrack; 7736 rtrack = file->rtrack; 7737 7738 memset(ai, 0, sizeof(*ai)); 7739 7740 if (ptrack) { 7741 pi->sample_rate = ptrack->usrbuf.fmt.sample_rate; 7742 pi->channels = ptrack->usrbuf.fmt.channels; 7743 pi->precision = ptrack->usrbuf.fmt.precision; 7744 pi->encoding = ptrack->usrbuf.fmt.encoding; 7745 pi->pause = ptrack->is_pause; 7746 } else { 7747 /* Use sticky parameters if the track is not available. */ 7748 pi->sample_rate = sc->sc_sound_pparams.sample_rate; 7749 pi->channels = sc->sc_sound_pparams.channels; 7750 pi->precision = sc->sc_sound_pparams.precision; 7751 pi->encoding = sc->sc_sound_pparams.encoding; 7752 pi->pause = sc->sc_sound_ppause; 7753 } 7754 if (rtrack) { 7755 ri->sample_rate = rtrack->usrbuf.fmt.sample_rate; 7756 ri->channels = rtrack->usrbuf.fmt.channels; 7757 ri->precision = rtrack->usrbuf.fmt.precision; 7758 ri->encoding = rtrack->usrbuf.fmt.encoding; 7759 ri->pause = rtrack->is_pause; 7760 } else { 7761 /* Use sticky parameters if the track is not available. */ 7762 ri->sample_rate = sc->sc_sound_rparams.sample_rate; 7763 ri->channels = sc->sc_sound_rparams.channels; 7764 ri->precision = sc->sc_sound_rparams.precision; 7765 ri->encoding = sc->sc_sound_rparams.encoding; 7766 ri->pause = sc->sc_sound_rpause; 7767 } 7768 7769 if (ptrack) { 7770 pi->seek = ptrack->usrbuf.used; 7771 pi->samples = ptrack->stamp * ptrack->usrbuf_blksize; 7772 pi->eof = ptrack->eofcounter; 7773 pi->error = (ptrack->dropframes != 0) ? 1 : 0; 7774 pi->open = 1; 7775 pi->buffer_size = ptrack->usrbuf.capacity; 7776 } 7777 pi->waiting = 0; /* open never hangs */ 7778 pi->active = sc->sc_pbusy; 7779 7780 if (rtrack) { 7781 ri->seek = audio_track_readablebytes(rtrack); 7782 ri->samples = rtrack->stamp * rtrack->usrbuf_blksize; 7783 ri->eof = 0; 7784 ri->error = (rtrack->dropframes != 0) ? 1 : 0; 7785 ri->open = 1; 7786 ri->buffer_size = audio_track_inputblk_as_usrbyte(rtrack, 7787 rtrack->input->capacity); 7788 } 7789 ri->waiting = 0; /* open never hangs */ 7790 ri->active = sc->sc_rbusy; 7791 7792 /* 7793 * XXX There may be different number of channels between playback 7794 * and recording, so that blocksize also may be different. 7795 * But struct audio_info has an united blocksize... 7796 * Here, I use play info precedencely if ptrack is available, 7797 * otherwise record info. 7798 * 7799 * XXX hiwat/lowat is a playback-only parameter. What should I 7800 * return for a record-only descriptor? 7801 */ 7802 track = ptrack ? ptrack : rtrack; 7803 if (track) { 7804 ai->blocksize = track->usrbuf_blksize; 7805 ai->hiwat = track->usrbuf_usedhigh / track->usrbuf_blksize; 7806 ai->lowat = track->usrbuf_usedlow / track->usrbuf_blksize; 7807 } 7808 ai->mode = file->mode; 7809 7810 /* 7811 * For backward compatibility, we have to pad these five fields 7812 * a fake non-zero value even if there are no tracks. 7813 */ 7814 if (ptrack == NULL) 7815 pi->buffer_size = 65536; 7816 if (rtrack == NULL) 7817 ri->buffer_size = 65536; 7818 if (ptrack == NULL && rtrack == NULL) { 7819 ai->blocksize = 2048; 7820 ai->hiwat = ai->play.buffer_size / ai->blocksize; 7821 ai->lowat = ai->hiwat * 3 / 4; 7822 } 7823 7824 if (need_mixerinfo) { 7825 mutex_enter(sc->sc_lock); 7826 7827 pi->port = au_get_port(sc, &sc->sc_outports); 7828 ri->port = au_get_port(sc, &sc->sc_inports); 7829 7830 pi->avail_ports = sc->sc_outports.allports; 7831 ri->avail_ports = sc->sc_inports.allports; 7832 7833 au_get_gain(sc, &sc->sc_outports, &pi->gain, &pi->balance); 7834 au_get_gain(sc, &sc->sc_inports, &ri->gain, &ri->balance); 7835 7836 if (sc->sc_monitor_port != -1) { 7837 gain = au_get_monitor_gain(sc); 7838 if (gain != -1) 7839 ai->monitor_gain = gain; 7840 } 7841 mutex_exit(sc->sc_lock); 7842 } 7843 7844 return 0; 7845 } 7846 7847 /* 7848 * Return true if playback is configured. 7849 * This function can be used after audioattach. 7850 */ 7851 static bool 7852 audio_can_playback(struct audio_softc *sc) 7853 { 7854 7855 return (sc->sc_pmixer != NULL); 7856 } 7857 7858 /* 7859 * Return true if recording is configured. 7860 * This function can be used after audioattach. 7861 */ 7862 static bool 7863 audio_can_capture(struct audio_softc *sc) 7864 { 7865 7866 return (sc->sc_rmixer != NULL); 7867 } 7868 7869 /* 7870 * Get the afp->index'th item from the valid one of format[]. 7871 * If found, stores it to afp->fmt and returns 0. Otherwise return EINVAL. 7872 * 7873 * This is common routines for query_format. 7874 * If your hardware driver has struct audio_format[], the simplest case 7875 * you can write your query_format interface as follows: 7876 * 7877 * struct audio_format foo_format[] = { ... }; 7878 * 7879 * int 7880 * foo_query_format(void *hdl, audio_format_query_t *afp) 7881 * { 7882 * return audio_query_format(foo_format, __arraycount(foo_format), afp); 7883 * } 7884 */ 7885 int 7886 audio_query_format(const struct audio_format *format, int nformats, 7887 audio_format_query_t *afp) 7888 { 7889 const struct audio_format *f; 7890 int idx; 7891 int i; 7892 7893 idx = 0; 7894 for (i = 0; i < nformats; i++) { 7895 f = &format[i]; 7896 if (!AUFMT_IS_VALID(f)) 7897 continue; 7898 if (afp->index == idx) { 7899 afp->fmt = *f; 7900 return 0; 7901 } 7902 idx++; 7903 } 7904 return EINVAL; 7905 } 7906 7907 /* 7908 * This function is provided for the hardware driver's set_format() to 7909 * find index matches with 'param' from array of audio_format_t 'formats'. 7910 * 'mode' is either of AUMODE_PLAY or AUMODE_RECORD. 7911 * It returns the matched index and never fails. Because param passed to 7912 * set_format() is selected from query_format(). 7913 * This function will be an alternative to auconv_set_converter() to 7914 * find index. 7915 */ 7916 int 7917 audio_indexof_format(const struct audio_format *formats, int nformats, 7918 int mode, const audio_params_t *param) 7919 { 7920 const struct audio_format *f; 7921 int index; 7922 int j; 7923 7924 for (index = 0; index < nformats; index++) { 7925 f = &formats[index]; 7926 7927 if (!AUFMT_IS_VALID(f)) 7928 continue; 7929 if ((f->mode & mode) == 0) 7930 continue; 7931 if (f->encoding != param->encoding) 7932 continue; 7933 if (f->validbits != param->precision) 7934 continue; 7935 if (f->channels != param->channels) 7936 continue; 7937 7938 if (f->frequency_type == 0) { 7939 if (param->sample_rate < f->frequency[0] || 7940 param->sample_rate > f->frequency[1]) 7941 continue; 7942 } else { 7943 for (j = 0; j < f->frequency_type; j++) { 7944 if (param->sample_rate == f->frequency[j]) 7945 break; 7946 } 7947 if (j == f->frequency_type) 7948 continue; 7949 } 7950 7951 /* Then, matched */ 7952 return index; 7953 } 7954 7955 /* Not matched. This should not be happened. */ 7956 panic("%s: cannot find matched format\n", __func__); 7957 } 7958 7959 /* 7960 * Get or set hardware blocksize in msec. 7961 * XXX It's for debug. 7962 */ 7963 static int 7964 audio_sysctl_blk_ms(SYSCTLFN_ARGS) 7965 { 7966 struct sysctlnode node; 7967 struct audio_softc *sc; 7968 audio_format2_t phwfmt; 7969 audio_format2_t rhwfmt; 7970 audio_filter_reg_t pfil; 7971 audio_filter_reg_t rfil; 7972 int t; 7973 int old_blk_ms; 7974 int mode; 7975 int error; 7976 7977 node = *rnode; 7978 sc = node.sysctl_data; 7979 7980 error = audio_exlock_enter(sc); 7981 if (error) 7982 return error; 7983 7984 old_blk_ms = sc->sc_blk_ms; 7985 t = old_blk_ms; 7986 node.sysctl_data = &t; 7987 error = sysctl_lookup(SYSCTLFN_CALL(&node)); 7988 if (error || newp == NULL) 7989 goto abort; 7990 7991 if (t < 0) { 7992 error = EINVAL; 7993 goto abort; 7994 } 7995 7996 if (sc->sc_popens + sc->sc_ropens > 0) { 7997 error = EBUSY; 7998 goto abort; 7999 } 8000 sc->sc_blk_ms = t; 8001 mode = 0; 8002 if (sc->sc_pmixer) { 8003 mode |= AUMODE_PLAY; 8004 phwfmt = sc->sc_pmixer->hwbuf.fmt; 8005 } 8006 if (sc->sc_rmixer) { 8007 mode |= AUMODE_RECORD; 8008 rhwfmt = sc->sc_rmixer->hwbuf.fmt; 8009 } 8010 8011 /* re-init hardware */ 8012 memset(&pfil, 0, sizeof(pfil)); 8013 memset(&rfil, 0, sizeof(rfil)); 8014 error = audio_hw_set_format(sc, mode, &phwfmt, &rhwfmt, &pfil, &rfil); 8015 if (error) { 8016 goto abort; 8017 } 8018 8019 /* re-init track mixer */ 8020 error = audio_mixers_init(sc, mode, &phwfmt, &rhwfmt, &pfil, &rfil); 8021 if (error) { 8022 /* Rollback */ 8023 sc->sc_blk_ms = old_blk_ms; 8024 audio_mixers_init(sc, mode, &phwfmt, &rhwfmt, &pfil, &rfil); 8025 goto abort; 8026 } 8027 error = 0; 8028 abort: 8029 audio_exlock_exit(sc); 8030 return error; 8031 } 8032 8033 /* 8034 * Get or set multiuser mode. 8035 */ 8036 static int 8037 audio_sysctl_multiuser(SYSCTLFN_ARGS) 8038 { 8039 struct sysctlnode node; 8040 struct audio_softc *sc; 8041 bool t; 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 t = sc->sc_multiuser; 8052 node.sysctl_data = &t; 8053 error = sysctl_lookup(SYSCTLFN_CALL(&node)); 8054 if (error || newp == NULL) 8055 goto abort; 8056 8057 sc->sc_multiuser = t; 8058 error = 0; 8059 abort: 8060 audio_exlock_exit(sc); 8061 return error; 8062 } 8063 8064 #if defined(AUDIO_DEBUG) 8065 /* 8066 * Get or set debug verbose level. (0..4) 8067 * XXX It's for debug. 8068 * XXX It is not separated per device. 8069 */ 8070 static int 8071 audio_sysctl_debug(SYSCTLFN_ARGS) 8072 { 8073 struct sysctlnode node; 8074 int t; 8075 int error; 8076 8077 node = *rnode; 8078 t = audiodebug; 8079 node.sysctl_data = &t; 8080 error = sysctl_lookup(SYSCTLFN_CALL(&node)); 8081 if (error || newp == NULL) 8082 return error; 8083 8084 if (t < 0 || t > 4) 8085 return EINVAL; 8086 audiodebug = t; 8087 printf("audio: audiodebug = %d\n", audiodebug); 8088 return 0; 8089 } 8090 #endif /* AUDIO_DEBUG */ 8091 8092 #ifdef AUDIO_PM_IDLE 8093 static void 8094 audio_idle(void *arg) 8095 { 8096 device_t dv = arg; 8097 struct audio_softc *sc = device_private(dv); 8098 8099 #ifdef PNP_DEBUG 8100 extern int pnp_debug_idle; 8101 if (pnp_debug_idle) 8102 printf("%s: idle handler called\n", device_xname(dv)); 8103 #endif 8104 8105 sc->sc_idle = true; 8106 8107 /* XXX joerg Make pmf_device_suspend handle children? */ 8108 if (!pmf_device_suspend(dv, PMF_Q_SELF)) 8109 return; 8110 8111 if (!pmf_device_suspend(sc->hw_dev, PMF_Q_SELF)) 8112 pmf_device_resume(dv, PMF_Q_SELF); 8113 } 8114 8115 static void 8116 audio_activity(device_t dv, devactive_t type) 8117 { 8118 struct audio_softc *sc = device_private(dv); 8119 8120 if (type != DVA_SYSTEM) 8121 return; 8122 8123 callout_schedule(&sc->sc_idle_counter, audio_idle_timeout * hz); 8124 8125 sc->sc_idle = false; 8126 if (!device_is_active(dv)) { 8127 /* XXX joerg How to deal with a failing resume... */ 8128 pmf_device_resume(sc->hw_dev, PMF_Q_SELF); 8129 pmf_device_resume(dv, PMF_Q_SELF); 8130 } 8131 } 8132 #endif 8133 8134 static bool 8135 audio_suspend(device_t dv, const pmf_qual_t *qual) 8136 { 8137 struct audio_softc *sc = device_private(dv); 8138 int error; 8139 8140 error = audio_exlock_mutex_enter(sc); 8141 if (error) 8142 return error; 8143 sc->sc_suspending = true; 8144 audio_mixer_capture(sc); 8145 8146 if (sc->sc_pbusy) { 8147 audio_pmixer_halt(sc); 8148 /* Reuse this as need-to-restart flag while suspending */ 8149 sc->sc_pbusy = true; 8150 } 8151 if (sc->sc_rbusy) { 8152 audio_rmixer_halt(sc); 8153 /* Reuse this as need-to-restart flag while suspending */ 8154 sc->sc_rbusy = true; 8155 } 8156 8157 #ifdef AUDIO_PM_IDLE 8158 callout_halt(&sc->sc_idle_counter, sc->sc_lock); 8159 #endif 8160 audio_exlock_mutex_exit(sc); 8161 8162 return true; 8163 } 8164 8165 static bool 8166 audio_resume(device_t dv, const pmf_qual_t *qual) 8167 { 8168 struct audio_softc *sc = device_private(dv); 8169 struct audio_info ai; 8170 int error; 8171 8172 error = audio_exlock_mutex_enter(sc); 8173 if (error) 8174 return error; 8175 8176 sc->sc_suspending = false; 8177 audio_mixer_restore(sc); 8178 /* XXX ? */ 8179 AUDIO_INITINFO(&ai); 8180 audio_hw_setinfo(sc, &ai, NULL); 8181 8182 /* 8183 * During from suspend to resume here, sc_[pr]busy is used as 8184 * need-to-restart flag temporarily. After this point, 8185 * sc_[pr]busy is returned to its original usage (busy flag). 8186 * And note that sc_[pr]busy must be false to call [pr]mixer_start(). 8187 */ 8188 if (sc->sc_pbusy) { 8189 /* pmixer_start() requires pbusy is false */ 8190 sc->sc_pbusy = false; 8191 audio_pmixer_start(sc, true); 8192 } 8193 if (sc->sc_rbusy) { 8194 /* rmixer_start() requires rbusy is false */ 8195 sc->sc_rbusy = false; 8196 audio_rmixer_start(sc); 8197 } 8198 8199 audio_exlock_mutex_exit(sc); 8200 8201 return true; 8202 } 8203 8204 #if defined(AUDIO_DEBUG) 8205 static void 8206 audio_format2_tostr(char *buf, size_t bufsize, const audio_format2_t *fmt) 8207 { 8208 int n; 8209 8210 n = 0; 8211 n += snprintf(buf + n, bufsize - n, "%s", 8212 audio_encoding_name(fmt->encoding)); 8213 if (fmt->precision == fmt->stride) { 8214 n += snprintf(buf + n, bufsize - n, " %dbit", fmt->precision); 8215 } else { 8216 n += snprintf(buf + n, bufsize - n, " %d/%dbit", 8217 fmt->precision, fmt->stride); 8218 } 8219 8220 snprintf(buf + n, bufsize - n, " %uch %uHz", 8221 fmt->channels, fmt->sample_rate); 8222 } 8223 #endif 8224 8225 #if defined(AUDIO_DEBUG) 8226 static void 8227 audio_print_format2(const char *s, const audio_format2_t *fmt) 8228 { 8229 char fmtstr[64]; 8230 8231 audio_format2_tostr(fmtstr, sizeof(fmtstr), fmt); 8232 printf("%s %s\n", s, fmtstr); 8233 } 8234 #endif 8235 8236 #ifdef DIAGNOSTIC 8237 void 8238 audio_diagnostic_format2(const char *where, const audio_format2_t *fmt) 8239 { 8240 8241 KASSERTMSG(fmt, "called from %s", where); 8242 8243 /* XXX MSM6258 vs(4) only has 4bit stride format. */ 8244 if (fmt->encoding == AUDIO_ENCODING_ADPCM) { 8245 KASSERTMSG(fmt->stride == 4 || fmt->stride == 8, 8246 "called from %s: fmt->stride=%d", where, fmt->stride); 8247 } else { 8248 KASSERTMSG(fmt->stride % NBBY == 0, 8249 "called from %s: fmt->stride=%d", where, fmt->stride); 8250 } 8251 KASSERTMSG(fmt->precision <= fmt->stride, 8252 "called from %s: fmt->precision=%d fmt->stride=%d", 8253 where, fmt->precision, fmt->stride); 8254 KASSERTMSG(1 <= fmt->channels && fmt->channels <= AUDIO_MAX_CHANNELS, 8255 "called from %s: fmt->channels=%d", where, fmt->channels); 8256 8257 /* XXX No check for encodings? */ 8258 } 8259 8260 void 8261 audio_diagnostic_filter_arg(const char *where, const audio_filter_arg_t *arg) 8262 { 8263 8264 KASSERT(arg != NULL); 8265 KASSERT(arg->src != NULL); 8266 KASSERT(arg->dst != NULL); 8267 audio_diagnostic_format2(where, arg->srcfmt); 8268 audio_diagnostic_format2(where, arg->dstfmt); 8269 KASSERT(arg->count > 0); 8270 } 8271 8272 void 8273 audio_diagnostic_ring(const char *where, const audio_ring_t *ring) 8274 { 8275 8276 KASSERTMSG(ring, "called from %s", where); 8277 audio_diagnostic_format2(where, &ring->fmt); 8278 KASSERTMSG(0 <= ring->capacity && ring->capacity < INT_MAX / 2, 8279 "called from %s: ring->capacity=%d", where, ring->capacity); 8280 KASSERTMSG(0 <= ring->used && ring->used <= ring->capacity, 8281 "called from %s: ring->used=%d ring->capacity=%d", 8282 where, ring->used, ring->capacity); 8283 if (ring->capacity == 0) { 8284 KASSERTMSG(ring->mem == NULL, 8285 "called from %s: capacity == 0 but mem != NULL", where); 8286 } else { 8287 KASSERTMSG(ring->mem != NULL, 8288 "called from %s: capacity != 0 but mem == NULL", where); 8289 KASSERTMSG(0 <= ring->head && ring->head < ring->capacity, 8290 "called from %s: ring->head=%d ring->capacity=%d", 8291 where, ring->head, ring->capacity); 8292 } 8293 } 8294 #endif /* DIAGNOSTIC */ 8295 8296 8297 /* 8298 * Mixer driver 8299 */ 8300 8301 /* 8302 * Must be called without sc_lock held. 8303 */ 8304 int 8305 mixer_open(dev_t dev, struct audio_softc *sc, int flags, int ifmt, 8306 struct lwp *l) 8307 { 8308 struct file *fp; 8309 audio_file_t *af; 8310 int error, fd; 8311 8312 TRACE(1, "flags=0x%x", flags); 8313 8314 error = fd_allocfile(&fp, &fd); 8315 if (error) 8316 return error; 8317 8318 af = kmem_zalloc(sizeof(*af), KM_SLEEP); 8319 af->sc = sc; 8320 af->dev = dev; 8321 8322 mutex_enter(sc->sc_lock); 8323 if (sc->sc_dying) { 8324 mutex_exit(sc->sc_lock); 8325 kmem_free(af, sizeof(*af)); 8326 fd_abort(curproc, fp, fd); 8327 return ENXIO; 8328 } 8329 mutex_enter(sc->sc_intr_lock); 8330 SLIST_INSERT_HEAD(&sc->sc_files, af, entry); 8331 mutex_exit(sc->sc_intr_lock); 8332 mutex_exit(sc->sc_lock); 8333 8334 error = fd_clone(fp, fd, flags, &audio_fileops, af); 8335 KASSERT(error == EMOVEFD); 8336 8337 return error; 8338 } 8339 8340 /* 8341 * Add a process to those to be signalled on mixer activity. 8342 * If the process has already been added, do nothing. 8343 * Must be called with sc_exlock held and without sc_lock held. 8344 */ 8345 static void 8346 mixer_async_add(struct audio_softc *sc, pid_t pid) 8347 { 8348 int i; 8349 8350 KASSERT(sc->sc_exlock); 8351 8352 /* If already exists, returns without doing anything. */ 8353 for (i = 0; i < sc->sc_am_used; i++) { 8354 if (sc->sc_am[i] == pid) 8355 return; 8356 } 8357 8358 /* Extend array if necessary. */ 8359 if (sc->sc_am_used >= sc->sc_am_capacity) { 8360 sc->sc_am_capacity += AM_CAPACITY; 8361 sc->sc_am = kern_realloc(sc->sc_am, 8362 sc->sc_am_capacity * sizeof(pid_t), M_WAITOK); 8363 TRACE(2, "realloc am_capacity=%d", sc->sc_am_capacity); 8364 } 8365 8366 TRACE(2, "am[%d]=%d", sc->sc_am_used, (int)pid); 8367 sc->sc_am[sc->sc_am_used++] = pid; 8368 } 8369 8370 /* 8371 * Remove a process from those to be signalled on mixer activity. 8372 * If the process has not been added, do nothing. 8373 * Must be called with sc_exlock held and without sc_lock held. 8374 */ 8375 static void 8376 mixer_async_remove(struct audio_softc *sc, pid_t pid) 8377 { 8378 int i; 8379 8380 KASSERT(sc->sc_exlock); 8381 8382 for (i = 0; i < sc->sc_am_used; i++) { 8383 if (sc->sc_am[i] == pid) { 8384 sc->sc_am[i] = sc->sc_am[--sc->sc_am_used]; 8385 TRACE(2, "am[%d](%d) removed, used=%d", 8386 i, (int)pid, sc->sc_am_used); 8387 8388 /* Empty array if no longer necessary. */ 8389 if (sc->sc_am_used == 0) { 8390 kern_free(sc->sc_am); 8391 sc->sc_am = NULL; 8392 sc->sc_am_capacity = 0; 8393 TRACE(2, "released"); 8394 } 8395 return; 8396 } 8397 } 8398 } 8399 8400 /* 8401 * Signal all processes waiting for the mixer. 8402 * Must be called with sc_exlock held. 8403 */ 8404 static void 8405 mixer_signal(struct audio_softc *sc) 8406 { 8407 proc_t *p; 8408 int i; 8409 8410 KASSERT(sc->sc_exlock); 8411 8412 for (i = 0; i < sc->sc_am_used; i++) { 8413 mutex_enter(&proc_lock); 8414 p = proc_find(sc->sc_am[i]); 8415 if (p) 8416 psignal(p, SIGIO); 8417 mutex_exit(&proc_lock); 8418 } 8419 } 8420 8421 /* 8422 * Close a mixer device 8423 */ 8424 int 8425 mixer_close(struct audio_softc *sc, audio_file_t *file) 8426 { 8427 int error; 8428 8429 error = audio_exlock_enter(sc); 8430 if (error) 8431 return error; 8432 TRACE(1, "called"); 8433 mixer_async_remove(sc, curproc->p_pid); 8434 audio_exlock_exit(sc); 8435 8436 return 0; 8437 } 8438 8439 /* 8440 * Must be called without sc_lock nor sc_exlock held. 8441 */ 8442 int 8443 mixer_ioctl(struct audio_softc *sc, u_long cmd, void *addr, int flag, 8444 struct lwp *l) 8445 { 8446 mixer_devinfo_t *mi; 8447 mixer_ctrl_t *mc; 8448 int val; 8449 int error; 8450 8451 #if defined(AUDIO_DEBUG) 8452 char pre[64]; 8453 snprintf(pre, sizeof(pre), "pid=%d.%d", 8454 (int)curproc->p_pid, (int)l->l_lid); 8455 #endif 8456 error = EINVAL; 8457 8458 /* we can return cached values if we are sleeping */ 8459 if (cmd != AUDIO_MIXER_READ) { 8460 mutex_enter(sc->sc_lock); 8461 device_active(sc->sc_dev, DVA_SYSTEM); 8462 mutex_exit(sc->sc_lock); 8463 } 8464 8465 switch (cmd) { 8466 case FIOASYNC: 8467 val = *(int *)addr; 8468 TRACE(2, "%s FIOASYNC %s", pre, val ? "on" : "off"); 8469 error = audio_exlock_enter(sc); 8470 if (error) 8471 break; 8472 if (val) { 8473 mixer_async_add(sc, curproc->p_pid); 8474 } else { 8475 mixer_async_remove(sc, curproc->p_pid); 8476 } 8477 audio_exlock_exit(sc); 8478 break; 8479 8480 case AUDIO_GETDEV: 8481 TRACE(2, "%s AUDIO_GETDEV", pre); 8482 error = sc->hw_if->getdev(sc->hw_hdl, (audio_device_t *)addr); 8483 break; 8484 8485 case AUDIO_MIXER_DEVINFO: 8486 TRACE(2, "%s AUDIO_MIXER_DEVINFO", pre); 8487 mi = (mixer_devinfo_t *)addr; 8488 8489 mi->un.v.delta = 0; /* default */ 8490 mutex_enter(sc->sc_lock); 8491 error = audio_query_devinfo(sc, mi); 8492 mutex_exit(sc->sc_lock); 8493 break; 8494 8495 case AUDIO_MIXER_READ: 8496 TRACE(2, "%s AUDIO_MIXER_READ", pre); 8497 mc = (mixer_ctrl_t *)addr; 8498 8499 error = audio_exlock_mutex_enter(sc); 8500 if (error) 8501 break; 8502 if (device_is_active(sc->hw_dev)) 8503 error = audio_get_port(sc, mc); 8504 else if (mc->dev < 0 || mc->dev >= sc->sc_nmixer_states) 8505 error = ENXIO; 8506 else { 8507 int dev = mc->dev; 8508 memcpy(mc, &sc->sc_mixer_state[dev], 8509 sizeof(mixer_ctrl_t)); 8510 error = 0; 8511 } 8512 audio_exlock_mutex_exit(sc); 8513 break; 8514 8515 case AUDIO_MIXER_WRITE: 8516 TRACE(2, "%s AUDIO_MIXER_WRITE", pre); 8517 error = audio_exlock_mutex_enter(sc); 8518 if (error) 8519 break; 8520 error = audio_set_port(sc, (mixer_ctrl_t *)addr); 8521 if (error) { 8522 audio_exlock_mutex_exit(sc); 8523 break; 8524 } 8525 8526 if (sc->hw_if->commit_settings) { 8527 error = sc->hw_if->commit_settings(sc->hw_hdl); 8528 if (error) { 8529 audio_exlock_mutex_exit(sc); 8530 break; 8531 } 8532 } 8533 mutex_exit(sc->sc_lock); 8534 mixer_signal(sc); 8535 audio_exlock_exit(sc); 8536 break; 8537 8538 default: 8539 TRACE(2, "(%lu,'%c',%lu)", 8540 IOCPARM_LEN(cmd), (char)IOCGROUP(cmd), cmd & 0xff); 8541 if (sc->hw_if->dev_ioctl) { 8542 mutex_enter(sc->sc_lock); 8543 error = sc->hw_if->dev_ioctl(sc->hw_hdl, 8544 cmd, addr, flag, l); 8545 mutex_exit(sc->sc_lock); 8546 } else 8547 error = EINVAL; 8548 break; 8549 } 8550 8551 if (error) 8552 TRACE(2, "error=%d", error); 8553 return error; 8554 } 8555 8556 /* 8557 * Must be called with sc_lock held. 8558 */ 8559 int 8560 au_portof(struct audio_softc *sc, char *name, int class) 8561 { 8562 mixer_devinfo_t mi; 8563 8564 KASSERT(mutex_owned(sc->sc_lock)); 8565 8566 for (mi.index = 0; audio_query_devinfo(sc, &mi) == 0; mi.index++) { 8567 if (mi.mixer_class == class && strcmp(mi.label.name, name) == 0) 8568 return mi.index; 8569 } 8570 return -1; 8571 } 8572 8573 /* 8574 * Must be called with sc_lock held. 8575 */ 8576 void 8577 au_setup_ports(struct audio_softc *sc, struct au_mixer_ports *ports, 8578 mixer_devinfo_t *mi, const struct portname *tbl) 8579 { 8580 int i, j; 8581 8582 KASSERT(mutex_owned(sc->sc_lock)); 8583 8584 ports->index = mi->index; 8585 if (mi->type == AUDIO_MIXER_ENUM) { 8586 ports->isenum = true; 8587 for(i = 0; tbl[i].name; i++) 8588 for(j = 0; j < mi->un.e.num_mem; j++) 8589 if (strcmp(mi->un.e.member[j].label.name, 8590 tbl[i].name) == 0) { 8591 ports->allports |= tbl[i].mask; 8592 ports->aumask[ports->nports] = tbl[i].mask; 8593 ports->misel[ports->nports] = 8594 mi->un.e.member[j].ord; 8595 ports->miport[ports->nports] = 8596 au_portof(sc, mi->un.e.member[j].label.name, 8597 mi->mixer_class); 8598 if (ports->mixerout != -1 && 8599 ports->miport[ports->nports] != -1) 8600 ports->isdual = true; 8601 ++ports->nports; 8602 } 8603 } else if (mi->type == AUDIO_MIXER_SET) { 8604 for(i = 0; tbl[i].name; i++) 8605 for(j = 0; j < mi->un.s.num_mem; j++) 8606 if (strcmp(mi->un.s.member[j].label.name, 8607 tbl[i].name) == 0) { 8608 ports->allports |= tbl[i].mask; 8609 ports->aumask[ports->nports] = tbl[i].mask; 8610 ports->misel[ports->nports] = 8611 mi->un.s.member[j].mask; 8612 ports->miport[ports->nports] = 8613 au_portof(sc, mi->un.s.member[j].label.name, 8614 mi->mixer_class); 8615 ++ports->nports; 8616 } 8617 } 8618 } 8619 8620 /* 8621 * Must be called with sc_lock && sc_exlock held. 8622 */ 8623 int 8624 au_set_lr_value(struct audio_softc *sc, mixer_ctrl_t *ct, int l, int r) 8625 { 8626 8627 KASSERT(mutex_owned(sc->sc_lock)); 8628 KASSERT(sc->sc_exlock); 8629 8630 ct->type = AUDIO_MIXER_VALUE; 8631 ct->un.value.num_channels = 2; 8632 ct->un.value.level[AUDIO_MIXER_LEVEL_LEFT] = l; 8633 ct->un.value.level[AUDIO_MIXER_LEVEL_RIGHT] = r; 8634 if (audio_set_port(sc, ct) == 0) 8635 return 0; 8636 ct->un.value.num_channels = 1; 8637 ct->un.value.level[AUDIO_MIXER_LEVEL_MONO] = (l+r)/2; 8638 return audio_set_port(sc, ct); 8639 } 8640 8641 /* 8642 * Must be called with sc_lock && sc_exlock held. 8643 */ 8644 int 8645 au_get_lr_value(struct audio_softc *sc, mixer_ctrl_t *ct, int *l, int *r) 8646 { 8647 int error; 8648 8649 KASSERT(mutex_owned(sc->sc_lock)); 8650 KASSERT(sc->sc_exlock); 8651 8652 ct->un.value.num_channels = 2; 8653 if (audio_get_port(sc, ct) == 0) { 8654 *l = ct->un.value.level[AUDIO_MIXER_LEVEL_LEFT]; 8655 *r = ct->un.value.level[AUDIO_MIXER_LEVEL_RIGHT]; 8656 } else { 8657 ct->un.value.num_channels = 1; 8658 error = audio_get_port(sc, ct); 8659 if (error) 8660 return error; 8661 *r = *l = ct->un.value.level[AUDIO_MIXER_LEVEL_MONO]; 8662 } 8663 return 0; 8664 } 8665 8666 /* 8667 * Must be called with sc_lock && sc_exlock held. 8668 */ 8669 int 8670 au_set_gain(struct audio_softc *sc, struct au_mixer_ports *ports, 8671 int gain, int balance) 8672 { 8673 mixer_ctrl_t ct; 8674 int i, error; 8675 int l, r; 8676 u_int mask; 8677 int nset; 8678 8679 KASSERT(mutex_owned(sc->sc_lock)); 8680 KASSERT(sc->sc_exlock); 8681 8682 if (balance == AUDIO_MID_BALANCE) { 8683 l = r = gain; 8684 } else if (balance < AUDIO_MID_BALANCE) { 8685 l = gain; 8686 r = (balance * gain) / AUDIO_MID_BALANCE; 8687 } else { 8688 r = gain; 8689 l = ((AUDIO_RIGHT_BALANCE - balance) * gain) 8690 / AUDIO_MID_BALANCE; 8691 } 8692 TRACE(2, "gain=%d balance=%d, l=%d r=%d", gain, balance, l, r); 8693 8694 if (ports->index == -1) { 8695 usemaster: 8696 if (ports->master == -1) 8697 return 0; /* just ignore it silently */ 8698 ct.dev = ports->master; 8699 error = au_set_lr_value(sc, &ct, l, r); 8700 } else { 8701 ct.dev = ports->index; 8702 if (ports->isenum) { 8703 ct.type = AUDIO_MIXER_ENUM; 8704 error = audio_get_port(sc, &ct); 8705 if (error) 8706 return error; 8707 if (ports->isdual) { 8708 if (ports->cur_port == -1) 8709 ct.dev = ports->master; 8710 else 8711 ct.dev = ports->miport[ports->cur_port]; 8712 error = au_set_lr_value(sc, &ct, l, r); 8713 } else { 8714 for(i = 0; i < ports->nports; i++) 8715 if (ports->misel[i] == ct.un.ord) { 8716 ct.dev = ports->miport[i]; 8717 if (ct.dev == -1 || 8718 au_set_lr_value(sc, &ct, l, r)) 8719 goto usemaster; 8720 else 8721 break; 8722 } 8723 } 8724 } else { 8725 ct.type = AUDIO_MIXER_SET; 8726 error = audio_get_port(sc, &ct); 8727 if (error) 8728 return error; 8729 mask = ct.un.mask; 8730 nset = 0; 8731 for(i = 0; i < ports->nports; i++) { 8732 if (ports->misel[i] & mask) { 8733 ct.dev = ports->miport[i]; 8734 if (ct.dev != -1 && 8735 au_set_lr_value(sc, &ct, l, r) == 0) 8736 nset++; 8737 } 8738 } 8739 if (nset == 0) 8740 goto usemaster; 8741 } 8742 } 8743 if (!error) 8744 mixer_signal(sc); 8745 return error; 8746 } 8747 8748 /* 8749 * Must be called with sc_lock && sc_exlock held. 8750 */ 8751 void 8752 au_get_gain(struct audio_softc *sc, struct au_mixer_ports *ports, 8753 u_int *pgain, u_char *pbalance) 8754 { 8755 mixer_ctrl_t ct; 8756 int i, l, r, n; 8757 int lgain, rgain; 8758 8759 KASSERT(mutex_owned(sc->sc_lock)); 8760 KASSERT(sc->sc_exlock); 8761 8762 lgain = AUDIO_MAX_GAIN / 2; 8763 rgain = AUDIO_MAX_GAIN / 2; 8764 if (ports->index == -1) { 8765 usemaster: 8766 if (ports->master == -1) 8767 goto bad; 8768 ct.dev = ports->master; 8769 ct.type = AUDIO_MIXER_VALUE; 8770 if (au_get_lr_value(sc, &ct, &lgain, &rgain)) 8771 goto bad; 8772 } else { 8773 ct.dev = ports->index; 8774 if (ports->isenum) { 8775 ct.type = AUDIO_MIXER_ENUM; 8776 if (audio_get_port(sc, &ct)) 8777 goto bad; 8778 ct.type = AUDIO_MIXER_VALUE; 8779 if (ports->isdual) { 8780 if (ports->cur_port == -1) 8781 ct.dev = ports->master; 8782 else 8783 ct.dev = ports->miport[ports->cur_port]; 8784 au_get_lr_value(sc, &ct, &lgain, &rgain); 8785 } else { 8786 for(i = 0; i < ports->nports; i++) 8787 if (ports->misel[i] == ct.un.ord) { 8788 ct.dev = ports->miport[i]; 8789 if (ct.dev == -1 || 8790 au_get_lr_value(sc, &ct, 8791 &lgain, &rgain)) 8792 goto usemaster; 8793 else 8794 break; 8795 } 8796 } 8797 } else { 8798 ct.type = AUDIO_MIXER_SET; 8799 if (audio_get_port(sc, &ct)) 8800 goto bad; 8801 ct.type = AUDIO_MIXER_VALUE; 8802 lgain = rgain = n = 0; 8803 for(i = 0; i < ports->nports; i++) { 8804 if (ports->misel[i] & ct.un.mask) { 8805 ct.dev = ports->miport[i]; 8806 if (ct.dev == -1 || 8807 au_get_lr_value(sc, &ct, &l, &r)) 8808 goto usemaster; 8809 else { 8810 lgain += l; 8811 rgain += r; 8812 n++; 8813 } 8814 } 8815 } 8816 if (n != 0) { 8817 lgain /= n; 8818 rgain /= n; 8819 } 8820 } 8821 } 8822 bad: 8823 if (lgain == rgain) { /* handles lgain==rgain==0 */ 8824 *pgain = lgain; 8825 *pbalance = AUDIO_MID_BALANCE; 8826 } else if (lgain < rgain) { 8827 *pgain = rgain; 8828 /* balance should be > AUDIO_MID_BALANCE */ 8829 *pbalance = AUDIO_RIGHT_BALANCE - 8830 (AUDIO_MID_BALANCE * lgain) / rgain; 8831 } else /* lgain > rgain */ { 8832 *pgain = lgain; 8833 /* balance should be < AUDIO_MID_BALANCE */ 8834 *pbalance = (AUDIO_MID_BALANCE * rgain) / lgain; 8835 } 8836 } 8837 8838 /* 8839 * Must be called with sc_lock && sc_exlock held. 8840 */ 8841 int 8842 au_set_port(struct audio_softc *sc, struct au_mixer_ports *ports, u_int port) 8843 { 8844 mixer_ctrl_t ct; 8845 int i, error, use_mixerout; 8846 8847 KASSERT(mutex_owned(sc->sc_lock)); 8848 KASSERT(sc->sc_exlock); 8849 8850 use_mixerout = 1; 8851 if (port == 0) { 8852 if (ports->allports == 0) 8853 return 0; /* Allow this special case. */ 8854 else if (ports->isdual) { 8855 if (ports->cur_port == -1) { 8856 return 0; 8857 } else { 8858 port = ports->aumask[ports->cur_port]; 8859 ports->cur_port = -1; 8860 use_mixerout = 0; 8861 } 8862 } 8863 } 8864 if (ports->index == -1) 8865 return EINVAL; 8866 ct.dev = ports->index; 8867 if (ports->isenum) { 8868 if (port & (port-1)) 8869 return EINVAL; /* Only one port allowed */ 8870 ct.type = AUDIO_MIXER_ENUM; 8871 error = EINVAL; 8872 for(i = 0; i < ports->nports; i++) 8873 if (ports->aumask[i] == port) { 8874 if (ports->isdual && use_mixerout) { 8875 ct.un.ord = ports->mixerout; 8876 ports->cur_port = i; 8877 } else { 8878 ct.un.ord = ports->misel[i]; 8879 } 8880 error = audio_set_port(sc, &ct); 8881 break; 8882 } 8883 } else { 8884 ct.type = AUDIO_MIXER_SET; 8885 ct.un.mask = 0; 8886 for(i = 0; i < ports->nports; i++) 8887 if (ports->aumask[i] & port) 8888 ct.un.mask |= ports->misel[i]; 8889 if (port != 0 && ct.un.mask == 0) 8890 error = EINVAL; 8891 else 8892 error = audio_set_port(sc, &ct); 8893 } 8894 if (!error) 8895 mixer_signal(sc); 8896 return error; 8897 } 8898 8899 /* 8900 * Must be called with sc_lock && sc_exlock held. 8901 */ 8902 int 8903 au_get_port(struct audio_softc *sc, struct au_mixer_ports *ports) 8904 { 8905 mixer_ctrl_t ct; 8906 int i, aumask; 8907 8908 KASSERT(mutex_owned(sc->sc_lock)); 8909 KASSERT(sc->sc_exlock); 8910 8911 if (ports->index == -1) 8912 return 0; 8913 ct.dev = ports->index; 8914 ct.type = ports->isenum ? AUDIO_MIXER_ENUM : AUDIO_MIXER_SET; 8915 if (audio_get_port(sc, &ct)) 8916 return 0; 8917 aumask = 0; 8918 if (ports->isenum) { 8919 if (ports->isdual && ports->cur_port != -1) { 8920 if (ports->mixerout == ct.un.ord) 8921 aumask = ports->aumask[ports->cur_port]; 8922 else 8923 ports->cur_port = -1; 8924 } 8925 if (aumask == 0) 8926 for(i = 0; i < ports->nports; i++) 8927 if (ports->misel[i] == ct.un.ord) 8928 aumask = ports->aumask[i]; 8929 } else { 8930 for(i = 0; i < ports->nports; i++) 8931 if (ct.un.mask & ports->misel[i]) 8932 aumask |= ports->aumask[i]; 8933 } 8934 return aumask; 8935 } 8936 8937 /* 8938 * It returns 0 if success, otherwise errno. 8939 * Must be called only if sc->sc_monitor_port != -1. 8940 * Must be called with sc_lock && sc_exlock held. 8941 */ 8942 static int 8943 au_set_monitor_gain(struct audio_softc *sc, int monitor_gain) 8944 { 8945 mixer_ctrl_t ct; 8946 8947 KASSERT(mutex_owned(sc->sc_lock)); 8948 KASSERT(sc->sc_exlock); 8949 8950 ct.dev = sc->sc_monitor_port; 8951 ct.type = AUDIO_MIXER_VALUE; 8952 ct.un.value.num_channels = 1; 8953 ct.un.value.level[AUDIO_MIXER_LEVEL_MONO] = monitor_gain; 8954 return audio_set_port(sc, &ct); 8955 } 8956 8957 /* 8958 * It returns monitor gain if success, otherwise -1. 8959 * Must be called only if sc->sc_monitor_port != -1. 8960 * Must be called with sc_lock && sc_exlock held. 8961 */ 8962 static int 8963 au_get_monitor_gain(struct audio_softc *sc) 8964 { 8965 mixer_ctrl_t ct; 8966 8967 KASSERT(mutex_owned(sc->sc_lock)); 8968 KASSERT(sc->sc_exlock); 8969 8970 ct.dev = sc->sc_monitor_port; 8971 ct.type = AUDIO_MIXER_VALUE; 8972 ct.un.value.num_channels = 1; 8973 if (audio_get_port(sc, &ct)) 8974 return -1; 8975 return ct.un.value.level[AUDIO_MIXER_LEVEL_MONO]; 8976 } 8977 8978 /* 8979 * Must be called with sc_lock && sc_exlock held. 8980 */ 8981 static int 8982 audio_set_port(struct audio_softc *sc, mixer_ctrl_t *mc) 8983 { 8984 8985 KASSERT(mutex_owned(sc->sc_lock)); 8986 KASSERT(sc->sc_exlock); 8987 8988 return sc->hw_if->set_port(sc->hw_hdl, mc); 8989 } 8990 8991 /* 8992 * Must be called with sc_lock && sc_exlock held. 8993 */ 8994 static int 8995 audio_get_port(struct audio_softc *sc, mixer_ctrl_t *mc) 8996 { 8997 8998 KASSERT(mutex_owned(sc->sc_lock)); 8999 KASSERT(sc->sc_exlock); 9000 9001 return sc->hw_if->get_port(sc->hw_hdl, mc); 9002 } 9003 9004 /* 9005 * Must be called with sc_lock && sc_exlock held. 9006 */ 9007 static void 9008 audio_mixer_capture(struct audio_softc *sc) 9009 { 9010 mixer_devinfo_t mi; 9011 mixer_ctrl_t *mc; 9012 9013 KASSERT(mutex_owned(sc->sc_lock)); 9014 KASSERT(sc->sc_exlock); 9015 9016 for (mi.index = 0;; mi.index++) { 9017 if (audio_query_devinfo(sc, &mi) != 0) 9018 break; 9019 KASSERT(mi.index < sc->sc_nmixer_states); 9020 if (mi.type == AUDIO_MIXER_CLASS) 9021 continue; 9022 mc = &sc->sc_mixer_state[mi.index]; 9023 mc->dev = mi.index; 9024 mc->type = mi.type; 9025 mc->un.value.num_channels = mi.un.v.num_channels; 9026 (void)audio_get_port(sc, mc); 9027 } 9028 9029 return; 9030 } 9031 9032 /* 9033 * Must be called with sc_lock && sc_exlock held. 9034 */ 9035 static void 9036 audio_mixer_restore(struct audio_softc *sc) 9037 { 9038 mixer_devinfo_t mi; 9039 mixer_ctrl_t *mc; 9040 9041 KASSERT(mutex_owned(sc->sc_lock)); 9042 KASSERT(sc->sc_exlock); 9043 9044 for (mi.index = 0; ; mi.index++) { 9045 if (audio_query_devinfo(sc, &mi) != 0) 9046 break; 9047 if (mi.type == AUDIO_MIXER_CLASS) 9048 continue; 9049 mc = &sc->sc_mixer_state[mi.index]; 9050 (void)audio_set_port(sc, mc); 9051 } 9052 if (sc->hw_if->commit_settings) 9053 sc->hw_if->commit_settings(sc->hw_hdl); 9054 9055 return; 9056 } 9057 9058 static void 9059 audio_volume_down(device_t dv) 9060 { 9061 struct audio_softc *sc = device_private(dv); 9062 mixer_devinfo_t mi; 9063 int newgain; 9064 u_int gain; 9065 u_char balance; 9066 9067 if (audio_exlock_mutex_enter(sc) != 0) 9068 return; 9069 if (sc->sc_outports.index == -1 && sc->sc_outports.master != -1) { 9070 mi.index = sc->sc_outports.master; 9071 mi.un.v.delta = 0; 9072 if (audio_query_devinfo(sc, &mi) == 0) { 9073 au_get_gain(sc, &sc->sc_outports, &gain, &balance); 9074 newgain = gain - mi.un.v.delta; 9075 if (newgain < AUDIO_MIN_GAIN) 9076 newgain = AUDIO_MIN_GAIN; 9077 au_set_gain(sc, &sc->sc_outports, newgain, balance); 9078 } 9079 } 9080 audio_exlock_mutex_exit(sc); 9081 } 9082 9083 static void 9084 audio_volume_up(device_t dv) 9085 { 9086 struct audio_softc *sc = device_private(dv); 9087 mixer_devinfo_t mi; 9088 u_int gain, newgain; 9089 u_char balance; 9090 9091 if (audio_exlock_mutex_enter(sc) != 0) 9092 return; 9093 if (sc->sc_outports.index == -1 && sc->sc_outports.master != -1) { 9094 mi.index = sc->sc_outports.master; 9095 mi.un.v.delta = 0; 9096 if (audio_query_devinfo(sc, &mi) == 0) { 9097 au_get_gain(sc, &sc->sc_outports, &gain, &balance); 9098 newgain = gain + mi.un.v.delta; 9099 if (newgain > AUDIO_MAX_GAIN) 9100 newgain = AUDIO_MAX_GAIN; 9101 au_set_gain(sc, &sc->sc_outports, newgain, balance); 9102 } 9103 } 9104 audio_exlock_mutex_exit(sc); 9105 } 9106 9107 static void 9108 audio_volume_toggle(device_t dv) 9109 { 9110 struct audio_softc *sc = device_private(dv); 9111 u_int gain, newgain; 9112 u_char balance; 9113 9114 if (audio_exlock_mutex_enter(sc) != 0) 9115 return; 9116 au_get_gain(sc, &sc->sc_outports, &gain, &balance); 9117 if (gain != 0) { 9118 sc->sc_lastgain = gain; 9119 newgain = 0; 9120 } else 9121 newgain = sc->sc_lastgain; 9122 au_set_gain(sc, &sc->sc_outports, newgain, balance); 9123 audio_exlock_mutex_exit(sc); 9124 } 9125 9126 /* 9127 * Must be called with sc_lock held. 9128 */ 9129 static int 9130 audio_query_devinfo(struct audio_softc *sc, mixer_devinfo_t *di) 9131 { 9132 9133 KASSERT(mutex_owned(sc->sc_lock)); 9134 9135 return sc->hw_if->query_devinfo(sc->hw_hdl, di); 9136 } 9137 9138 #endif /* NAUDIO > 0 */ 9139 9140 #if NAUDIO == 0 && (NMIDI > 0 || NMIDIBUS > 0) 9141 #include <sys/param.h> 9142 #include <sys/systm.h> 9143 #include <sys/device.h> 9144 #include <sys/audioio.h> 9145 #include <dev/audio/audio_if.h> 9146 #endif 9147 9148 #if NAUDIO > 0 || (NMIDI > 0 || NMIDIBUS > 0) 9149 int 9150 audioprint(void *aux, const char *pnp) 9151 { 9152 struct audio_attach_args *arg; 9153 const char *type; 9154 9155 if (pnp != NULL) { 9156 arg = aux; 9157 switch (arg->type) { 9158 case AUDIODEV_TYPE_AUDIO: 9159 type = "audio"; 9160 break; 9161 case AUDIODEV_TYPE_MIDI: 9162 type = "midi"; 9163 break; 9164 case AUDIODEV_TYPE_OPL: 9165 type = "opl"; 9166 break; 9167 case AUDIODEV_TYPE_MPU: 9168 type = "mpu"; 9169 break; 9170 case AUDIODEV_TYPE_AUX: 9171 type = "aux"; 9172 break; 9173 default: 9174 panic("audioprint: unknown type %d", arg->type); 9175 } 9176 aprint_normal("%s at %s", type, pnp); 9177 } 9178 return UNCONF; 9179 } 9180 9181 #endif /* NAUDIO > 0 || (NMIDI > 0 || NMIDIBUS > 0) */ 9182 9183 #ifdef _MODULE 9184 9185 devmajor_t audio_bmajor = -1, audio_cmajor = -1; 9186 9187 #include "ioconf.c" 9188 9189 #endif 9190 9191 MODULE(MODULE_CLASS_DRIVER, audio, NULL); 9192 9193 static int 9194 audio_modcmd(modcmd_t cmd, void *arg) 9195 { 9196 int error = 0; 9197 9198 switch (cmd) { 9199 case MODULE_CMD_INIT: 9200 /* XXX interrupt level? */ 9201 audio_psref_class = psref_class_create("audio", IPL_SOFTSERIAL); 9202 #ifdef _MODULE 9203 error = devsw_attach(audio_cd.cd_name, NULL, &audio_bmajor, 9204 &audio_cdevsw, &audio_cmajor); 9205 if (error) 9206 break; 9207 9208 error = config_init_component(cfdriver_ioconf_audio, 9209 cfattach_ioconf_audio, cfdata_ioconf_audio); 9210 if (error) { 9211 devsw_detach(NULL, &audio_cdevsw); 9212 } 9213 #endif 9214 break; 9215 case MODULE_CMD_FINI: 9216 #ifdef _MODULE 9217 error = config_fini_component(cfdriver_ioconf_audio, 9218 cfattach_ioconf_audio, cfdata_ioconf_audio); 9219 if (error == 0) 9220 devsw_detach(NULL, &audio_cdevsw); 9221 #endif 9222 if (error == 0) 9223 psref_class_destroy(audio_psref_class); 9224 break; 9225 default: 9226 error = ENOTTY; 9227 break; 9228 } 9229 9230 return error; 9231 } 9232
Indexes created Tue Sep 23 07:09:52 GMT 2025