audio.c revision 1.7
1 /* $NetBSD: audio.c,v 1.7 2019/05/13 08:50:25 nakayama Exp $ */ 2 3 /*- 4 * Copyright (c) 2008 The NetBSD Foundation, Inc. 5 * All rights reserved. 6 * 7 * This code is derived from software contributed to The NetBSD Foundation 8 * by Andrew Doran. 9 * 10 * Redistribution and use in source and binary forms, with or without 11 * modification, are permitted provided that the following conditions 12 * are met: 13 * 1. Redistributions of source code must retain the above copyright 14 * notice, this list of conditions and the following disclaimer. 15 * 2. Redistributions in binary form must reproduce the above copyright 16 * notice, this list of conditions and the following disclaimer in the 17 * documentation and/or other materials provided with the distribution. 18 * 19 * THIS SOFTWARE IS PROVIDED BY THE NETBSD FOUNDATION, INC. AND CONTRIBUTORS 20 * ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED 21 * TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR 22 * PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE FOUNDATION OR CONTRIBUTORS 23 * BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR 24 * CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF 25 * SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS 26 * INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN 27 * CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) 28 * ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE 29 * POSSIBILITY OF SUCH DAMAGE. 30 */ 31 32 /* 33 * Copyright (c) 1991-1993 Regents of the University of California. 34 * All rights reserved. 35 * 36 * Redistribution and use in source and binary forms, with or without 37 * modification, are permitted provided that the following conditions 38 * are met: 39 * 1. Redistributions of source code must retain the above copyright 40 * notice, this list of conditions and the following disclaimer. 41 * 2. Redistributions in binary form must reproduce the above copyright 42 * notice, this list of conditions and the following disclaimer in the 43 * documentation and/or other materials provided with the distribution. 44 * 3. All advertising materials mentioning features or use of this software 45 * must display the following acknowledgement: 46 * This product includes software developed by the Computer Systems 47 * Engineering Group at Lawrence Berkeley Laboratory. 48 * 4. Neither the name of the University nor of the Laboratory may be used 49 * to endorse or promote products derived from this software without 50 * specific prior written permission. 51 * 52 * THIS SOFTWARE IS PROVIDED BY THE REGENTS AND CONTRIBUTORS ``AS IS'' AND 53 * ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE 54 * IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE 55 * ARE DISCLAIMED. IN NO EVENT SHALL THE REGENTS OR CONTRIBUTORS BE LIABLE 56 * FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL 57 * DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS 58 * OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) 59 * HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT 60 * LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY 61 * OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF 62 * SUCH DAMAGE. 63 */ 64 65 /* 66 * Locking: there are three locks per device. 67 * 68 * - sc_lock, provided by the underlying driver. This is an adaptive lock, 69 * returned in the second parameter to hw_if->get_locks(). It is known 70 * as the "thread lock". 71 * 72 * It serializes access to state in all places except the 73 * driver's interrupt service routine. This lock is taken from process 74 * context (example: access to /dev/audio). It is also taken from soft 75 * interrupt handlers in this module, primarily to serialize delivery of 76 * wakeups. This lock may be used/provided by modules external to the 77 * audio subsystem, so take care not to introduce a lock order problem. 78 * LONG TERM SLEEPS MUST NOT OCCUR WITH THIS LOCK HELD. 79 * 80 * - sc_intr_lock, provided by the underlying driver. This may be either a 81 * spinlock (at IPL_SCHED or IPL_VM) or an adaptive lock (IPL_NONE or 82 * IPL_SOFT*), returned in the first parameter to hw_if->get_locks(). It 83 * is known as the "interrupt lock". 84 * 85 * It provides atomic access to the device's hardware state, and to audio 86 * channel data that may be accessed by the hardware driver's ISR. 87 * In all places outside the ISR, sc_lock must be held before taking 88 * sc_intr_lock. This is to ensure that groups of hardware operations are 89 * made atomically. SLEEPS CANNOT OCCUR WITH THIS LOCK HELD. 90 * 91 * - sc_exlock, private to this module. This is a variable protected by 92 * sc_lock. It is known as the "critical section". 93 * Some operations release sc_lock in order to allocate memory, to wait 94 * for in-flight I/O to complete, to copy to/from user context, etc. 95 * sc_exlock provides a critical section even under the circumstance. 96 * "+" in following list indicates the interfaces which necessary to be 97 * protected by sc_exlock. 98 * 99 * List of hardware interface methods, and which locks are held when each 100 * is called by this module: 101 * 102 * METHOD INTR THREAD NOTES 103 * ----------------------- ------- ------- ------------------------- 104 * open x x + 105 * close x x + 106 * query_format - x 107 * set_format - x 108 * round_blocksize - x 109 * commit_settings - x 110 * init_output x x 111 * init_input x x 112 * start_output x x + 113 * start_input x x + 114 * halt_output x x + 115 * halt_input x x + 116 * speaker_ctl x x 117 * getdev - x 118 * set_port - x + 119 * get_port - x + 120 * query_devinfo - x 121 * allocm - - + (*1) 122 * freem - - + (*1) 123 * round_buffersize - x 124 * get_props - x 125 * trigger_output x x + 126 * trigger_input x x + 127 * dev_ioctl - x 128 * get_locks - - Called at attach time 129 * 130 * *1 Note: Before 8.0, since these have been called only at attach time, 131 * neither lock were necessary. Currently, on the other hand, since 132 * these may be also called after attach, the thread lock is required. 133 * 134 * In addition, there are two additional locks. 135 * 136 * - file->lock. This is a variable protected by sc_lock and is similar 137 * to the "thread lock". This is one for each file. If any thread 138 * context and software interrupt context who want to access the file 139 * structure, they must acquire this lock before. It protects 140 * descriptor's consistency among multithreaded accesses. Since this 141 * lock uses sc_lock, don't acquire from hardware interrupt context. 142 * 143 * - track->lock. This is an atomic variable and is similar to the 144 * "interrupt lock". This is one for each track. If any thread context 145 * (and software interrupt context) and hardware interrupt context who 146 * want to access some variables on this track, they must acquire this 147 * lock before. It protects track's consistency between hardware 148 * interrupt context and others. 149 */ 150 151 #include <sys/cdefs.h> 152 __KERNEL_RCSID(0, "$NetBSD: audio.c,v 1.7 2019/05/13 08:50:25 nakayama Exp $"); 153 154 #ifdef _KERNEL_OPT 155 #include "audio.h" 156 #include "midi.h" 157 #endif 158 159 #if NAUDIO > 0 160 161 #ifdef _KERNEL 162 163 #include <sys/types.h> 164 #include <sys/param.h> 165 #include <sys/atomic.h> 166 #include <sys/audioio.h> 167 #include <sys/conf.h> 168 #include <sys/cpu.h> 169 #include <sys/device.h> 170 #include <sys/fcntl.h> 171 #include <sys/file.h> 172 #include <sys/filedesc.h> 173 #include <sys/intr.h> 174 #include <sys/ioctl.h> 175 #include <sys/kauth.h> 176 #include <sys/kernel.h> 177 #include <sys/kmem.h> 178 #include <sys/malloc.h> 179 #include <sys/mman.h> 180 #include <sys/module.h> 181 #include <sys/poll.h> 182 #include <sys/proc.h> 183 #include <sys/queue.h> 184 #include <sys/select.h> 185 #include <sys/signalvar.h> 186 #include <sys/stat.h> 187 #include <sys/sysctl.h> 188 #include <sys/systm.h> 189 #include <sys/syslog.h> 190 #include <sys/vnode.h> 191 192 #include <dev/audio/audio_if.h> 193 #include <dev/audio/audiovar.h> 194 #include <dev/audio/audiodef.h> 195 #include <dev/audio/linear.h> 196 #include <dev/audio/mulaw.h> 197 198 #include <machine/endian.h> 199 200 #include <uvm/uvm.h> 201 202 #include "ioconf.h" 203 #endif /* _KERNEL */ 204 205 /* 206 * 0: No debug logs 207 * 1: action changes like open/close/set_format... 208 * 2: + normal operations like read/write/ioctl... 209 * 3: + TRACEs except interrupt 210 * 4: + TRACEs including interrupt 211 */ 212 //#define AUDIO_DEBUG 1 213 214 #if defined(AUDIO_DEBUG) 215 216 int audiodebug = AUDIO_DEBUG; 217 static void audio_vtrace(struct audio_softc *sc, const char *, const char *, 218 const char *, va_list); 219 static void audio_trace(struct audio_softc *sc, const char *, const char *, ...) 220 __printflike(3, 4); 221 static void audio_tracet(const char *, audio_track_t *, const char *, ...) 222 __printflike(3, 4); 223 static void audio_tracef(const char *, audio_file_t *, const char *, ...) 224 __printflike(3, 4); 225 226 /* XXX sloppy memory logger */ 227 static void audio_mlog_init(void); 228 static void audio_mlog_free(void); 229 static void audio_mlog_softintr(void *); 230 extern void audio_mlog_flush(void); 231 extern void audio_mlog_printf(const char *, ...); 232 233 static int mlog_refs; /* reference counter */ 234 static char *mlog_buf[2]; /* double buffer */ 235 static int mlog_buflen; /* buffer length */ 236 static int mlog_used; /* used length */ 237 static int mlog_full; /* number of dropped lines by buffer full */ 238 static int mlog_drop; /* number of dropped lines by busy */ 239 static volatile uint32_t mlog_inuse; /* in-use */ 240 static int mlog_wpage; /* active page */ 241 static void *mlog_sih; /* softint handle */ 242 243 static void 244 audio_mlog_init(void) 245 { 246 mlog_refs++; 247 if (mlog_refs > 1) 248 return; 249 mlog_buflen = 4096; 250 mlog_buf[0] = kmem_zalloc(mlog_buflen, KM_SLEEP); 251 mlog_buf[1] = kmem_zalloc(mlog_buflen, KM_SLEEP); 252 mlog_used = 0; 253 mlog_full = 0; 254 mlog_drop = 0; 255 mlog_inuse = 0; 256 mlog_wpage = 0; 257 mlog_sih = softint_establish(SOFTINT_SERIAL, audio_mlog_softintr, NULL); 258 if (mlog_sih == NULL) 259 printf("%s: softint_establish failed\n", __func__); 260 } 261 262 static void 263 audio_mlog_free(void) 264 { 265 mlog_refs--; 266 if (mlog_refs > 0) 267 return; 268 269 audio_mlog_flush(); 270 if (mlog_sih) 271 softint_disestablish(mlog_sih); 272 kmem_free(mlog_buf[0], mlog_buflen); 273 kmem_free(mlog_buf[1], mlog_buflen); 274 } 275 276 /* 277 * Flush memory buffer. 278 * It must not be called from hardware interrupt context. 279 */ 280 void 281 audio_mlog_flush(void) 282 { 283 if (mlog_refs == 0) 284 return; 285 286 /* Nothing to do if already in use ? */ 287 if (atomic_swap_32(&mlog_inuse, 1) == 1) 288 return; 289 290 int rpage = mlog_wpage; 291 mlog_wpage ^= 1; 292 mlog_buf[mlog_wpage][0] = '\0'; 293 mlog_used = 0; 294 295 atomic_swap_32(&mlog_inuse, 0); 296 297 if (mlog_buf[rpage][0] != '\0') { 298 printf("%s", mlog_buf[rpage]); 299 if (mlog_drop > 0) 300 printf("mlog_drop %d\n", mlog_drop); 301 if (mlog_full > 0) 302 printf("mlog_full %d\n", mlog_full); 303 } 304 mlog_full = 0; 305 mlog_drop = 0; 306 } 307 308 static void 309 audio_mlog_softintr(void *cookie) 310 { 311 audio_mlog_flush(); 312 } 313 314 void 315 audio_mlog_printf(const char *fmt, ...) 316 { 317 int len; 318 va_list ap; 319 320 if (atomic_swap_32(&mlog_inuse, 1) == 1) { 321 /* already inuse */ 322 mlog_drop++; 323 return; 324 } 325 326 va_start(ap, fmt); 327 len = vsnprintf( 328 mlog_buf[mlog_wpage] + mlog_used, 329 mlog_buflen - mlog_used, 330 fmt, ap); 331 va_end(ap); 332 333 mlog_used += len; 334 if (mlog_buflen - mlog_used <= 1) { 335 mlog_full++; 336 } 337 338 atomic_swap_32(&mlog_inuse, 0); 339 340 if (mlog_sih) 341 softint_schedule(mlog_sih); 342 } 343 344 /* trace functions */ 345 static void 346 audio_vtrace(struct audio_softc *sc, const char *funcname, const char *header, 347 const char *fmt, va_list ap) 348 { 349 char buf[256]; 350 int n; 351 352 n = 0; 353 buf[0] = '\0'; 354 n += snprintf(buf + n, sizeof(buf) - n, "%s@%d %s", 355 funcname, device_unit(sc->sc_dev), header); 356 n += vsnprintf(buf + n, sizeof(buf) - n, fmt, ap); 357 358 if (cpu_intr_p()) { 359 audio_mlog_printf("%s\n", buf); 360 } else { 361 audio_mlog_flush(); 362 printf("%s\n", buf); 363 } 364 } 365 366 static void 367 audio_trace(struct audio_softc *sc, const char *funcname, const char *fmt, ...) 368 { 369 va_list ap; 370 371 va_start(ap, fmt); 372 audio_vtrace(sc, funcname, "", fmt, ap); 373 va_end(ap); 374 } 375 376 static void 377 audio_tracet(const char *funcname, audio_track_t *track, const char *fmt, ...) 378 { 379 char hdr[16]; 380 va_list ap; 381 382 snprintf(hdr, sizeof(hdr), "#%d ", track->id); 383 va_start(ap, fmt); 384 audio_vtrace(track->mixer->sc, funcname, hdr, fmt, ap); 385 va_end(ap); 386 } 387 388 static void 389 audio_tracef(const char *funcname, audio_file_t *file, const char *fmt, ...) 390 { 391 char hdr[32]; 392 char phdr[16], rhdr[16]; 393 va_list ap; 394 395 phdr[0] = '\0'; 396 rhdr[0] = '\0'; 397 if (file->ptrack) 398 snprintf(phdr, sizeof(phdr), "#%d", file->ptrack->id); 399 if (file->rtrack) 400 snprintf(rhdr, sizeof(rhdr), "#%d", file->rtrack->id); 401 snprintf(hdr, sizeof(hdr), "{%s,%s} ", phdr, rhdr); 402 403 va_start(ap, fmt); 404 audio_vtrace(file->sc, funcname, hdr, fmt, ap); 405 va_end(ap); 406 } 407 408 #define DPRINTF(n, fmt...) do { \ 409 if (audiodebug >= (n)) { \ 410 audio_mlog_flush(); \ 411 printf(fmt); \ 412 } \ 413 } while (0) 414 #define TRACE(n, fmt...) do { \ 415 if (audiodebug >= (n)) audio_trace(sc, __func__, fmt); \ 416 } while (0) 417 #define TRACET(n, t, fmt...) do { \ 418 if (audiodebug >= (n)) audio_tracet(__func__, t, fmt); \ 419 } while (0) 420 #define TRACEF(n, f, fmt...) do { \ 421 if (audiodebug >= (n)) audio_tracef(__func__, f, fmt); \ 422 } while (0) 423 424 struct audio_track_debugbuf { 425 char usrbuf[32]; 426 char codec[32]; 427 char chvol[32]; 428 char chmix[32]; 429 char freq[32]; 430 char outbuf[32]; 431 }; 432 433 static void 434 audio_track_bufstat(audio_track_t *track, struct audio_track_debugbuf *buf) 435 { 436 437 memset(buf, 0, sizeof(*buf)); 438 439 snprintf(buf->outbuf, sizeof(buf->outbuf), " out=%d/%d/%d", 440 track->outbuf.head, track->outbuf.used, track->outbuf.capacity); 441 if (track->freq.filter) 442 snprintf(buf->freq, sizeof(buf->freq), " f=%d/%d/%d", 443 track->freq.srcbuf.head, 444 track->freq.srcbuf.used, 445 track->freq.srcbuf.capacity); 446 if (track->chmix.filter) 447 snprintf(buf->chmix, sizeof(buf->chmix), " m=%d", 448 track->chmix.srcbuf.used); 449 if (track->chvol.filter) 450 snprintf(buf->chvol, sizeof(buf->chvol), " v=%d", 451 track->chvol.srcbuf.used); 452 if (track->codec.filter) 453 snprintf(buf->codec, sizeof(buf->codec), " e=%d", 454 track->codec.srcbuf.used); 455 snprintf(buf->usrbuf, sizeof(buf->usrbuf), " usr=%d/%d/H%d", 456 track->usrbuf.head, track->usrbuf.used, track->usrbuf_usedhigh); 457 } 458 #else 459 #define DPRINTF(n, fmt...) do { } while (0) 460 #define TRACE(n, fmt, ...) do { } while (0) 461 #define TRACET(n, t, fmt, ...) do { } while (0) 462 #define TRACEF(n, f, fmt, ...) do { } while (0) 463 #endif 464 465 #define SPECIFIED(x) ((x) != ~0) 466 #define SPECIFIED_CH(x) ((x) != (u_char)~0) 467 468 /* Device timeout in msec */ 469 #define AUDIO_TIMEOUT (3000) 470 471 /* #define AUDIO_PM_IDLE */ 472 #ifdef AUDIO_PM_IDLE 473 int audio_idle_timeout = 30; 474 #endif 475 476 struct portname { 477 const char *name; 478 int mask; 479 }; 480 481 static int audiomatch(device_t, cfdata_t, void *); 482 static void audioattach(device_t, device_t, void *); 483 static int audiodetach(device_t, int); 484 static int audioactivate(device_t, enum devact); 485 static void audiochilddet(device_t, device_t); 486 static int audiorescan(device_t, const char *, const int *); 487 488 static int audio_modcmd(modcmd_t, void *); 489 490 #ifdef AUDIO_PM_IDLE 491 static void audio_idle(void *); 492 static void audio_activity(device_t, devactive_t); 493 #endif 494 495 static bool audio_suspend(device_t dv, const pmf_qual_t *); 496 static bool audio_resume(device_t dv, const pmf_qual_t *); 497 static void audio_volume_down(device_t); 498 static void audio_volume_up(device_t); 499 static void audio_volume_toggle(device_t); 500 501 static void audio_mixer_capture(struct audio_softc *); 502 static void audio_mixer_restore(struct audio_softc *); 503 504 static void audio_softintr_rd(void *); 505 static void audio_softintr_wr(void *); 506 507 static int audio_enter_exclusive(struct audio_softc *); 508 static void audio_exit_exclusive(struct audio_softc *); 509 static int audio_track_waitio(struct audio_softc *, audio_track_t *); 510 static int audio_file_acquire(struct audio_softc *, audio_file_t *); 511 static void audio_file_release(struct audio_softc *, audio_file_t *); 512 513 static int audioclose(struct file *); 514 static int audioread(struct file *, off_t *, struct uio *, kauth_cred_t, int); 515 static int audiowrite(struct file *, off_t *, struct uio *, kauth_cred_t, int); 516 static int audioioctl(struct file *, u_long, void *); 517 static int audiopoll(struct file *, int); 518 static int audiokqfilter(struct file *, struct knote *); 519 static int audiommap(struct file *, off_t *, size_t, int, int *, int *, 520 struct uvm_object **, int *); 521 static int audiostat(struct file *, struct stat *); 522 523 static void filt_audiowrite_detach(struct knote *); 524 static int filt_audiowrite_event(struct knote *, long); 525 static void filt_audioread_detach(struct knote *); 526 static int filt_audioread_event(struct knote *, long); 527 528 static int audio_open(dev_t, struct audio_softc *, int, int, struct lwp *, 529 struct audiobell_arg *); 530 static int audio_close(struct audio_softc *, audio_file_t *); 531 static int audio_read(struct audio_softc *, struct uio *, int, audio_file_t *); 532 static int audio_write(struct audio_softc *, struct uio *, int, audio_file_t *); 533 static void audio_file_clear(struct audio_softc *, audio_file_t *); 534 static int audio_ioctl(dev_t, struct audio_softc *, u_long, void *, int, 535 struct lwp *, audio_file_t *); 536 static int audio_poll(struct audio_softc *, int, struct lwp *, audio_file_t *); 537 static int audio_kqfilter(struct audio_softc *, audio_file_t *, struct knote *); 538 static int audio_mmap(struct audio_softc *, off_t *, size_t, int, int *, int *, 539 struct uvm_object **, int *, audio_file_t *); 540 541 static int audioctl_open(dev_t, struct audio_softc *, int, int, struct lwp *); 542 543 static void audio_pintr(void *); 544 static void audio_rintr(void *); 545 546 static int audio_query_devinfo(struct audio_softc *, mixer_devinfo_t *); 547 548 static __inline int audio_track_readablebytes(const audio_track_t *); 549 static int audio_file_setinfo(struct audio_softc *, audio_file_t *, 550 const struct audio_info *); 551 static int audio_track_setinfo_check(audio_format2_t *, 552 const struct audio_prinfo *); 553 static void audio_track_setinfo_water(audio_track_t *, 554 const struct audio_info *); 555 static int audio_hw_setinfo(struct audio_softc *, const struct audio_info *, 556 struct audio_info *); 557 static int audio_hw_set_format(struct audio_softc *, int, 558 audio_format2_t *, audio_format2_t *, 559 audio_filter_reg_t *, audio_filter_reg_t *); 560 static int audiogetinfo(struct audio_softc *, struct audio_info *, int, 561 audio_file_t *); 562 static int audio_get_props(struct audio_softc *); 563 static bool audio_can_playback(struct audio_softc *); 564 static bool audio_can_capture(struct audio_softc *); 565 static int audio_check_params(audio_format2_t *); 566 static int audio_mixers_init(struct audio_softc *sc, int, 567 const audio_format2_t *, const audio_format2_t *, 568 const audio_filter_reg_t *, const audio_filter_reg_t *); 569 static int audio_select_freq(const struct audio_format *); 570 static int audio_hw_probe(struct audio_softc *, int, int *, 571 audio_format2_t *, audio_format2_t *); 572 static int audio_hw_probe_fmt(struct audio_softc *, audio_format2_t *, int); 573 static int audio_hw_validate_format(struct audio_softc *, int, 574 const audio_format2_t *); 575 static int audio_mixers_set_format(struct audio_softc *, 576 const struct audio_info *); 577 static void audio_mixers_get_format(struct audio_softc *, struct audio_info *); 578 static int audio_sysctl_volume(SYSCTLFN_PROTO); 579 static int audio_sysctl_blk_ms(SYSCTLFN_PROTO); 580 static int audio_sysctl_multiuser(SYSCTLFN_PROTO); 581 #if defined(AUDIO_DEBUG) 582 static int audio_sysctl_debug(SYSCTLFN_PROTO); 583 #endif 584 #if defined(DIAGNOSTIC) || defined(AUDIO_DEBUG) 585 static void audio_format2_tostr(char *, size_t, const audio_format2_t *); 586 #endif 587 #if defined(AUDIO_DEBUG) 588 static void audio_print_format2(const char *, const audio_format2_t *) __unused; 589 #endif 590 591 static void *audio_realloc(void *, size_t); 592 static int audio_realloc_usrbuf(audio_track_t *, int); 593 static void audio_free_usrbuf(audio_track_t *); 594 595 static audio_track_t *audio_track_create(struct audio_softc *, 596 audio_trackmixer_t *); 597 static void audio_track_destroy(audio_track_t *); 598 static audio_filter_t audio_track_get_codec(audio_track_t *, 599 const audio_format2_t *, const audio_format2_t *); 600 static int audio_track_set_format(audio_track_t *, audio_format2_t *); 601 static void audio_track_play(audio_track_t *); 602 static int audio_track_drain(struct audio_softc *, audio_track_t *); 603 static void audio_track_record(audio_track_t *); 604 static void audio_track_clear(struct audio_softc *, audio_track_t *); 605 606 static int audio_mixer_init(struct audio_softc *, int, 607 const audio_format2_t *, const audio_filter_reg_t *); 608 static void audio_mixer_destroy(struct audio_softc *, audio_trackmixer_t *); 609 static void audio_pmixer_start(struct audio_softc *, bool); 610 static void audio_pmixer_process(struct audio_softc *); 611 static int audio_pmixer_mix_track(audio_trackmixer_t *, audio_track_t *, int); 612 static void audio_pmixer_output(struct audio_softc *); 613 static int audio_pmixer_halt(struct audio_softc *); 614 static void audio_rmixer_start(struct audio_softc *); 615 static void audio_rmixer_process(struct audio_softc *); 616 static void audio_rmixer_input(struct audio_softc *); 617 static int audio_rmixer_halt(struct audio_softc *); 618 619 static void mixer_init(struct audio_softc *); 620 static int mixer_open(dev_t, struct audio_softc *, int, int, struct lwp *); 621 static int mixer_close(struct audio_softc *, audio_file_t *); 622 static int mixer_ioctl(struct audio_softc *, u_long, void *, int, struct lwp *); 623 static void mixer_remove(struct audio_softc *); 624 static void mixer_signal(struct audio_softc *); 625 626 static int au_portof(struct audio_softc *, char *, int); 627 628 static void au_setup_ports(struct audio_softc *, struct au_mixer_ports *, 629 mixer_devinfo_t *, const struct portname *); 630 static int au_set_lr_value(struct audio_softc *, mixer_ctrl_t *, int, int); 631 static int au_get_lr_value(struct audio_softc *, mixer_ctrl_t *, int *, int *); 632 static int au_set_gain(struct audio_softc *, struct au_mixer_ports *, int, int); 633 static void au_get_gain(struct audio_softc *, struct au_mixer_ports *, 634 u_int *, u_char *); 635 static int au_set_port(struct audio_softc *, struct au_mixer_ports *, u_int); 636 static int au_get_port(struct audio_softc *, struct au_mixer_ports *); 637 static int au_set_monitor_gain(struct audio_softc *, int); 638 static int au_get_monitor_gain(struct audio_softc *); 639 static int audio_get_port(struct audio_softc *, mixer_ctrl_t *); 640 static int audio_set_port(struct audio_softc *, mixer_ctrl_t *); 641 642 static __inline struct audio_params 643 format2_to_params(const audio_format2_t *f2) 644 { 645 audio_params_t p; 646 647 /* validbits/precision <-> precision/stride */ 648 p.sample_rate = f2->sample_rate; 649 p.channels = f2->channels; 650 p.encoding = f2->encoding; 651 p.validbits = f2->precision; 652 p.precision = f2->stride; 653 return p; 654 } 655 656 static __inline audio_format2_t 657 params_to_format2(const struct audio_params *p) 658 { 659 audio_format2_t f2; 660 661 /* precision/stride <-> validbits/precision */ 662 f2.sample_rate = p->sample_rate; 663 f2.channels = p->channels; 664 f2.encoding = p->encoding; 665 f2.precision = p->validbits; 666 f2.stride = p->precision; 667 return f2; 668 } 669 670 /* Return true if this track is a playback track. */ 671 static __inline bool 672 audio_track_is_playback(const audio_track_t *track) 673 { 674 675 return ((track->mode & AUMODE_PLAY) != 0); 676 } 677 678 /* Return true if this track is a recording track. */ 679 static __inline bool 680 audio_track_is_record(const audio_track_t *track) 681 { 682 683 return ((track->mode & AUMODE_RECORD) != 0); 684 } 685 686 #if 0 /* XXX Not used yet */ 687 /* 688 * Convert 0..255 volume used in userland to internal presentation 0..256. 689 */ 690 static __inline u_int 691 audio_volume_to_inner(u_int v) 692 { 693 694 return v < 127 ? v : v + 1; 695 } 696 697 /* 698 * Convert 0..256 internal presentation to 0..255 volume used in userland. 699 */ 700 static __inline u_int 701 audio_volume_to_outer(u_int v) 702 { 703 704 return v < 127 ? v : v - 1; 705 } 706 #endif /* 0 */ 707 708 static dev_type_open(audioopen); 709 /* XXXMRG use more dev_type_xxx */ 710 711 const struct cdevsw audio_cdevsw = { 712 .d_open = audioopen, 713 .d_close = noclose, 714 .d_read = noread, 715 .d_write = nowrite, 716 .d_ioctl = noioctl, 717 .d_stop = nostop, 718 .d_tty = notty, 719 .d_poll = nopoll, 720 .d_mmap = nommap, 721 .d_kqfilter = nokqfilter, 722 .d_discard = nodiscard, 723 .d_flag = D_OTHER | D_MPSAFE 724 }; 725 726 const struct fileops audio_fileops = { 727 .fo_name = "audio", 728 .fo_read = audioread, 729 .fo_write = audiowrite, 730 .fo_ioctl = audioioctl, 731 .fo_fcntl = fnullop_fcntl, 732 .fo_stat = audiostat, 733 .fo_poll = audiopoll, 734 .fo_close = audioclose, 735 .fo_mmap = audiommap, 736 .fo_kqfilter = audiokqfilter, 737 .fo_restart = fnullop_restart 738 }; 739 740 /* The default audio mode: 8 kHz mono mu-law */ 741 static const struct audio_params audio_default = { 742 .sample_rate = 8000, 743 .encoding = AUDIO_ENCODING_ULAW, 744 .precision = 8, 745 .validbits = 8, 746 .channels = 1, 747 }; 748 749 static const char *encoding_names[] = { 750 "none", 751 AudioEmulaw, 752 AudioEalaw, 753 "pcm16", 754 "pcm8", 755 AudioEadpcm, 756 AudioEslinear_le, 757 AudioEslinear_be, 758 AudioEulinear_le, 759 AudioEulinear_be, 760 AudioEslinear, 761 AudioEulinear, 762 AudioEmpeg_l1_stream, 763 AudioEmpeg_l1_packets, 764 AudioEmpeg_l1_system, 765 AudioEmpeg_l2_stream, 766 AudioEmpeg_l2_packets, 767 AudioEmpeg_l2_system, 768 AudioEac3, 769 }; 770 771 /* 772 * Returns encoding name corresponding to AUDIO_ENCODING_*. 773 * Note that it may return a local buffer because it is mainly for debugging. 774 */ 775 const char * 776 audio_encoding_name(int encoding) 777 { 778 static char buf[16]; 779 780 if (0 <= encoding && encoding < __arraycount(encoding_names)) { 781 return encoding_names[encoding]; 782 } else { 783 snprintf(buf, sizeof(buf), "enc=%d", encoding); 784 return buf; 785 } 786 } 787 788 /* 789 * Supported encodings used by AUDIO_GETENC. 790 * index and flags are set by code. 791 * XXX is there any needs for SLINEAR_OE:>=16/ULINEAR_OE:>=16 ? 792 */ 793 static const audio_encoding_t audio_encodings[] = { 794 { 0, AudioEmulaw, AUDIO_ENCODING_ULAW, 8, 0 }, 795 { 0, AudioEalaw, AUDIO_ENCODING_ALAW, 8, 0 }, 796 { 0, AudioEslinear, AUDIO_ENCODING_SLINEAR, 8, 0 }, 797 { 0, AudioEulinear, AUDIO_ENCODING_ULINEAR, 8, 0 }, 798 { 0, AudioEslinear_le, AUDIO_ENCODING_SLINEAR_LE, 16, 0 }, 799 { 0, AudioEulinear_le, AUDIO_ENCODING_ULINEAR_LE, 16, 0 }, 800 { 0, AudioEslinear_be, AUDIO_ENCODING_SLINEAR_BE, 16, 0 }, 801 { 0, AudioEulinear_be, AUDIO_ENCODING_ULINEAR_BE, 16, 0 }, 802 #if defined(AUDIO_SUPPORT_LINEAR24) 803 { 0, AudioEslinear_le, AUDIO_ENCODING_SLINEAR_LE, 24, 0 }, 804 { 0, AudioEulinear_le, AUDIO_ENCODING_ULINEAR_LE, 24, 0 }, 805 { 0, AudioEslinear_be, AUDIO_ENCODING_SLINEAR_BE, 24, 0 }, 806 { 0, AudioEulinear_be, AUDIO_ENCODING_ULINEAR_BE, 24, 0 }, 807 #endif 808 { 0, AudioEslinear_le, AUDIO_ENCODING_SLINEAR_LE, 32, 0 }, 809 { 0, AudioEulinear_le, AUDIO_ENCODING_ULINEAR_LE, 32, 0 }, 810 { 0, AudioEslinear_be, AUDIO_ENCODING_SLINEAR_BE, 32, 0 }, 811 { 0, AudioEulinear_be, AUDIO_ENCODING_ULINEAR_BE, 32, 0 }, 812 }; 813 814 static const struct portname itable[] = { 815 { AudioNmicrophone, AUDIO_MICROPHONE }, 816 { AudioNline, AUDIO_LINE_IN }, 817 { AudioNcd, AUDIO_CD }, 818 { 0, 0 } 819 }; 820 static const struct portname otable[] = { 821 { AudioNspeaker, AUDIO_SPEAKER }, 822 { AudioNheadphone, AUDIO_HEADPHONE }, 823 { AudioNline, AUDIO_LINE_OUT }, 824 { 0, 0 } 825 }; 826 827 CFATTACH_DECL3_NEW(audio, sizeof(struct audio_softc), 828 audiomatch, audioattach, audiodetach, audioactivate, audiorescan, 829 audiochilddet, DVF_DETACH_SHUTDOWN); 830 831 static int 832 audiomatch(device_t parent, cfdata_t match, void *aux) 833 { 834 struct audio_attach_args *sa; 835 836 sa = aux; 837 DPRINTF(1, "%s: type=%d sa=%p hw=%p\n", 838 __func__, sa->type, sa, sa->hwif); 839 return (sa->type == AUDIODEV_TYPE_AUDIO) ? 1 : 0; 840 } 841 842 static void 843 audioattach(device_t parent, device_t self, void *aux) 844 { 845 struct audio_softc *sc; 846 struct audio_attach_args *sa; 847 const struct audio_hw_if *hw_if; 848 audio_format2_t phwfmt; 849 audio_format2_t rhwfmt; 850 audio_filter_reg_t pfil; 851 audio_filter_reg_t rfil; 852 const struct sysctlnode *node; 853 void *hdlp; 854 bool is_indep; 855 int mode; 856 int props; 857 int error; 858 859 sc = device_private(self); 860 sc->sc_dev = self; 861 sa = (struct audio_attach_args *)aux; 862 hw_if = sa->hwif; 863 hdlp = sa->hdl; 864 865 if (hw_if == NULL || hw_if->get_locks == NULL) { 866 panic("audioattach: missing hw_if method"); 867 } 868 869 hw_if->get_locks(hdlp, &sc->sc_intr_lock, &sc->sc_lock); 870 871 #ifdef DIAGNOSTIC 872 if (hw_if->query_format == NULL || 873 hw_if->set_format == NULL || 874 (hw_if->start_output == NULL && hw_if->trigger_output == NULL) || 875 (hw_if->start_input == NULL && hw_if->trigger_input == NULL) || 876 hw_if->halt_output == NULL || 877 hw_if->halt_input == NULL || 878 hw_if->getdev == NULL || 879 hw_if->set_port == NULL || 880 hw_if->get_port == NULL || 881 hw_if->query_devinfo == NULL || 882 hw_if->get_props == NULL) { 883 aprint_error(": missing method\n"); 884 return; 885 } 886 #endif 887 888 sc->hw_if = hw_if; 889 sc->hw_hdl = hdlp; 890 sc->hw_dev = parent; 891 892 sc->sc_blk_ms = AUDIO_BLK_MS; 893 SLIST_INIT(&sc->sc_files); 894 cv_init(&sc->sc_exlockcv, "audiolk"); 895 896 mutex_enter(sc->sc_lock); 897 props = audio_get_props(sc); 898 mutex_exit(sc->sc_lock); 899 900 if ((props & AUDIO_PROP_FULLDUPLEX)) 901 aprint_normal(": full duplex"); 902 else 903 aprint_normal(": half duplex"); 904 905 is_indep = (props & AUDIO_PROP_INDEPENDENT); 906 mode = 0; 907 if ((props & AUDIO_PROP_PLAYBACK)) { 908 mode |= AUMODE_PLAY; 909 aprint_normal(", playback"); 910 } 911 if ((props & AUDIO_PROP_CAPTURE)) { 912 mode |= AUMODE_RECORD; 913 aprint_normal(", capture"); 914 } 915 if ((props & AUDIO_PROP_MMAP) != 0) 916 aprint_normal(", mmap"); 917 if (is_indep) 918 aprint_normal(", independent"); 919 920 aprint_naive("\n"); 921 aprint_normal("\n"); 922 923 KASSERT((mode & (AUMODE_PLAY | AUMODE_RECORD)) != 0); 924 925 /* probe hw params */ 926 memset(&phwfmt, 0, sizeof(phwfmt)); 927 memset(&rhwfmt, 0, sizeof(rhwfmt)); 928 memset(&pfil, 0, sizeof(pfil)); 929 memset(&rfil, 0, sizeof(rfil)); 930 mutex_enter(sc->sc_lock); 931 error = audio_hw_probe(sc, is_indep, &mode, &phwfmt, &rhwfmt); 932 if (error) { 933 mutex_exit(sc->sc_lock); 934 aprint_error_dev(self, "audio_hw_probe failed, " 935 "error = %d\n", error); 936 goto bad; 937 } 938 if (mode == 0) { 939 mutex_exit(sc->sc_lock); 940 aprint_error_dev(self, "audio_hw_probe failed, no mode\n"); 941 goto bad; 942 } 943 /* Init hardware. */ 944 /* hw_probe() also validates [pr]hwfmt. */ 945 error = audio_hw_set_format(sc, mode, &phwfmt, &rhwfmt, &pfil, &rfil); 946 if (error) { 947 mutex_exit(sc->sc_lock); 948 aprint_error_dev(self, "audio_hw_set_format failed, " 949 "error = %d\n", error); 950 goto bad; 951 } 952 953 /* 954 * Init track mixers. If at least one direction is available on 955 * attach time, we assume a success. 956 */ 957 error = audio_mixers_init(sc, mode, &phwfmt, &rhwfmt, &pfil, &rfil); 958 mutex_exit(sc->sc_lock); 959 if (sc->sc_pmixer == NULL && sc->sc_rmixer == NULL) { 960 aprint_error_dev(self, "audio_mixers_init failed, " 961 "error = %d\n", error); 962 goto bad; 963 } 964 965 selinit(&sc->sc_wsel); 966 selinit(&sc->sc_rsel); 967 968 /* Initial parameter of /dev/sound */ 969 sc->sc_sound_pparams = params_to_format2(&audio_default); 970 sc->sc_sound_rparams = params_to_format2(&audio_default); 971 sc->sc_sound_ppause = false; 972 sc->sc_sound_rpause = false; 973 974 /* XXX TODO: consider about sc_ai */ 975 976 mixer_init(sc); 977 TRACE(2, "inputs ports=0x%x, input master=%d, " 978 "output ports=0x%x, output master=%d", 979 sc->sc_inports.allports, sc->sc_inports.master, 980 sc->sc_outports.allports, sc->sc_outports.master); 981 982 sysctl_createv(&sc->sc_log, 0, NULL, &node, 983 0, 984 CTLTYPE_NODE, device_xname(sc->sc_dev), 985 SYSCTL_DESCR("audio test"), 986 NULL, 0, 987 NULL, 0, 988 CTL_HW, 989 CTL_CREATE, CTL_EOL); 990 991 if (node != NULL) { 992 sysctl_createv(&sc->sc_log, 0, NULL, NULL, 993 CTLFLAG_READWRITE, 994 CTLTYPE_INT, "volume", 995 SYSCTL_DESCR("software volume test"), 996 audio_sysctl_volume, 0, (void *)sc, 0, 997 CTL_HW, node->sysctl_num, CTL_CREATE, CTL_EOL); 998 999 sysctl_createv(&sc->sc_log, 0, NULL, NULL, 1000 CTLFLAG_READWRITE, 1001 CTLTYPE_INT, "blk_ms", 1002 SYSCTL_DESCR("blocksize in msec"), 1003 audio_sysctl_blk_ms, 0, (void *)sc, 0, 1004 CTL_HW, node->sysctl_num, CTL_CREATE, CTL_EOL); 1005 1006 sysctl_createv(&sc->sc_log, 0, NULL, NULL, 1007 CTLFLAG_READWRITE, 1008 CTLTYPE_BOOL, "multiuser", 1009 SYSCTL_DESCR("allow multiple user access"), 1010 audio_sysctl_multiuser, 0, (void *)sc, 0, 1011 CTL_HW, node->sysctl_num, CTL_CREATE, CTL_EOL); 1012 1013 #if defined(AUDIO_DEBUG) 1014 sysctl_createv(&sc->sc_log, 0, NULL, NULL, 1015 CTLFLAG_READWRITE, 1016 CTLTYPE_INT, "debug", 1017 SYSCTL_DESCR("debug level (0..4)"), 1018 audio_sysctl_debug, 0, (void *)sc, 0, 1019 CTL_HW, node->sysctl_num, CTL_CREATE, CTL_EOL); 1020 #endif 1021 } 1022 1023 #ifdef AUDIO_PM_IDLE 1024 callout_init(&sc->sc_idle_counter, 0); 1025 callout_setfunc(&sc->sc_idle_counter, audio_idle, self); 1026 #endif 1027 1028 if (!pmf_device_register(self, audio_suspend, audio_resume)) 1029 aprint_error_dev(self, "couldn't establish power handler\n"); 1030 #ifdef AUDIO_PM_IDLE 1031 if (!device_active_register(self, audio_activity)) 1032 aprint_error_dev(self, "couldn't register activity handler\n"); 1033 #endif 1034 1035 if (!pmf_event_register(self, PMFE_AUDIO_VOLUME_DOWN, 1036 audio_volume_down, true)) 1037 aprint_error_dev(self, "couldn't add volume down handler\n"); 1038 if (!pmf_event_register(self, PMFE_AUDIO_VOLUME_UP, 1039 audio_volume_up, true)) 1040 aprint_error_dev(self, "couldn't add volume up handler\n"); 1041 if (!pmf_event_register(self, PMFE_AUDIO_VOLUME_TOGGLE, 1042 audio_volume_toggle, true)) 1043 aprint_error_dev(self, "couldn't add volume toggle handler\n"); 1044 1045 #ifdef AUDIO_PM_IDLE 1046 callout_schedule(&sc->sc_idle_counter, audio_idle_timeout * hz); 1047 #endif 1048 1049 #if defined(AUDIO_DEBUG) 1050 audio_mlog_init(); 1051 #endif 1052 1053 audiorescan(self, "audio", NULL); 1054 return; 1055 1056 bad: 1057 /* Clearing hw_if means that device is attached but disabled. */ 1058 sc->hw_if = NULL; 1059 aprint_error_dev(sc->sc_dev, "disabled\n"); 1060 return; 1061 } 1062 1063 /* 1064 * Initialize hardware mixer. 1065 * This function is called from audioattach(). 1066 */ 1067 static void 1068 mixer_init(struct audio_softc *sc) 1069 { 1070 mixer_devinfo_t mi; 1071 int iclass, mclass, oclass, rclass; 1072 int record_master_found, record_source_found; 1073 1074 iclass = mclass = oclass = rclass = -1; 1075 sc->sc_inports.index = -1; 1076 sc->sc_inports.master = -1; 1077 sc->sc_inports.nports = 0; 1078 sc->sc_inports.isenum = false; 1079 sc->sc_inports.allports = 0; 1080 sc->sc_inports.isdual = false; 1081 sc->sc_inports.mixerout = -1; 1082 sc->sc_inports.cur_port = -1; 1083 sc->sc_outports.index = -1; 1084 sc->sc_outports.master = -1; 1085 sc->sc_outports.nports = 0; 1086 sc->sc_outports.isenum = false; 1087 sc->sc_outports.allports = 0; 1088 sc->sc_outports.isdual = false; 1089 sc->sc_outports.mixerout = -1; 1090 sc->sc_outports.cur_port = -1; 1091 sc->sc_monitor_port = -1; 1092 /* 1093 * Read through the underlying driver's list, picking out the class 1094 * names from the mixer descriptions. We'll need them to decode the 1095 * mixer descriptions on the next pass through the loop. 1096 */ 1097 mutex_enter(sc->sc_lock); 1098 for(mi.index = 0; ; mi.index++) { 1099 if (audio_query_devinfo(sc, &mi) != 0) 1100 break; 1101 /* 1102 * The type of AUDIO_MIXER_CLASS merely introduces a class. 1103 * All the other types describe an actual mixer. 1104 */ 1105 if (mi.type == AUDIO_MIXER_CLASS) { 1106 if (strcmp(mi.label.name, AudioCinputs) == 0) 1107 iclass = mi.mixer_class; 1108 if (strcmp(mi.label.name, AudioCmonitor) == 0) 1109 mclass = mi.mixer_class; 1110 if (strcmp(mi.label.name, AudioCoutputs) == 0) 1111 oclass = mi.mixer_class; 1112 if (strcmp(mi.label.name, AudioCrecord) == 0) 1113 rclass = mi.mixer_class; 1114 } 1115 } 1116 mutex_exit(sc->sc_lock); 1117 1118 /* Allocate save area. Ensure non-zero allocation. */ 1119 sc->sc_nmixer_states = mi.index; 1120 sc->sc_mixer_state = kmem_zalloc(sizeof(mixer_ctrl_t) * 1121 (sc->sc_nmixer_states + 1), KM_SLEEP); 1122 1123 /* 1124 * This is where we assign each control in the "audio" model, to the 1125 * underlying "mixer" control. We walk through the whole list once, 1126 * assigning likely candidates as we come across them. 1127 */ 1128 record_master_found = 0; 1129 record_source_found = 0; 1130 mutex_enter(sc->sc_lock); 1131 for(mi.index = 0; ; mi.index++) { 1132 if (audio_query_devinfo(sc, &mi) != 0) 1133 break; 1134 KASSERT(mi.index < sc->sc_nmixer_states); 1135 if (mi.type == AUDIO_MIXER_CLASS) 1136 continue; 1137 if (mi.mixer_class == iclass) { 1138 /* 1139 * AudioCinputs is only a fallback, when we don't 1140 * find what we're looking for in AudioCrecord, so 1141 * check the flags before accepting one of these. 1142 */ 1143 if (strcmp(mi.label.name, AudioNmaster) == 0 1144 && record_master_found == 0) 1145 sc->sc_inports.master = mi.index; 1146 if (strcmp(mi.label.name, AudioNsource) == 0 1147 && record_source_found == 0) { 1148 if (mi.type == AUDIO_MIXER_ENUM) { 1149 int i; 1150 for(i = 0; i < mi.un.e.num_mem; i++) 1151 if (strcmp(mi.un.e.member[i].label.name, 1152 AudioNmixerout) == 0) 1153 sc->sc_inports.mixerout = 1154 mi.un.e.member[i].ord; 1155 } 1156 au_setup_ports(sc, &sc->sc_inports, &mi, 1157 itable); 1158 } 1159 if (strcmp(mi.label.name, AudioNdac) == 0 && 1160 sc->sc_outports.master == -1) 1161 sc->sc_outports.master = mi.index; 1162 } else if (mi.mixer_class == mclass) { 1163 if (strcmp(mi.label.name, AudioNmonitor) == 0) 1164 sc->sc_monitor_port = mi.index; 1165 } else if (mi.mixer_class == oclass) { 1166 if (strcmp(mi.label.name, AudioNmaster) == 0) 1167 sc->sc_outports.master = mi.index; 1168 if (strcmp(mi.label.name, AudioNselect) == 0) 1169 au_setup_ports(sc, &sc->sc_outports, &mi, 1170 otable); 1171 } else if (mi.mixer_class == rclass) { 1172 /* 1173 * These are the preferred mixers for the audio record 1174 * controls, so set the flags here, but don't check. 1175 */ 1176 if (strcmp(mi.label.name, AudioNmaster) == 0) { 1177 sc->sc_inports.master = mi.index; 1178 record_master_found = 1; 1179 } 1180 #if 1 /* Deprecated. Use AudioNmaster. */ 1181 if (strcmp(mi.label.name, AudioNrecord) == 0) { 1182 sc->sc_inports.master = mi.index; 1183 record_master_found = 1; 1184 } 1185 if (strcmp(mi.label.name, AudioNvolume) == 0) { 1186 sc->sc_inports.master = mi.index; 1187 record_master_found = 1; 1188 } 1189 #endif 1190 if (strcmp(mi.label.name, AudioNsource) == 0) { 1191 if (mi.type == AUDIO_MIXER_ENUM) { 1192 int i; 1193 for(i = 0; i < mi.un.e.num_mem; i++) 1194 if (strcmp(mi.un.e.member[i].label.name, 1195 AudioNmixerout) == 0) 1196 sc->sc_inports.mixerout = 1197 mi.un.e.member[i].ord; 1198 } 1199 au_setup_ports(sc, &sc->sc_inports, &mi, 1200 itable); 1201 record_source_found = 1; 1202 } 1203 } 1204 } 1205 mutex_exit(sc->sc_lock); 1206 } 1207 1208 static int 1209 audioactivate(device_t self, enum devact act) 1210 { 1211 struct audio_softc *sc = device_private(self); 1212 1213 switch (act) { 1214 case DVACT_DEACTIVATE: 1215 mutex_enter(sc->sc_lock); 1216 sc->sc_dying = true; 1217 cv_broadcast(&sc->sc_exlockcv); 1218 mutex_exit(sc->sc_lock); 1219 return 0; 1220 default: 1221 return EOPNOTSUPP; 1222 } 1223 } 1224 1225 static int 1226 audiodetach(device_t self, int flags) 1227 { 1228 struct audio_softc *sc; 1229 int maj, mn; 1230 int error; 1231 1232 sc = device_private(self); 1233 TRACE(2, "flags=%d", flags); 1234 1235 /* device is not initialized */ 1236 if (sc->hw_if == NULL) 1237 return 0; 1238 1239 /* Start draining existing accessors of the device. */ 1240 error = config_detach_children(self, flags); 1241 if (error) 1242 return error; 1243 1244 mutex_enter(sc->sc_lock); 1245 sc->sc_dying = true; 1246 cv_broadcast(&sc->sc_exlockcv); 1247 if (sc->sc_pmixer) 1248 cv_broadcast(&sc->sc_pmixer->outcv); 1249 if (sc->sc_rmixer) 1250 cv_broadcast(&sc->sc_rmixer->outcv); 1251 mutex_exit(sc->sc_lock); 1252 1253 /* locate the major number */ 1254 maj = cdevsw_lookup_major(&audio_cdevsw); 1255 1256 /* 1257 * Nuke the vnodes for any open instances (calls close). 1258 * Will wait until any activity on the device nodes has ceased. 1259 */ 1260 mn = device_unit(self); 1261 vdevgone(maj, mn | SOUND_DEVICE, mn | SOUND_DEVICE, VCHR); 1262 vdevgone(maj, mn | AUDIO_DEVICE, mn | AUDIO_DEVICE, VCHR); 1263 vdevgone(maj, mn | AUDIOCTL_DEVICE, mn | AUDIOCTL_DEVICE, VCHR); 1264 vdevgone(maj, mn | MIXER_DEVICE, mn | MIXER_DEVICE, VCHR); 1265 1266 pmf_event_deregister(self, PMFE_AUDIO_VOLUME_DOWN, 1267 audio_volume_down, true); 1268 pmf_event_deregister(self, PMFE_AUDIO_VOLUME_UP, 1269 audio_volume_up, true); 1270 pmf_event_deregister(self, PMFE_AUDIO_VOLUME_TOGGLE, 1271 audio_volume_toggle, true); 1272 1273 #ifdef AUDIO_PM_IDLE 1274 callout_halt(&sc->sc_idle_counter, sc->sc_lock); 1275 1276 device_active_deregister(self, audio_activity); 1277 #endif 1278 1279 pmf_device_deregister(self); 1280 1281 /* Free resources */ 1282 mutex_enter(sc->sc_lock); 1283 if (sc->sc_pmixer) { 1284 audio_mixer_destroy(sc, sc->sc_pmixer); 1285 kmem_free(sc->sc_pmixer, sizeof(*sc->sc_pmixer)); 1286 } 1287 if (sc->sc_rmixer) { 1288 audio_mixer_destroy(sc, sc->sc_rmixer); 1289 kmem_free(sc->sc_rmixer, sizeof(*sc->sc_rmixer)); 1290 } 1291 mutex_exit(sc->sc_lock); 1292 1293 seldestroy(&sc->sc_wsel); 1294 seldestroy(&sc->sc_rsel); 1295 1296 #ifdef AUDIO_PM_IDLE 1297 callout_destroy(&sc->sc_idle_counter); 1298 #endif 1299 1300 cv_destroy(&sc->sc_exlockcv); 1301 1302 #if defined(AUDIO_DEBUG) 1303 audio_mlog_free(); 1304 #endif 1305 1306 return 0; 1307 } 1308 1309 static void 1310 audiochilddet(device_t self, device_t child) 1311 { 1312 1313 /* we hold no child references, so do nothing */ 1314 } 1315 1316 static int 1317 audiosearch(device_t parent, cfdata_t cf, const int *locs, void *aux) 1318 { 1319 1320 if (config_match(parent, cf, aux)) 1321 config_attach_loc(parent, cf, locs, aux, NULL); 1322 1323 return 0; 1324 } 1325 1326 static int 1327 audiorescan(device_t self, const char *ifattr, const int *flags) 1328 { 1329 struct audio_softc *sc = device_private(self); 1330 1331 if (!ifattr_match(ifattr, "audio")) 1332 return 0; 1333 1334 config_search_loc(audiosearch, sc->sc_dev, "audio", NULL, NULL); 1335 1336 return 0; 1337 } 1338 1339 /* 1340 * Called from hardware driver. This is where the MI audio driver gets 1341 * probed/attached to the hardware driver. 1342 */ 1343 device_t 1344 audio_attach_mi(const struct audio_hw_if *ahwp, void *hdlp, device_t dev) 1345 { 1346 struct audio_attach_args arg; 1347 1348 #ifdef DIAGNOSTIC 1349 if (ahwp == NULL) { 1350 aprint_error("audio_attach_mi: NULL\n"); 1351 return 0; 1352 } 1353 #endif 1354 arg.type = AUDIODEV_TYPE_AUDIO; 1355 arg.hwif = ahwp; 1356 arg.hdl = hdlp; 1357 return config_found(dev, &arg, audioprint); 1358 } 1359 1360 /* 1361 * Acquire sc_lock and enter exlock critical section. 1362 * If successful, it returns 0. Otherwise returns errno. 1363 */ 1364 static int 1365 audio_enter_exclusive(struct audio_softc *sc) 1366 { 1367 int error; 1368 1369 KASSERT(!mutex_owned(sc->sc_lock)); 1370 1371 mutex_enter(sc->sc_lock); 1372 if (sc->sc_dying) { 1373 mutex_exit(sc->sc_lock); 1374 return EIO; 1375 } 1376 1377 while (__predict_false(sc->sc_exlock != 0)) { 1378 error = cv_wait_sig(&sc->sc_exlockcv, sc->sc_lock); 1379 if (sc->sc_dying) 1380 error = EIO; 1381 if (error) { 1382 mutex_exit(sc->sc_lock); 1383 return error; 1384 } 1385 } 1386 1387 /* Acquire */ 1388 sc->sc_exlock = 1; 1389 return 0; 1390 } 1391 1392 /* 1393 * Leave exlock critical section and release sc_lock. 1394 * Must be called with sc_lock held. 1395 */ 1396 static void 1397 audio_exit_exclusive(struct audio_softc *sc) 1398 { 1399 1400 KASSERT(mutex_owned(sc->sc_lock)); 1401 KASSERT(sc->sc_exlock); 1402 1403 /* Leave critical section */ 1404 sc->sc_exlock = 0; 1405 cv_broadcast(&sc->sc_exlockcv); 1406 mutex_exit(sc->sc_lock); 1407 } 1408 1409 /* 1410 * Wait for I/O to complete, releasing sc_lock. 1411 * Must be called with sc_lock held. 1412 */ 1413 static int 1414 audio_track_waitio(struct audio_softc *sc, audio_track_t *track) 1415 { 1416 int error; 1417 1418 KASSERT(track); 1419 KASSERT(mutex_owned(sc->sc_lock)); 1420 1421 /* Wait for pending I/O to complete. */ 1422 error = cv_timedwait_sig(&track->mixer->outcv, sc->sc_lock, 1423 mstohz(AUDIO_TIMEOUT)); 1424 if (sc->sc_dying) { 1425 error = EIO; 1426 } 1427 if (error) { 1428 TRACET(2, track, "cv_timedwait_sig failed %d", error); 1429 if (error == EWOULDBLOCK) 1430 device_printf(sc->sc_dev, "device timeout\n"); 1431 } else { 1432 TRACET(3, track, "wakeup"); 1433 } 1434 return error; 1435 } 1436 1437 /* 1438 * Acquire the file lock. 1439 * If file is acquired successfully, returns 0. Otherwise returns errno. 1440 * In both case, sc_lock is released. 1441 */ 1442 static int 1443 audio_file_acquire(struct audio_softc *sc, audio_file_t *file) 1444 { 1445 int error; 1446 1447 KASSERT(!mutex_owned(sc->sc_lock)); 1448 1449 mutex_enter(sc->sc_lock); 1450 if (sc->sc_dying) { 1451 mutex_exit(sc->sc_lock); 1452 return EIO; 1453 } 1454 1455 while (__predict_false(file->lock != 0)) { 1456 error = cv_wait_sig(&sc->sc_exlockcv, sc->sc_lock); 1457 if (sc->sc_dying) 1458 error = EIO; 1459 if (error) { 1460 mutex_exit(sc->sc_lock); 1461 return error; 1462 } 1463 } 1464 1465 /* Mark this file locked */ 1466 file->lock = 1; 1467 mutex_exit(sc->sc_lock); 1468 1469 return 0; 1470 } 1471 1472 /* 1473 * Release the file lock. 1474 */ 1475 static void 1476 audio_file_release(struct audio_softc *sc, audio_file_t *file) 1477 { 1478 1479 KASSERT(!mutex_owned(sc->sc_lock)); 1480 1481 mutex_enter(sc->sc_lock); 1482 KASSERT(file->lock); 1483 file->lock = 0; 1484 cv_broadcast(&sc->sc_exlockcv); 1485 mutex_exit(sc->sc_lock); 1486 } 1487 1488 /* 1489 * Try to acquire track lock. 1490 * It doesn't block if the track lock is already aquired. 1491 * Returns true if the track lock was acquired, or false if the track 1492 * lock was already acquired. 1493 */ 1494 static __inline bool 1495 audio_track_lock_tryenter(audio_track_t *track) 1496 { 1497 return (atomic_cas_uint(&track->lock, 0, 1) == 0); 1498 } 1499 1500 /* 1501 * Acquire track lock. 1502 */ 1503 static __inline void 1504 audio_track_lock_enter(audio_track_t *track) 1505 { 1506 /* Don't sleep here. */ 1507 while (audio_track_lock_tryenter(track) == false) 1508 ; 1509 } 1510 1511 /* 1512 * Release track lock. 1513 */ 1514 static __inline void 1515 audio_track_lock_exit(audio_track_t *track) 1516 { 1517 atomic_swap_uint(&track->lock, 0); 1518 } 1519 1520 1521 static int 1522 audioopen(dev_t dev, int flags, int ifmt, struct lwp *l) 1523 { 1524 struct audio_softc *sc; 1525 int error; 1526 1527 /* Find the device */ 1528 sc = device_lookup_private(&audio_cd, AUDIOUNIT(dev)); 1529 if (sc == NULL || sc->hw_if == NULL) 1530 return ENXIO; 1531 1532 error = audio_enter_exclusive(sc); 1533 if (error) 1534 return error; 1535 1536 device_active(sc->sc_dev, DVA_SYSTEM); 1537 switch (AUDIODEV(dev)) { 1538 case SOUND_DEVICE: 1539 case AUDIO_DEVICE: 1540 error = audio_open(dev, sc, flags, ifmt, l, NULL); 1541 break; 1542 case AUDIOCTL_DEVICE: 1543 error = audioctl_open(dev, sc, flags, ifmt, l); 1544 break; 1545 case MIXER_DEVICE: 1546 error = mixer_open(dev, sc, flags, ifmt, l); 1547 break; 1548 default: 1549 error = ENXIO; 1550 break; 1551 } 1552 audio_exit_exclusive(sc); 1553 1554 return error; 1555 } 1556 1557 static int 1558 audioclose(struct file *fp) 1559 { 1560 struct audio_softc *sc; 1561 audio_file_t *file; 1562 int error; 1563 dev_t dev; 1564 1565 KASSERT(fp->f_audioctx); 1566 file = fp->f_audioctx; 1567 sc = file->sc; 1568 dev = file->dev; 1569 1570 /* Acquire file lock and exlock */ 1571 /* XXX what should I do when an error occurs? */ 1572 error = audio_file_acquire(sc, file); 1573 if (error) 1574 return error; 1575 1576 device_active(sc->sc_dev, DVA_SYSTEM); 1577 switch (AUDIODEV(dev)) { 1578 case SOUND_DEVICE: 1579 case AUDIO_DEVICE: 1580 error = audio_close(sc, file); 1581 break; 1582 case AUDIOCTL_DEVICE: 1583 error = 0; 1584 break; 1585 case MIXER_DEVICE: 1586 error = mixer_close(sc, file); 1587 break; 1588 default: 1589 error = ENXIO; 1590 break; 1591 } 1592 if (error == 0) { 1593 kmem_free(fp->f_audioctx, sizeof(audio_file_t)); 1594 fp->f_audioctx = NULL; 1595 } 1596 1597 /* 1598 * Since file has already been destructed, 1599 * audio_file_release() is not necessary. 1600 */ 1601 1602 return error; 1603 } 1604 1605 static int 1606 audioread(struct file *fp, off_t *offp, struct uio *uio, kauth_cred_t cred, 1607 int ioflag) 1608 { 1609 struct audio_softc *sc; 1610 audio_file_t *file; 1611 int error; 1612 dev_t dev; 1613 1614 KASSERT(fp->f_audioctx); 1615 file = fp->f_audioctx; 1616 sc = file->sc; 1617 dev = file->dev; 1618 1619 error = audio_file_acquire(sc, file); 1620 if (error) 1621 return error; 1622 1623 if (fp->f_flag & O_NONBLOCK) 1624 ioflag |= IO_NDELAY; 1625 1626 switch (AUDIODEV(dev)) { 1627 case SOUND_DEVICE: 1628 case AUDIO_DEVICE: 1629 error = audio_read(sc, uio, ioflag, file); 1630 break; 1631 case AUDIOCTL_DEVICE: 1632 case MIXER_DEVICE: 1633 error = ENODEV; 1634 break; 1635 default: 1636 error = ENXIO; 1637 break; 1638 } 1639 audio_file_release(sc, file); 1640 1641 return error; 1642 } 1643 1644 static int 1645 audiowrite(struct file *fp, off_t *offp, struct uio *uio, kauth_cred_t cred, 1646 int ioflag) 1647 { 1648 struct audio_softc *sc; 1649 audio_file_t *file; 1650 int error; 1651 dev_t dev; 1652 1653 KASSERT(fp->f_audioctx); 1654 file = fp->f_audioctx; 1655 sc = file->sc; 1656 dev = file->dev; 1657 1658 error = audio_file_acquire(sc, file); 1659 if (error) 1660 return error; 1661 1662 if (fp->f_flag & O_NONBLOCK) 1663 ioflag |= IO_NDELAY; 1664 1665 switch (AUDIODEV(dev)) { 1666 case SOUND_DEVICE: 1667 case AUDIO_DEVICE: 1668 error = audio_write(sc, uio, ioflag, file); 1669 break; 1670 case AUDIOCTL_DEVICE: 1671 case MIXER_DEVICE: 1672 error = ENODEV; 1673 break; 1674 default: 1675 error = ENXIO; 1676 break; 1677 } 1678 audio_file_release(sc, file); 1679 1680 return error; 1681 } 1682 1683 static int 1684 audioioctl(struct file *fp, u_long cmd, void *addr) 1685 { 1686 struct audio_softc *sc; 1687 audio_file_t *file; 1688 struct lwp *l = curlwp; 1689 int error; 1690 dev_t dev; 1691 1692 KASSERT(fp->f_audioctx); 1693 file = fp->f_audioctx; 1694 sc = file->sc; 1695 dev = file->dev; 1696 1697 error = audio_file_acquire(sc, file); 1698 if (error) 1699 return error; 1700 1701 switch (AUDIODEV(dev)) { 1702 case SOUND_DEVICE: 1703 case AUDIO_DEVICE: 1704 case AUDIOCTL_DEVICE: 1705 mutex_enter(sc->sc_lock); 1706 device_active(sc->sc_dev, DVA_SYSTEM); 1707 mutex_exit(sc->sc_lock); 1708 if (IOCGROUP(cmd) == IOCGROUP(AUDIO_MIXER_READ)) 1709 error = mixer_ioctl(sc, cmd, addr, fp->f_flag, l); 1710 else 1711 error = audio_ioctl(dev, sc, cmd, addr, fp->f_flag, l, 1712 file); 1713 break; 1714 case MIXER_DEVICE: 1715 error = mixer_ioctl(sc, cmd, addr, fp->f_flag, l); 1716 break; 1717 default: 1718 error = ENXIO; 1719 break; 1720 } 1721 audio_file_release(sc, file); 1722 1723 return error; 1724 } 1725 1726 static int 1727 audiostat(struct file *fp, struct stat *st) 1728 { 1729 audio_file_t *file; 1730 1731 KASSERT(fp->f_audioctx); 1732 file = fp->f_audioctx; 1733 1734 memset(st, 0, sizeof(*st)); 1735 1736 st->st_dev = file->dev; 1737 st->st_uid = kauth_cred_geteuid(fp->f_cred); 1738 st->st_gid = kauth_cred_getegid(fp->f_cred); 1739 st->st_mode = S_IFCHR; 1740 return 0; 1741 } 1742 1743 static int 1744 audiopoll(struct file *fp, int events) 1745 { 1746 struct audio_softc *sc; 1747 audio_file_t *file; 1748 struct lwp *l = curlwp; 1749 int revents; 1750 dev_t dev; 1751 1752 KASSERT(fp->f_audioctx); 1753 file = fp->f_audioctx; 1754 sc = file->sc; 1755 dev = file->dev; 1756 1757 if (audio_file_acquire(sc, file) != 0) 1758 return 0; 1759 1760 switch (AUDIODEV(dev)) { 1761 case SOUND_DEVICE: 1762 case AUDIO_DEVICE: 1763 revents = audio_poll(sc, events, l, file); 1764 break; 1765 case AUDIOCTL_DEVICE: 1766 case MIXER_DEVICE: 1767 revents = 0; 1768 break; 1769 default: 1770 revents = POLLERR; 1771 break; 1772 } 1773 audio_file_release(sc, file); 1774 1775 return revents; 1776 } 1777 1778 static int 1779 audiokqfilter(struct file *fp, struct knote *kn) 1780 { 1781 struct audio_softc *sc; 1782 audio_file_t *file; 1783 dev_t dev; 1784 int error; 1785 1786 KASSERT(fp->f_audioctx); 1787 file = fp->f_audioctx; 1788 sc = file->sc; 1789 dev = file->dev; 1790 1791 error = audio_file_acquire(sc, file); 1792 if (error) 1793 return error; 1794 1795 switch (AUDIODEV(dev)) { 1796 case SOUND_DEVICE: 1797 case AUDIO_DEVICE: 1798 error = audio_kqfilter(sc, file, kn); 1799 break; 1800 case AUDIOCTL_DEVICE: 1801 case MIXER_DEVICE: 1802 error = ENODEV; 1803 break; 1804 default: 1805 error = ENXIO; 1806 break; 1807 } 1808 audio_file_release(sc, file); 1809 1810 return error; 1811 } 1812 1813 static int 1814 audiommap(struct file *fp, off_t *offp, size_t len, int prot, int *flagsp, 1815 int *advicep, struct uvm_object **uobjp, int *maxprotp) 1816 { 1817 struct audio_softc *sc; 1818 audio_file_t *file; 1819 dev_t dev; 1820 int error; 1821 1822 KASSERT(fp->f_audioctx); 1823 file = fp->f_audioctx; 1824 sc = file->sc; 1825 dev = file->dev; 1826 1827 error = audio_file_acquire(sc, file); 1828 if (error) 1829 return error; 1830 1831 mutex_enter(sc->sc_lock); 1832 device_active(sc->sc_dev, DVA_SYSTEM); /* XXXJDM */ 1833 mutex_exit(sc->sc_lock); 1834 1835 switch (AUDIODEV(dev)) { 1836 case SOUND_DEVICE: 1837 case AUDIO_DEVICE: 1838 error = audio_mmap(sc, offp, len, prot, flagsp, advicep, 1839 uobjp, maxprotp, file); 1840 break; 1841 case AUDIOCTL_DEVICE: 1842 case MIXER_DEVICE: 1843 default: 1844 error = ENOTSUP; 1845 break; 1846 } 1847 audio_file_release(sc, file); 1848 1849 return error; 1850 } 1851 1852 1853 /* Exported interfaces for audiobell. */ 1854 1855 /* 1856 * Open for audiobell. 1857 * sample_rate, encoding, precision and channels in arg are in-parameter 1858 * and indicates input encoding. 1859 * Stores allocated file to arg->file. 1860 * Stores blocksize to arg->blocksize. 1861 * If successful returns 0, otherwise errno. 1862 */ 1863 int 1864 audiobellopen(dev_t dev, struct audiobell_arg *arg) 1865 { 1866 struct audio_softc *sc; 1867 int error; 1868 1869 /* Find the device */ 1870 sc = device_lookup_private(&audio_cd, AUDIOUNIT(dev)); 1871 if (sc == NULL || sc->hw_if == NULL) 1872 return ENXIO; 1873 1874 error = audio_enter_exclusive(sc); 1875 if (error) 1876 return error; 1877 1878 device_active(sc->sc_dev, DVA_SYSTEM); 1879 error = audio_open(dev, sc, FWRITE, 0, curlwp, arg); 1880 1881 audio_exit_exclusive(sc); 1882 return error; 1883 } 1884 1885 /* Close for audiobell */ 1886 int 1887 audiobellclose(audio_file_t *file) 1888 { 1889 struct audio_softc *sc; 1890 int error; 1891 1892 sc = file->sc; 1893 1894 /* XXX what should I do when an error occurs? */ 1895 error = audio_file_acquire(sc, file); 1896 if (error) 1897 return error; 1898 1899 device_active(sc->sc_dev, DVA_SYSTEM); 1900 error = audio_close(sc, file); 1901 1902 /* 1903 * Since file has already been destructed, 1904 * audio_file_release() is not necessary. 1905 */ 1906 1907 return error; 1908 } 1909 1910 /* Playback for audiobell */ 1911 int 1912 audiobellwrite(audio_file_t *file, struct uio *uio) 1913 { 1914 struct audio_softc *sc; 1915 int error; 1916 1917 sc = file->sc; 1918 error = audio_file_acquire(sc, file); 1919 if (error) 1920 return error; 1921 1922 error = audio_write(sc, uio, 0, file); 1923 1924 audio_file_release(sc, file); 1925 return error; 1926 } 1927 1928 1929 /* 1930 * Audio driver 1931 */ 1932 int 1933 audio_open(dev_t dev, struct audio_softc *sc, int flags, int ifmt, 1934 struct lwp *l, struct audiobell_arg *bell) 1935 { 1936 struct audio_info ai; 1937 struct file *fp; 1938 audio_file_t *af; 1939 audio_ring_t *hwbuf; 1940 bool fullduplex; 1941 int fd; 1942 int error; 1943 1944 KASSERT(mutex_owned(sc->sc_lock)); 1945 KASSERT(sc->sc_exlock); 1946 1947 TRACE(1, "%sflags=0x%x po=%d ro=%d", 1948 (audiodebug >= 3) ? "start " : "", 1949 flags, sc->sc_popens, sc->sc_ropens); 1950 1951 af = kmem_zalloc(sizeof(audio_file_t), KM_SLEEP); 1952 af->sc = sc; 1953 af->dev = dev; 1954 if ((flags & FWRITE) != 0 && audio_can_playback(sc)) 1955 af->mode |= AUMODE_PLAY | AUMODE_PLAY_ALL; 1956 if ((flags & FREAD) != 0 && audio_can_capture(sc)) 1957 af->mode |= AUMODE_RECORD; 1958 if (af->mode == 0) { 1959 error = ENXIO; 1960 goto bad1; 1961 } 1962 1963 fullduplex = (audio_get_props(sc) & AUDIO_PROP_FULLDUPLEX); 1964 1965 /* 1966 * On half duplex hardware, 1967 * 1. if mode is (PLAY | REC), let mode PLAY. 1968 * 2. if mode is PLAY, let mode PLAY if no rec tracks, otherwise error. 1969 * 3. if mode is REC, let mode REC if no play tracks, otherwise error. 1970 */ 1971 if (fullduplex == false) { 1972 if ((af->mode & AUMODE_PLAY)) { 1973 if (sc->sc_ropens != 0) { 1974 TRACE(1, "record track already exists"); 1975 error = ENODEV; 1976 goto bad1; 1977 } 1978 /* Play takes precedence */ 1979 af->mode &= ~AUMODE_RECORD; 1980 } 1981 if ((af->mode & AUMODE_RECORD)) { 1982 if (sc->sc_popens != 0) { 1983 TRACE(1, "play track already exists"); 1984 error = ENODEV; 1985 goto bad1; 1986 } 1987 } 1988 } 1989 1990 /* Create tracks */ 1991 if ((af->mode & AUMODE_PLAY)) 1992 af->ptrack = audio_track_create(sc, sc->sc_pmixer); 1993 if ((af->mode & AUMODE_RECORD)) 1994 af->rtrack = audio_track_create(sc, sc->sc_rmixer); 1995 1996 /* Set parameters */ 1997 AUDIO_INITINFO(&ai); 1998 if (bell) { 1999 ai.play.sample_rate = bell->sample_rate; 2000 ai.play.encoding = bell->encoding; 2001 ai.play.channels = bell->channels; 2002 ai.play.precision = bell->precision; 2003 ai.play.pause = false; 2004 } else if (ISDEVAUDIO(dev)) { 2005 /* If /dev/audio, initialize everytime. */ 2006 ai.play.sample_rate = audio_default.sample_rate; 2007 ai.play.encoding = audio_default.encoding; 2008 ai.play.channels = audio_default.channels; 2009 ai.play.precision = audio_default.precision; 2010 ai.play.pause = false; 2011 ai.record.sample_rate = audio_default.sample_rate; 2012 ai.record.encoding = audio_default.encoding; 2013 ai.record.channels = audio_default.channels; 2014 ai.record.precision = audio_default.precision; 2015 ai.record.pause = false; 2016 } else { 2017 /* If /dev/sound, take over the previous parameters. */ 2018 ai.play.sample_rate = sc->sc_sound_pparams.sample_rate; 2019 ai.play.encoding = sc->sc_sound_pparams.encoding; 2020 ai.play.channels = sc->sc_sound_pparams.channels; 2021 ai.play.precision = sc->sc_sound_pparams.precision; 2022 ai.play.pause = sc->sc_sound_ppause; 2023 ai.record.sample_rate = sc->sc_sound_rparams.sample_rate; 2024 ai.record.encoding = sc->sc_sound_rparams.encoding; 2025 ai.record.channels = sc->sc_sound_rparams.channels; 2026 ai.record.precision = sc->sc_sound_rparams.precision; 2027 ai.record.pause = sc->sc_sound_rpause; 2028 } 2029 error = audio_file_setinfo(sc, af, &ai); 2030 if (error) 2031 goto bad2; 2032 2033 if (sc->sc_popens + sc->sc_ropens == 0) { 2034 /* First open */ 2035 2036 sc->sc_cred = kauth_cred_get(); 2037 kauth_cred_hold(sc->sc_cred); 2038 2039 if (sc->hw_if->open) { 2040 int hwflags; 2041 2042 /* 2043 * Call hw_if->open() only at first open of 2044 * combination of playback and recording. 2045 * On full duplex hardware, the flags passed to 2046 * hw_if->open() is always (FREAD | FWRITE) 2047 * regardless of this open()'s flags. 2048 * see also dev/isa/aria.c 2049 * but ckeck its playback or recording capability. 2050 * On half duplex hardware, the flags passed to 2051 * hw_if->open() is either FREAD or FWRITE. 2052 * see also arch/evbarm/mini2440/audio_mini2440.c 2053 */ 2054 if (fullduplex) { 2055 hwflags = FREAD | FWRITE; 2056 if (!audio_can_playback(sc)) 2057 hwflags &= ~FWRITE; 2058 if (!audio_can_capture(sc)) 2059 hwflags &= ~FREAD; 2060 } else { 2061 /* Construct hwflags from af->mode. */ 2062 hwflags = 0; 2063 if ((af->mode & AUMODE_PLAY) != 0) 2064 hwflags |= FWRITE; 2065 if ((af->mode & AUMODE_RECORD) != 0) 2066 hwflags |= FREAD; 2067 } 2068 2069 mutex_enter(sc->sc_intr_lock); 2070 error = sc->hw_if->open(sc->hw_hdl, hwflags); 2071 mutex_exit(sc->sc_intr_lock); 2072 if (error) 2073 goto bad2; 2074 } 2075 2076 /* 2077 * Set speaker mode when a half duplex. 2078 * XXX I'm not sure this is correct. 2079 */ 2080 if (1/*XXX*/) { 2081 if (sc->hw_if->speaker_ctl) { 2082 int on; 2083 if (af->ptrack) { 2084 on = 1; 2085 } else { 2086 on = 0; 2087 } 2088 mutex_enter(sc->sc_intr_lock); 2089 error = sc->hw_if->speaker_ctl(sc->hw_hdl, on); 2090 mutex_exit(sc->sc_intr_lock); 2091 if (error) 2092 goto bad3; 2093 } 2094 } 2095 } else if (sc->sc_multiuser == false) { 2096 uid_t euid = kauth_cred_geteuid(kauth_cred_get()); 2097 if (euid != 0 && euid != kauth_cred_geteuid(sc->sc_cred)) { 2098 error = EPERM; 2099 goto bad2; 2100 } 2101 } 2102 2103 /* Call init_output if this is the first playback open. */ 2104 if (af->ptrack && sc->sc_popens == 0) { 2105 if (sc->hw_if->init_output) { 2106 hwbuf = &sc->sc_pmixer->hwbuf; 2107 mutex_enter(sc->sc_intr_lock); 2108 error = sc->hw_if->init_output(sc->hw_hdl, 2109 hwbuf->mem, 2110 hwbuf->capacity * 2111 hwbuf->fmt.channels * hwbuf->fmt.stride / NBBY); 2112 mutex_exit(sc->sc_intr_lock); 2113 if (error) 2114 goto bad3; 2115 } 2116 } 2117 /* Call init_input if this is the first recording open. */ 2118 if (af->rtrack && sc->sc_ropens == 0) { 2119 if (sc->hw_if->init_input) { 2120 hwbuf = &sc->sc_rmixer->hwbuf; 2121 mutex_enter(sc->sc_intr_lock); 2122 error = sc->hw_if->init_input(sc->hw_hdl, 2123 hwbuf->mem, 2124 hwbuf->capacity * 2125 hwbuf->fmt.channels * hwbuf->fmt.stride / NBBY); 2126 mutex_exit(sc->sc_intr_lock); 2127 if (error) 2128 goto bad3; 2129 } 2130 } 2131 2132 if (bell == NULL) { 2133 error = fd_allocfile(&fp, &fd); 2134 if (error) 2135 goto bad3; 2136 } 2137 2138 /* 2139 * Count up finally. 2140 * Don't fail from here. 2141 */ 2142 if (af->ptrack) 2143 sc->sc_popens++; 2144 if (af->rtrack) 2145 sc->sc_ropens++; 2146 mutex_enter(sc->sc_intr_lock); 2147 SLIST_INSERT_HEAD(&sc->sc_files, af, entry); 2148 mutex_exit(sc->sc_intr_lock); 2149 2150 if (bell) { 2151 bell->file = af; 2152 } else { 2153 error = fd_clone(fp, fd, flags, &audio_fileops, af); 2154 KASSERT(error == EMOVEFD); 2155 } 2156 2157 TRACEF(3, af, "done"); 2158 return error; 2159 2160 /* 2161 * Since track here is not yet linked to sc_files, 2162 * you can call track_destroy() without sc_intr_lock. 2163 */ 2164 bad3: 2165 if (sc->sc_popens + sc->sc_ropens == 0) { 2166 if (sc->hw_if->close) { 2167 mutex_enter(sc->sc_intr_lock); 2168 sc->hw_if->close(sc->hw_hdl); 2169 mutex_exit(sc->sc_intr_lock); 2170 } 2171 } 2172 bad2: 2173 if (af->rtrack) { 2174 audio_track_destroy(af->rtrack); 2175 af->rtrack = NULL; 2176 } 2177 if (af->ptrack) { 2178 audio_track_destroy(af->ptrack); 2179 af->ptrack = NULL; 2180 } 2181 bad1: 2182 kmem_free(af, sizeof(*af)); 2183 return error; 2184 } 2185 2186 int 2187 audio_close(struct audio_softc *sc, audio_file_t *file) 2188 { 2189 audio_track_t *oldtrack; 2190 int error; 2191 2192 KASSERT(!mutex_owned(sc->sc_lock)); 2193 KASSERT(file->lock); 2194 2195 TRACEF(1, file, "%spid=%d.%d po=%d ro=%d", 2196 (audiodebug >= 3) ? "start " : "", 2197 (int)curproc->p_pid, (int)curlwp->l_lid, 2198 sc->sc_popens, sc->sc_ropens); 2199 KASSERTMSG(sc->sc_popens + sc->sc_ropens > 0, 2200 "sc->sc_popens=%d, sc->sc_ropens=%d", 2201 sc->sc_popens, sc->sc_ropens); 2202 2203 /* 2204 * Drain first. 2205 * It must be done before acquiring exclusive lock. 2206 */ 2207 if (file->ptrack) { 2208 mutex_enter(sc->sc_lock); 2209 audio_track_drain(sc, file->ptrack); 2210 mutex_exit(sc->sc_lock); 2211 } 2212 2213 /* Then, acquire exclusive lock to protect counters. */ 2214 /* XXX what should I do when an error occurs? */ 2215 error = audio_enter_exclusive(sc); 2216 if (error) { 2217 audio_file_release(sc, file); 2218 return error; 2219 } 2220 2221 if (file->ptrack) { 2222 /* Call hw halt_output if this is the last playback track. */ 2223 if (sc->sc_popens == 1 && sc->sc_pbusy) { 2224 error = audio_pmixer_halt(sc); 2225 if (error) { 2226 device_printf(sc->sc_dev, 2227 "halt_output failed with %d\n", error); 2228 } 2229 } 2230 2231 /* Destroy the track. */ 2232 oldtrack = file->ptrack; 2233 mutex_enter(sc->sc_intr_lock); 2234 file->ptrack = NULL; 2235 mutex_exit(sc->sc_intr_lock); 2236 TRACET(3, oldtrack, "dropframes=%" PRIu64, 2237 oldtrack->dropframes); 2238 audio_track_destroy(oldtrack); 2239 2240 KASSERT(sc->sc_popens > 0); 2241 sc->sc_popens--; 2242 } 2243 if (file->rtrack) { 2244 /* Call hw halt_input if this is the last recording track. */ 2245 if (sc->sc_ropens == 1 && sc->sc_rbusy) { 2246 error = audio_rmixer_halt(sc); 2247 if (error) { 2248 device_printf(sc->sc_dev, 2249 "halt_input failed with %d\n", error); 2250 } 2251 } 2252 2253 /* Destroy the track. */ 2254 oldtrack = file->rtrack; 2255 mutex_enter(sc->sc_intr_lock); 2256 file->rtrack = NULL; 2257 mutex_exit(sc->sc_intr_lock); 2258 TRACET(3, oldtrack, "dropframes=%" PRIu64, 2259 oldtrack->dropframes); 2260 audio_track_destroy(oldtrack); 2261 2262 KASSERT(sc->sc_ropens > 0); 2263 sc->sc_ropens--; 2264 } 2265 2266 /* Call hw close if this is the last track. */ 2267 if (sc->sc_popens + sc->sc_ropens == 0) { 2268 if (sc->hw_if->close) { 2269 TRACE(2, "hw_if close"); 2270 mutex_enter(sc->sc_intr_lock); 2271 sc->hw_if->close(sc->hw_hdl); 2272 mutex_exit(sc->sc_intr_lock); 2273 } 2274 2275 kauth_cred_free(sc->sc_cred); 2276 } 2277 2278 mutex_enter(sc->sc_intr_lock); 2279 SLIST_REMOVE(&sc->sc_files, file, audio_file, entry); 2280 mutex_exit(sc->sc_intr_lock); 2281 2282 TRACE(3, "done"); 2283 audio_exit_exclusive(sc); 2284 return 0; 2285 } 2286 2287 int 2288 audio_read(struct audio_softc *sc, struct uio *uio, int ioflag, 2289 audio_file_t *file) 2290 { 2291 audio_track_t *track; 2292 audio_ring_t *usrbuf; 2293 audio_ring_t *input; 2294 int error; 2295 2296 track = file->rtrack; 2297 KASSERT(track); 2298 TRACET(2, track, "resid=%zd", uio->uio_resid); 2299 2300 KASSERT(!mutex_owned(sc->sc_lock)); 2301 KASSERT(file->lock); 2302 2303 /* I think it's better than EINVAL. */ 2304 if (track->mmapped) 2305 return EPERM; 2306 2307 #ifdef AUDIO_PM_IDLE 2308 mutex_enter(sc->sc_lock); 2309 if (device_is_active(&sc->sc_dev) || sc->sc_idle) 2310 device_active(&sc->sc_dev, DVA_SYSTEM); 2311 mutex_exit(sc->sc_lock); 2312 #endif 2313 2314 /* 2315 * On half-duplex hardware, O_RDWR is treated as O_WRONLY. 2316 * However read() system call itself can be called because it's 2317 * opened with O_RDWR. So in this case, deny this read(). 2318 */ 2319 if ((file->mode & AUMODE_RECORD) == 0) { 2320 return EBADF; 2321 } 2322 2323 TRACET(3, track, "resid=%zd", uio->uio_resid); 2324 2325 usrbuf = &track->usrbuf; 2326 input = track->input; 2327 2328 /* 2329 * The first read starts rmixer. 2330 */ 2331 error = audio_enter_exclusive(sc); 2332 if (error) 2333 return error; 2334 if (sc->sc_rbusy == false) 2335 audio_rmixer_start(sc); 2336 audio_exit_exclusive(sc); 2337 2338 error = 0; 2339 while (uio->uio_resid > 0 && error == 0) { 2340 int bytes; 2341 2342 TRACET(3, track, 2343 "while resid=%zd input=%d/%d/%d usrbuf=%d/%d/H%d", 2344 uio->uio_resid, 2345 input->head, input->used, input->capacity, 2346 usrbuf->head, usrbuf->used, track->usrbuf_usedhigh); 2347 2348 /* Wait when buffers are empty. */ 2349 mutex_enter(sc->sc_lock); 2350 for (;;) { 2351 bool empty; 2352 audio_track_lock_enter(track); 2353 empty = (input->used == 0 && usrbuf->used == 0); 2354 audio_track_lock_exit(track); 2355 if (!empty) 2356 break; 2357 2358 if ((ioflag & IO_NDELAY)) { 2359 mutex_exit(sc->sc_lock); 2360 return EWOULDBLOCK; 2361 } 2362 2363 TRACET(3, track, "sleep"); 2364 error = audio_track_waitio(sc, track); 2365 if (error) { 2366 mutex_exit(sc->sc_lock); 2367 return error; 2368 } 2369 } 2370 mutex_exit(sc->sc_lock); 2371 2372 audio_track_lock_enter(track); 2373 audio_track_record(track); 2374 audio_track_lock_exit(track); 2375 2376 /* uiomove from usrbuf as much as possible. */ 2377 bytes = uimin(usrbuf->used, uio->uio_resid); 2378 while (bytes > 0) { 2379 int head = usrbuf->head; 2380 int len = uimin(bytes, usrbuf->capacity - head); 2381 error = uiomove((uint8_t *)usrbuf->mem + head, len, 2382 uio); 2383 if (error) { 2384 device_printf(sc->sc_dev, 2385 "uiomove(len=%d) failed with %d\n", 2386 len, error); 2387 goto abort; 2388 } 2389 auring_take(usrbuf, len); 2390 track->useriobytes += len; 2391 TRACET(3, track, "uiomove(len=%d) usrbuf=%d/%d/C%d", 2392 len, 2393 usrbuf->head, usrbuf->used, usrbuf->capacity); 2394 bytes -= len; 2395 } 2396 } 2397 2398 abort: 2399 return error; 2400 } 2401 2402 2403 /* 2404 * Clear file's playback and/or record track buffer immediately. 2405 */ 2406 static void 2407 audio_file_clear(struct audio_softc *sc, audio_file_t *file) 2408 { 2409 2410 if (file->ptrack) 2411 audio_track_clear(sc, file->ptrack); 2412 if (file->rtrack) 2413 audio_track_clear(sc, file->rtrack); 2414 } 2415 2416 int 2417 audio_write(struct audio_softc *sc, struct uio *uio, int ioflag, 2418 audio_file_t *file) 2419 { 2420 audio_track_t *track; 2421 audio_ring_t *usrbuf; 2422 audio_ring_t *outbuf; 2423 int error; 2424 2425 track = file->ptrack; 2426 KASSERT(track); 2427 TRACET(2, track, "%sresid=%zd pid=%d.%d ioflag=0x%x", 2428 audiodebug >= 3 ? "begin " : "", 2429 uio->uio_resid, (int)curproc->p_pid, (int)curlwp->l_lid, ioflag); 2430 2431 KASSERT(!mutex_owned(sc->sc_lock)); 2432 KASSERT(file->lock); 2433 2434 /* I think it's better than EINVAL. */ 2435 if (track->mmapped) 2436 return EPERM; 2437 2438 if (uio->uio_resid == 0) { 2439 track->eofcounter++; 2440 return 0; 2441 } 2442 2443 #ifdef AUDIO_PM_IDLE 2444 mutex_enter(sc->sc_lock); 2445 if (device_is_active(&sc->sc_dev) || sc->sc_idle) 2446 device_active(&sc->sc_dev, DVA_SYSTEM); 2447 mutex_exit(sc->sc_lock); 2448 #endif 2449 2450 usrbuf = &track->usrbuf; 2451 outbuf = &track->outbuf; 2452 2453 /* 2454 * The first write starts pmixer. 2455 */ 2456 error = audio_enter_exclusive(sc); 2457 if (error) 2458 return error; 2459 if (sc->sc_pbusy == false) 2460 audio_pmixer_start(sc, false); 2461 audio_exit_exclusive(sc); 2462 2463 track->pstate = AUDIO_STATE_RUNNING; 2464 error = 0; 2465 while (uio->uio_resid > 0 && error == 0) { 2466 int bytes; 2467 2468 TRACET(3, track, "while resid=%zd usrbuf=%d/%d/H%d", 2469 uio->uio_resid, 2470 usrbuf->head, usrbuf->used, track->usrbuf_usedhigh); 2471 2472 /* Wait when buffers are full. */ 2473 mutex_enter(sc->sc_lock); 2474 for (;;) { 2475 bool full; 2476 audio_track_lock_enter(track); 2477 full = (usrbuf->used >= track->usrbuf_usedhigh && 2478 outbuf->used >= outbuf->capacity); 2479 audio_track_lock_exit(track); 2480 if (!full) 2481 break; 2482 2483 if ((ioflag & IO_NDELAY)) { 2484 error = EWOULDBLOCK; 2485 mutex_exit(sc->sc_lock); 2486 goto abort; 2487 } 2488 2489 TRACET(3, track, "sleep usrbuf=%d/H%d", 2490 usrbuf->used, track->usrbuf_usedhigh); 2491 error = audio_track_waitio(sc, track); 2492 if (error) { 2493 mutex_exit(sc->sc_lock); 2494 goto abort; 2495 } 2496 } 2497 mutex_exit(sc->sc_lock); 2498 2499 /* uiomove to usrbuf as much as possible. */ 2500 bytes = uimin(track->usrbuf_usedhigh - usrbuf->used, 2501 uio->uio_resid); 2502 while (bytes > 0) { 2503 int tail = auring_tail(usrbuf); 2504 int len = uimin(bytes, usrbuf->capacity - tail); 2505 error = uiomove((uint8_t *)usrbuf->mem + tail, len, 2506 uio); 2507 if (error) { 2508 device_printf(sc->sc_dev, 2509 "uiomove(len=%d) failed with %d\n", 2510 len, error); 2511 goto abort; 2512 } 2513 auring_push(usrbuf, len); 2514 track->useriobytes += len; 2515 TRACET(3, track, "uiomove(len=%d) usrbuf=%d/%d/C%d", 2516 len, 2517 usrbuf->head, usrbuf->used, usrbuf->capacity); 2518 bytes -= len; 2519 } 2520 2521 /* Convert them as much as possible. */ 2522 audio_track_lock_enter(track); 2523 while (usrbuf->used >= track->usrbuf_blksize && 2524 outbuf->used < outbuf->capacity) { 2525 audio_track_play(track); 2526 } 2527 audio_track_lock_exit(track); 2528 } 2529 2530 abort: 2531 TRACET(3, track, "done error=%d", error); 2532 return error; 2533 } 2534 2535 int 2536 audio_ioctl(dev_t dev, struct audio_softc *sc, u_long cmd, void *addr, int flag, 2537 struct lwp *l, audio_file_t *file) 2538 { 2539 struct audio_offset *ao; 2540 struct audio_info ai; 2541 audio_track_t *track; 2542 audio_encoding_t *ae; 2543 audio_format_query_t *query; 2544 u_int stamp; 2545 u_int offs; 2546 int fd; 2547 int index; 2548 int error; 2549 2550 KASSERT(!mutex_owned(sc->sc_lock)); 2551 KASSERT(file->lock); 2552 2553 #if defined(AUDIO_DEBUG) 2554 const char *ioctlnames[] = { 2555 " AUDIO_GETINFO", /* 21 */ 2556 " AUDIO_SETINFO", /* 22 */ 2557 " AUDIO_DRAIN", /* 23 */ 2558 " AUDIO_FLUSH", /* 24 */ 2559 " AUDIO_WSEEK", /* 25 */ 2560 " AUDIO_RERROR", /* 26 */ 2561 " AUDIO_GETDEV", /* 27 */ 2562 " AUDIO_GETENC", /* 28 */ 2563 " AUDIO_GETFD", /* 29 */ 2564 " AUDIO_SETFD", /* 30 */ 2565 " AUDIO_PERROR", /* 31 */ 2566 " AUDIO_GETIOFFS", /* 32 */ 2567 " AUDIO_GETOOFFS", /* 33 */ 2568 " AUDIO_GETPROPS", /* 34 */ 2569 " AUDIO_GETBUFINFO", /* 35 */ 2570 " AUDIO_SETCHAN", /* 36 */ 2571 " AUDIO_GETCHAN", /* 37 */ 2572 " AUDIO_QUERYFORMAT", /* 38 */ 2573 " AUDIO_GETFORMAT", /* 39 */ 2574 " AUDIO_SETFORMAT", /* 40 */ 2575 }; 2576 int nameidx = (cmd & 0xff); 2577 const char *ioctlname = ""; 2578 if (21 <= nameidx && nameidx <= 21 + __arraycount(ioctlnames)) 2579 ioctlname = ioctlnames[nameidx - 21]; 2580 TRACEF(2, file, "(%lu,'%c',%lu)%s pid=%d.%d", 2581 IOCPARM_LEN(cmd), (char)IOCGROUP(cmd), cmd&0xff, ioctlname, 2582 (int)curproc->p_pid, (int)l->l_lid); 2583 #endif 2584 2585 error = 0; 2586 switch (cmd) { 2587 case FIONBIO: 2588 /* All handled in the upper FS layer. */ 2589 break; 2590 2591 case FIONREAD: 2592 /* Get the number of bytes that can be read. */ 2593 if (file->rtrack) { 2594 *(int *)addr = audio_track_readablebytes(file->rtrack); 2595 } else { 2596 *(int *)addr = 0; 2597 } 2598 break; 2599 2600 case FIOASYNC: 2601 /* Set/Clear ASYNC I/O. */ 2602 if (*(int *)addr) { 2603 file->async_audio = curproc->p_pid; 2604 TRACEF(2, file, "FIOASYNC pid %d", file->async_audio); 2605 } else { 2606 file->async_audio = 0; 2607 TRACEF(2, file, "FIOASYNC off"); 2608 } 2609 break; 2610 2611 case AUDIO_FLUSH: 2612 /* XXX TODO: clear errors and restart? */ 2613 audio_file_clear(sc, file); 2614 break; 2615 2616 case AUDIO_RERROR: 2617 /* 2618 * Number of read bytes dropped. We don't know where 2619 * or when they were dropped (including conversion stage). 2620 * Therefore, the number of accurate bytes or samples is 2621 * also unknown. 2622 */ 2623 track = file->rtrack; 2624 if (track) { 2625 *(int *)addr = frametobyte(&track->usrbuf.fmt, 2626 track->dropframes); 2627 } 2628 break; 2629 2630 case AUDIO_PERROR: 2631 /* 2632 * Number of write bytes dropped. We don't know where 2633 * or when they were dropped (including conversion stage). 2634 * Therefore, the number of accurate bytes or samples is 2635 * also unknown. 2636 */ 2637 track = file->ptrack; 2638 if (track) { 2639 *(int *)addr = frametobyte(&track->usrbuf.fmt, 2640 track->dropframes); 2641 } 2642 break; 2643 2644 case AUDIO_GETIOFFS: 2645 /* XXX TODO */ 2646 ao = (struct audio_offset *)addr; 2647 ao->samples = 0; 2648 ao->deltablks = 0; 2649 ao->offset = 0; 2650 break; 2651 2652 case AUDIO_GETOOFFS: 2653 ao = (struct audio_offset *)addr; 2654 track = file->ptrack; 2655 if (track == NULL) { 2656 ao->samples = 0; 2657 ao->deltablks = 0; 2658 ao->offset = 0; 2659 break; 2660 } 2661 mutex_enter(sc->sc_lock); 2662 mutex_enter(sc->sc_intr_lock); 2663 /* figure out where next DMA will start */ 2664 stamp = track->usrbuf_stamp; 2665 offs = track->usrbuf.head; 2666 mutex_exit(sc->sc_intr_lock); 2667 mutex_exit(sc->sc_lock); 2668 2669 ao->samples = stamp; 2670 ao->deltablks = (stamp / track->usrbuf_blksize) - 2671 (track->usrbuf_stamp_last / track->usrbuf_blksize); 2672 track->usrbuf_stamp_last = stamp; 2673 offs = rounddown(offs, track->usrbuf_blksize) 2674 + track->usrbuf_blksize; 2675 if (offs >= track->usrbuf.capacity) 2676 offs -= track->usrbuf.capacity; 2677 ao->offset = offs; 2678 2679 TRACET(3, track, "GETOOFFS: samples=%u deltablks=%u offset=%u", 2680 ao->samples, ao->deltablks, ao->offset); 2681 break; 2682 2683 case AUDIO_WSEEK: 2684 /* XXX return value does not include outbuf one. */ 2685 if (file->ptrack) 2686 *(u_long *)addr = file->ptrack->usrbuf.used; 2687 break; 2688 2689 case AUDIO_SETINFO: 2690 error = audio_enter_exclusive(sc); 2691 if (error) 2692 break; 2693 error = audio_file_setinfo(sc, file, (struct audio_info *)addr); 2694 if (error) { 2695 audio_exit_exclusive(sc); 2696 break; 2697 } 2698 /* XXX TODO: update last_ai if /dev/sound ? */ 2699 if (ISDEVSOUND(dev)) 2700 error = audiogetinfo(sc, &sc->sc_ai, 0, file); 2701 audio_exit_exclusive(sc); 2702 break; 2703 2704 case AUDIO_GETINFO: 2705 error = audio_enter_exclusive(sc); 2706 if (error) 2707 break; 2708 error = audiogetinfo(sc, (struct audio_info *)addr, 1, file); 2709 audio_exit_exclusive(sc); 2710 break; 2711 2712 case AUDIO_GETBUFINFO: 2713 mutex_enter(sc->sc_lock); 2714 error = audiogetinfo(sc, (struct audio_info *)addr, 0, file); 2715 mutex_exit(sc->sc_lock); 2716 break; 2717 2718 case AUDIO_DRAIN: 2719 if (file->ptrack) { 2720 mutex_enter(sc->sc_lock); 2721 error = audio_track_drain(sc, file->ptrack); 2722 mutex_exit(sc->sc_lock); 2723 } 2724 break; 2725 2726 case AUDIO_GETDEV: 2727 mutex_enter(sc->sc_lock); 2728 error = sc->hw_if->getdev(sc->hw_hdl, (audio_device_t *)addr); 2729 mutex_exit(sc->sc_lock); 2730 break; 2731 2732 case AUDIO_GETENC: 2733 ae = (audio_encoding_t *)addr; 2734 index = ae->index; 2735 if (index < 0 || index >= __arraycount(audio_encodings)) { 2736 error = EINVAL; 2737 break; 2738 } 2739 *ae = audio_encodings[index]; 2740 ae->index = index; 2741 /* 2742 * EMULATED always. 2743 * EMULATED flag at that time used to mean that it could 2744 * not be passed directly to the hardware as-is. But 2745 * currently, all formats including hardware native is not 2746 * passed directly to the hardware. So I set EMULATED 2747 * flag for all formats. 2748 */ 2749 ae->flags = AUDIO_ENCODINGFLAG_EMULATED; 2750 break; 2751 2752 case AUDIO_GETFD: 2753 /* 2754 * Returns the current setting of full duplex mode. 2755 * If HW has full duplex mode and there are two mixers, 2756 * it is full duplex. Otherwise half duplex. 2757 */ 2758 mutex_enter(sc->sc_lock); 2759 fd = (audio_get_props(sc) & AUDIO_PROP_FULLDUPLEX) 2760 && (sc->sc_pmixer && sc->sc_rmixer); 2761 mutex_exit(sc->sc_lock); 2762 *(int *)addr = fd; 2763 break; 2764 2765 case AUDIO_GETPROPS: 2766 mutex_enter(sc->sc_lock); 2767 *(int *)addr = audio_get_props(sc); 2768 mutex_exit(sc->sc_lock); 2769 break; 2770 2771 case AUDIO_QUERYFORMAT: 2772 query = (audio_format_query_t *)addr; 2773 if (sc->hw_if->query_format) { 2774 mutex_enter(sc->sc_lock); 2775 error = sc->hw_if->query_format(sc->hw_hdl, query); 2776 mutex_exit(sc->sc_lock); 2777 /* Hide internal infomations */ 2778 query->fmt.driver_data = NULL; 2779 } else { 2780 error = ENODEV; 2781 } 2782 break; 2783 2784 case AUDIO_GETFORMAT: 2785 audio_mixers_get_format(sc, (struct audio_info *)addr); 2786 break; 2787 2788 case AUDIO_SETFORMAT: 2789 mutex_enter(sc->sc_lock); 2790 audio_mixers_get_format(sc, &ai); 2791 error = audio_mixers_set_format(sc, (struct audio_info *)addr); 2792 if (error) { 2793 /* Rollback */ 2794 audio_mixers_set_format(sc, &ai); 2795 } 2796 mutex_exit(sc->sc_lock); 2797 break; 2798 2799 case AUDIO_SETFD: 2800 case AUDIO_SETCHAN: 2801 case AUDIO_GETCHAN: 2802 /* Obsoleted */ 2803 break; 2804 2805 default: 2806 if (sc->hw_if->dev_ioctl) { 2807 error = audio_enter_exclusive(sc); 2808 if (error) 2809 break; 2810 error = sc->hw_if->dev_ioctl(sc->hw_hdl, 2811 cmd, addr, flag, l); 2812 audio_exit_exclusive(sc); 2813 } else { 2814 TRACEF(2, file, "unknown ioctl"); 2815 error = EINVAL; 2816 } 2817 break; 2818 } 2819 TRACEF(2, file, "(%lu,'%c',%lu)%s result %d", 2820 IOCPARM_LEN(cmd), (char)IOCGROUP(cmd), cmd&0xff, ioctlname, 2821 error); 2822 return error; 2823 } 2824 2825 /* 2826 * Returns the number of bytes that can be read on recording buffer. 2827 */ 2828 static __inline int 2829 audio_track_readablebytes(const audio_track_t *track) 2830 { 2831 int bytes; 2832 2833 KASSERT(track); 2834 KASSERT(track->mode == AUMODE_RECORD); 2835 2836 /* 2837 * Although usrbuf is primarily readable data, recorded data 2838 * also stays in track->input until reading. So it is necessary 2839 * to add it. track->input is in frame, usrbuf is in byte. 2840 */ 2841 bytes = track->usrbuf.used + 2842 track->input->used * frametobyte(&track->usrbuf.fmt, 1); 2843 return bytes; 2844 } 2845 2846 int 2847 audio_poll(struct audio_softc *sc, int events, struct lwp *l, 2848 audio_file_t *file) 2849 { 2850 audio_track_t *track; 2851 int revents; 2852 bool in_is_valid; 2853 bool out_is_valid; 2854 2855 KASSERT(!mutex_owned(sc->sc_lock)); 2856 KASSERT(file->lock); 2857 2858 #if defined(AUDIO_DEBUG) 2859 #define POLLEV_BITMAP "\177\020" \ 2860 "b\10WRBAND\0" \ 2861 "b\7RDBAND\0" "b\6RDNORM\0" "b\5NVAL\0" "b\4HUP\0" \ 2862 "b\3ERR\0" "b\2OUT\0" "b\1PRI\0" "b\0IN\0" 2863 char evbuf[64]; 2864 snprintb(evbuf, sizeof(evbuf), POLLEV_BITMAP, events); 2865 TRACEF(2, file, "pid=%d.%d events=%s", 2866 (int)curproc->p_pid, (int)l->l_lid, evbuf); 2867 #endif 2868 2869 revents = 0; 2870 in_is_valid = false; 2871 out_is_valid = false; 2872 if (events & (POLLIN | POLLRDNORM)) { 2873 track = file->rtrack; 2874 if (track) { 2875 int used; 2876 in_is_valid = true; 2877 used = audio_track_readablebytes(track); 2878 if (used > 0) 2879 revents |= events & (POLLIN | POLLRDNORM); 2880 } 2881 } 2882 if (events & (POLLOUT | POLLWRNORM)) { 2883 track = file->ptrack; 2884 if (track) { 2885 out_is_valid = true; 2886 if (track->usrbuf.used <= track->usrbuf_usedlow) 2887 revents |= events & (POLLOUT | POLLWRNORM); 2888 } 2889 } 2890 2891 if (revents == 0) { 2892 mutex_enter(sc->sc_lock); 2893 if (in_is_valid) { 2894 TRACEF(3, file, "selrecord rsel"); 2895 selrecord(l, &sc->sc_rsel); 2896 } 2897 if (out_is_valid) { 2898 TRACEF(3, file, "selrecord wsel"); 2899 selrecord(l, &sc->sc_wsel); 2900 } 2901 mutex_exit(sc->sc_lock); 2902 } 2903 2904 #if defined(AUDIO_DEBUG) 2905 snprintb(evbuf, sizeof(evbuf), POLLEV_BITMAP, revents); 2906 TRACEF(2, file, "revents=%s", evbuf); 2907 #endif 2908 return revents; 2909 } 2910 2911 static const struct filterops audioread_filtops = { 2912 .f_isfd = 1, 2913 .f_attach = NULL, 2914 .f_detach = filt_audioread_detach, 2915 .f_event = filt_audioread_event, 2916 }; 2917 2918 static void 2919 filt_audioread_detach(struct knote *kn) 2920 { 2921 struct audio_softc *sc; 2922 audio_file_t *file; 2923 2924 file = kn->kn_hook; 2925 sc = file->sc; 2926 TRACEF(3, file, ""); 2927 2928 mutex_enter(sc->sc_lock); 2929 SLIST_REMOVE(&sc->sc_rsel.sel_klist, kn, knote, kn_selnext); 2930 mutex_exit(sc->sc_lock); 2931 } 2932 2933 static int 2934 filt_audioread_event(struct knote *kn, long hint) 2935 { 2936 audio_file_t *file; 2937 audio_track_t *track; 2938 2939 file = kn->kn_hook; 2940 track = file->rtrack; 2941 2942 /* 2943 * kn_data must contain the number of bytes can be read. 2944 * The return value indicates whether the event occurs or not. 2945 */ 2946 2947 if (track == NULL) { 2948 /* can not read with this descriptor. */ 2949 kn->kn_data = 0; 2950 return 0; 2951 } 2952 2953 kn->kn_data = audio_track_readablebytes(track); 2954 TRACEF(3, file, "data=%" PRId64, kn->kn_data); 2955 return kn->kn_data > 0; 2956 } 2957 2958 static const struct filterops audiowrite_filtops = { 2959 .f_isfd = 1, 2960 .f_attach = NULL, 2961 .f_detach = filt_audiowrite_detach, 2962 .f_event = filt_audiowrite_event, 2963 }; 2964 2965 static void 2966 filt_audiowrite_detach(struct knote *kn) 2967 { 2968 struct audio_softc *sc; 2969 audio_file_t *file; 2970 2971 file = kn->kn_hook; 2972 sc = file->sc; 2973 TRACEF(3, file, ""); 2974 2975 mutex_enter(sc->sc_lock); 2976 SLIST_REMOVE(&sc->sc_wsel.sel_klist, kn, knote, kn_selnext); 2977 mutex_exit(sc->sc_lock); 2978 } 2979 2980 static int 2981 filt_audiowrite_event(struct knote *kn, long hint) 2982 { 2983 audio_file_t *file; 2984 audio_track_t *track; 2985 2986 file = kn->kn_hook; 2987 track = file->ptrack; 2988 2989 /* 2990 * kn_data must contain the number of bytes can be write. 2991 * The return value indicates whether the event occurs or not. 2992 */ 2993 2994 if (track == NULL) { 2995 /* can not write with this descriptor. */ 2996 kn->kn_data = 0; 2997 return 0; 2998 } 2999 3000 kn->kn_data = track->usrbuf_usedhigh - track->usrbuf.used; 3001 TRACEF(3, file, "data=%" PRId64, kn->kn_data); 3002 return (track->usrbuf.used < track->usrbuf_usedlow); 3003 } 3004 3005 int 3006 audio_kqfilter(struct audio_softc *sc, audio_file_t *file, struct knote *kn) 3007 { 3008 struct klist *klist; 3009 3010 KASSERT(!mutex_owned(sc->sc_lock)); 3011 KASSERT(file->lock); 3012 3013 TRACEF(3, file, "kn=%p kn_filter=%x", kn, (int)kn->kn_filter); 3014 3015 switch (kn->kn_filter) { 3016 case EVFILT_READ: 3017 klist = &sc->sc_rsel.sel_klist; 3018 kn->kn_fop = &audioread_filtops; 3019 break; 3020 3021 case EVFILT_WRITE: 3022 klist = &sc->sc_wsel.sel_klist; 3023 kn->kn_fop = &audiowrite_filtops; 3024 break; 3025 3026 default: 3027 return EINVAL; 3028 } 3029 3030 kn->kn_hook = file; 3031 3032 mutex_enter(sc->sc_lock); 3033 SLIST_INSERT_HEAD(klist, kn, kn_selnext); 3034 mutex_exit(sc->sc_lock); 3035 3036 return 0; 3037 } 3038 3039 int 3040 audio_mmap(struct audio_softc *sc, off_t *offp, size_t len, int prot, 3041 int *flagsp, int *advicep, struct uvm_object **uobjp, int *maxprotp, 3042 audio_file_t *file) 3043 { 3044 audio_track_t *track; 3045 vsize_t vsize; 3046 int error; 3047 3048 KASSERT(!mutex_owned(sc->sc_lock)); 3049 KASSERT(file->lock); 3050 3051 TRACEF(2, file, "off=%lld, prot=%d", (long long)(*offp), prot); 3052 3053 if (*offp < 0) 3054 return EINVAL; 3055 3056 #if 0 3057 /* XXX 3058 * The idea here was to use the protection to determine if 3059 * we are mapping the read or write buffer, but it fails. 3060 * The VM system is broken in (at least) two ways. 3061 * 1) If you map memory VM_PROT_WRITE you SIGSEGV 3062 * when writing to it, so VM_PROT_READ|VM_PROT_WRITE 3063 * has to be used for mmapping the play buffer. 3064 * 2) Even if calling mmap() with VM_PROT_READ|VM_PROT_WRITE 3065 * audio_mmap will get called at some point with VM_PROT_READ 3066 * only. 3067 * So, alas, we always map the play buffer for now. 3068 */ 3069 if (prot == (VM_PROT_READ|VM_PROT_WRITE) || 3070 prot == VM_PROT_WRITE) 3071 track = file->ptrack; 3072 else if (prot == VM_PROT_READ) 3073 track = file->rtrack; 3074 else 3075 return EINVAL; 3076 #else 3077 track = file->ptrack; 3078 #endif 3079 if (track == NULL) 3080 return EACCES; 3081 3082 vsize = roundup2(MAX(track->usrbuf.capacity, PAGE_SIZE), PAGE_SIZE); 3083 if (len > vsize) 3084 return EOVERFLOW; 3085 if (*offp > (uint)(vsize - len)) 3086 return EOVERFLOW; 3087 3088 /* XXX TODO: what happens when mmap twice. */ 3089 if (!track->mmapped) { 3090 track->mmapped = true; 3091 3092 if (!track->is_pause) { 3093 error = audio_enter_exclusive(sc); 3094 if (error) 3095 return error; 3096 if (sc->sc_pbusy == false) 3097 audio_pmixer_start(sc, true); 3098 audio_exit_exclusive(sc); 3099 } 3100 /* XXX mmapping record buffer is not supported */ 3101 } 3102 3103 /* get ringbuffer */ 3104 *uobjp = track->uobj; 3105 3106 /* Acquire a reference for the mmap. munmap will release. */ 3107 uao_reference(*uobjp); 3108 *maxprotp = prot; 3109 *advicep = UVM_ADV_RANDOM; 3110 *flagsp = MAP_SHARED; 3111 return 0; 3112 } 3113 3114 /* 3115 * /dev/audioctl has to be able to open at any time without interference 3116 * with any /dev/audio or /dev/sound. 3117 */ 3118 static int 3119 audioctl_open(dev_t dev, struct audio_softc *sc, int flags, int ifmt, 3120 struct lwp *l) 3121 { 3122 struct file *fp; 3123 audio_file_t *af; 3124 int fd; 3125 int error; 3126 3127 KASSERT(mutex_owned(sc->sc_lock)); 3128 KASSERT(sc->sc_exlock); 3129 3130 TRACE(1, ""); 3131 3132 error = fd_allocfile(&fp, &fd); 3133 if (error) 3134 return error; 3135 3136 af = kmem_zalloc(sizeof(audio_file_t), KM_SLEEP); 3137 af->sc = sc; 3138 af->dev = dev; 3139 3140 /* Not necessary to insert sc_files. */ 3141 3142 error = fd_clone(fp, fd, flags, &audio_fileops, af); 3143 KASSERT(error == EMOVEFD); 3144 3145 return error; 3146 } 3147 3148 /* 3149 * Reallocate 'memblock' with specified 'bytes' if 'bytes' > 0. 3150 * Or free 'memblock' and return NULL if 'byte' is zero. 3151 */ 3152 static void * 3153 audio_realloc(void *memblock, size_t bytes) 3154 { 3155 3156 if (memblock != NULL) { 3157 if (bytes != 0) { 3158 return kern_realloc(memblock, bytes, M_NOWAIT); 3159 } else { 3160 kern_free(memblock); 3161 return NULL; 3162 } 3163 } else { 3164 if (bytes != 0) { 3165 return kern_malloc(bytes, M_NOWAIT); 3166 } else { 3167 return NULL; 3168 } 3169 } 3170 } 3171 3172 /* 3173 * Free 'mem' if available, and initialize the pointer. 3174 * For this reason, this is implemented as macro. 3175 */ 3176 #define audio_free(mem) do { \ 3177 if (mem != NULL) { \ 3178 kern_free(mem); \ 3179 mem = NULL; \ 3180 } \ 3181 } while (0) 3182 3183 /* 3184 * (Re)allocate usrbuf with 'newbufsize' bytes. 3185 * Use this function for usrbuf because only usrbuf can be mmapped. 3186 * If successful, it updates track->usrbuf.mem, track->usrbuf.capacity and 3187 * returns 0. Otherwise, it clears track->usrbuf.mem, track->usrbuf.capacity 3188 * and returns errno. 3189 * It must be called before updating usrbuf.capacity. 3190 */ 3191 static int 3192 audio_realloc_usrbuf(audio_track_t *track, int newbufsize) 3193 { 3194 struct audio_softc *sc; 3195 vaddr_t vstart; 3196 vsize_t oldvsize; 3197 vsize_t newvsize; 3198 int error; 3199 3200 KASSERT(newbufsize > 0); 3201 sc = track->mixer->sc; 3202 3203 /* Get a nonzero multiple of PAGE_SIZE */ 3204 newvsize = roundup2(MAX(newbufsize, PAGE_SIZE), PAGE_SIZE); 3205 3206 if (track->usrbuf.mem != NULL) { 3207 oldvsize = roundup2(MAX(track->usrbuf.capacity, PAGE_SIZE), 3208 PAGE_SIZE); 3209 if (oldvsize == newvsize) { 3210 track->usrbuf.capacity = newbufsize; 3211 return 0; 3212 } 3213 vstart = (vaddr_t)track->usrbuf.mem; 3214 uvm_unmap(kernel_map, vstart, vstart + oldvsize); 3215 /* uvm_unmap also detach uobj */ 3216 track->uobj = NULL; /* paranoia */ 3217 track->usrbuf.mem = NULL; 3218 } 3219 3220 /* Create a uvm anonymous object */ 3221 track->uobj = uao_create(newvsize, 0); 3222 3223 /* Map it into the kernel virtual address space */ 3224 vstart = 0; 3225 error = uvm_map(kernel_map, &vstart, newvsize, track->uobj, 0, 0, 3226 UVM_MAPFLAG(UVM_PROT_RW, UVM_PROT_RW, UVM_INH_NONE, 3227 UVM_ADV_RANDOM, 0)); 3228 if (error) { 3229 device_printf(sc->sc_dev, "uvm_map failed with %d\n", error); 3230 uao_detach(track->uobj); /* release reference */ 3231 goto abort; 3232 } 3233 3234 error = uvm_map_pageable(kernel_map, vstart, vstart + newvsize, 3235 false, 0); 3236 if (error) { 3237 device_printf(sc->sc_dev, "uvm_map_pageable failed with %d\n", 3238 error); 3239 uvm_unmap(kernel_map, vstart, vstart + newvsize); 3240 /* uvm_unmap also detach uobj */ 3241 goto abort; 3242 } 3243 3244 track->usrbuf.mem = (void *)vstart; 3245 track->usrbuf.capacity = newbufsize; 3246 memset(track->usrbuf.mem, 0, newvsize); 3247 return 0; 3248 3249 /* failure */ 3250 abort: 3251 track->uobj = NULL; /* paranoia */ 3252 track->usrbuf.mem = NULL; 3253 track->usrbuf.capacity = 0; 3254 return error; 3255 } 3256 3257 /* 3258 * Free usrbuf (if available). 3259 */ 3260 static void 3261 audio_free_usrbuf(audio_track_t *track) 3262 { 3263 vaddr_t vstart; 3264 vsize_t vsize; 3265 3266 vstart = (vaddr_t)track->usrbuf.mem; 3267 vsize = roundup2(MAX(track->usrbuf.capacity, PAGE_SIZE), PAGE_SIZE); 3268 if (track->usrbuf.mem != NULL) { 3269 /* 3270 * Unmap the kernel mapping. uvm_unmap releases the 3271 * reference to the uvm object, and this should be the 3272 * last virtual mapping of the uvm object, so no need 3273 * to explicitly release (`detach') the object. 3274 */ 3275 uvm_unmap(kernel_map, vstart, vstart + vsize); 3276 3277 track->uobj = NULL; 3278 track->usrbuf.mem = NULL; 3279 track->usrbuf.capacity = 0; 3280 } 3281 } 3282 3283 /* 3284 * This filter changes the volume for each channel. 3285 * arg->context points track->ch_volume[]. 3286 */ 3287 static void 3288 audio_track_chvol(audio_filter_arg_t *arg) 3289 { 3290 int16_t *ch_volume; 3291 const aint_t *s; 3292 aint_t *d; 3293 u_int i; 3294 u_int ch; 3295 u_int channels; 3296 3297 DIAGNOSTIC_filter_arg(arg); 3298 KASSERT(arg->srcfmt->channels == arg->dstfmt->channels); 3299 KASSERT(arg->context != NULL); 3300 KASSERT(arg->srcfmt->channels <= AUDIO_MAX_CHANNELS); 3301 3302 s = arg->src; 3303 d = arg->dst; 3304 ch_volume = arg->context; 3305 3306 channels = arg->srcfmt->channels; 3307 for (i = 0; i < arg->count; i++) { 3308 for (ch = 0; ch < channels; ch++) { 3309 aint2_t val; 3310 val = *s++; 3311 #if defined(AUDIO_USE_C_IMPLEMENTATION_DEFINED_BEHAVIOR) && defined(__GNUC__) 3312 val = val * ch_volume[ch] >> 8; 3313 #else 3314 val = val * ch_volume[ch] / 256; 3315 #endif 3316 *d++ = (aint_t)val; 3317 } 3318 } 3319 } 3320 3321 /* 3322 * This filter performs conversion from stereo (or more channels) to mono. 3323 */ 3324 static void 3325 audio_track_chmix_mixLR(audio_filter_arg_t *arg) 3326 { 3327 const aint_t *s; 3328 aint_t *d; 3329 u_int i; 3330 3331 DIAGNOSTIC_filter_arg(arg); 3332 3333 s = arg->src; 3334 d = arg->dst; 3335 3336 for (i = 0; i < arg->count; i++) { 3337 #if defined(AUDIO_USE_C_IMPLEMENTATION_DEFINED_BEHAVIOR) && defined(__GNUC__) 3338 *d++ = (s[0] >> 1) + (s[1] >> 1); 3339 #else 3340 *d++ = (s[0] / 2) + (s[1] / 2); 3341 #endif 3342 s += arg->srcfmt->channels; 3343 } 3344 } 3345 3346 /* 3347 * This filter performs conversion from mono to stereo (or more channels). 3348 */ 3349 static void 3350 audio_track_chmix_dupLR(audio_filter_arg_t *arg) 3351 { 3352 const aint_t *s; 3353 aint_t *d; 3354 u_int i; 3355 u_int ch; 3356 u_int dstchannels; 3357 3358 DIAGNOSTIC_filter_arg(arg); 3359 3360 s = arg->src; 3361 d = arg->dst; 3362 dstchannels = arg->dstfmt->channels; 3363 3364 for (i = 0; i < arg->count; i++) { 3365 d[0] = s[0]; 3366 d[1] = s[0]; 3367 s++; 3368 d += dstchannels; 3369 } 3370 if (dstchannels > 2) { 3371 d = arg->dst; 3372 for (i = 0; i < arg->count; i++) { 3373 for (ch = 2; ch < dstchannels; ch++) { 3374 d[ch] = 0; 3375 } 3376 d += dstchannels; 3377 } 3378 } 3379 } 3380 3381 /* 3382 * This filter shrinks M channels into N channels. 3383 * Extra channels are discarded. 3384 */ 3385 static void 3386 audio_track_chmix_shrink(audio_filter_arg_t *arg) 3387 { 3388 const aint_t *s; 3389 aint_t *d; 3390 u_int i; 3391 u_int ch; 3392 3393 DIAGNOSTIC_filter_arg(arg); 3394 3395 s = arg->src; 3396 d = arg->dst; 3397 3398 for (i = 0; i < arg->count; i++) { 3399 for (ch = 0; ch < arg->dstfmt->channels; ch++) { 3400 *d++ = s[ch]; 3401 } 3402 s += arg->srcfmt->channels; 3403 } 3404 } 3405 3406 /* 3407 * This filter expands M channels into N channels. 3408 * Silence is inserted for missing channels. 3409 */ 3410 static void 3411 audio_track_chmix_expand(audio_filter_arg_t *arg) 3412 { 3413 const aint_t *s; 3414 aint_t *d; 3415 u_int i; 3416 u_int ch; 3417 u_int srcchannels; 3418 u_int dstchannels; 3419 3420 DIAGNOSTIC_filter_arg(arg); 3421 3422 s = arg->src; 3423 d = arg->dst; 3424 3425 srcchannels = arg->srcfmt->channels; 3426 dstchannels = arg->dstfmt->channels; 3427 for (i = 0; i < arg->count; i++) { 3428 for (ch = 0; ch < srcchannels; ch++) { 3429 *d++ = *s++; 3430 } 3431 for (; ch < dstchannels; ch++) { 3432 *d++ = 0; 3433 } 3434 } 3435 } 3436 3437 /* 3438 * This filter performs frequency conversion (up sampling). 3439 * It uses linear interpolation. 3440 */ 3441 static void 3442 audio_track_freq_up(audio_filter_arg_t *arg) 3443 { 3444 audio_track_t *track; 3445 audio_ring_t *src; 3446 audio_ring_t *dst; 3447 const aint_t *s; 3448 aint_t *d; 3449 aint_t prev[AUDIO_MAX_CHANNELS]; 3450 aint_t curr[AUDIO_MAX_CHANNELS]; 3451 aint_t grad[AUDIO_MAX_CHANNELS]; 3452 u_int i; 3453 u_int t; 3454 u_int step; 3455 u_int channels; 3456 u_int ch; 3457 int srcused; 3458 3459 track = arg->context; 3460 KASSERT(track); 3461 src = &track->freq.srcbuf; 3462 dst = track->freq.dst; 3463 DIAGNOSTIC_ring(dst); 3464 DIAGNOSTIC_ring(src); 3465 KASSERT(src->used > 0); 3466 KASSERT(src->fmt.channels == dst->fmt.channels); 3467 KASSERT(src->head % track->mixer->frames_per_block == 0); 3468 3469 s = arg->src; 3470 d = arg->dst; 3471 3472 /* 3473 * In order to faciliate interpolation for each block, slide (delay) 3474 * input by one sample. As a result, strictly speaking, the output 3475 * phase is delayed by 1/dstfreq. However, I believe there is no 3476 * observable impact. 3477 * 3478 * Example) 3479 * srcfreq:dstfreq = 1:3 3480 * 3481 * A - - 3482 * | 3483 * | 3484 * | B - - 3485 * +-----+-----> input timeframe 3486 * 0 1 3487 * 3488 * 0 1 3489 * +-----+-----> input timeframe 3490 * | A 3491 * | x x 3492 * | x x 3493 * x (B) 3494 * +-+-+-+-+-+-> output timeframe 3495 * 0 1 2 3 4 5 3496 */ 3497 3498 /* Last samples in previous block */ 3499 channels = src->fmt.channels; 3500 for (ch = 0; ch < channels; ch++) { 3501 prev[ch] = track->freq_prev[ch]; 3502 curr[ch] = track->freq_curr[ch]; 3503 grad[ch] = curr[ch] - prev[ch]; 3504 } 3505 3506 step = track->freq_step; 3507 t = track->freq_current; 3508 //#define FREQ_DEBUG 3509 #if defined(FREQ_DEBUG) 3510 #define PRINTF(fmt...) printf(fmt) 3511 #else 3512 #define PRINTF(fmt...) do { } while (0) 3513 #endif 3514 srcused = src->used; 3515 PRINTF("upstart step=%d leap=%d", step, track->freq_leap); 3516 PRINTF(" srcused=%d arg->count=%u", src->used, arg->count); 3517 PRINTF(" prev=%d curr=%d grad=%d", prev[0], curr[0], grad[0]); 3518 PRINTF(" t=%d\n", t); 3519 3520 for (i = 0; i < arg->count; i++) { 3521 PRINTF("i=%d t=%5d", i, t); 3522 if (t >= 65536) { 3523 for (ch = 0; ch < channels; ch++) { 3524 prev[ch] = curr[ch]; 3525 curr[ch] = *s++; 3526 grad[ch] = curr[ch] - prev[ch]; 3527 } 3528 PRINTF(" prev=%d s[%d]=%d", 3529 prev[0], src->used - srcused, curr[0]); 3530 3531 /* Update */ 3532 t -= 65536; 3533 srcused--; 3534 if (srcused < 0) { 3535 PRINTF(" break\n"); 3536 break; 3537 } 3538 } 3539 3540 for (ch = 0; ch < channels; ch++) { 3541 *d++ = prev[ch] + (aint2_t)grad[ch] * t / 65536; 3542 #if defined(FREQ_DEBUG) 3543 if (ch == 0) 3544 printf(" t=%5d *d=%d", t, d[-1]); 3545 #endif 3546 } 3547 t += step; 3548 3549 PRINTF("\n"); 3550 } 3551 PRINTF("end prev=%d curr=%d\n", prev[0], curr[0]); 3552 3553 auring_take(src, src->used); 3554 auring_push(dst, i); 3555 3556 /* Adjust */ 3557 t += track->freq_leap; 3558 3559 track->freq_current = t; 3560 for (ch = 0; ch < channels; ch++) { 3561 track->freq_prev[ch] = prev[ch]; 3562 track->freq_curr[ch] = curr[ch]; 3563 } 3564 } 3565 3566 /* 3567 * This filter performs frequency conversion (down sampling). 3568 * It uses simple thinning. 3569 */ 3570 static void 3571 audio_track_freq_down(audio_filter_arg_t *arg) 3572 { 3573 audio_track_t *track; 3574 audio_ring_t *src; 3575 audio_ring_t *dst; 3576 const aint_t *s0; 3577 aint_t *d; 3578 u_int i; 3579 u_int t; 3580 u_int step; 3581 u_int ch; 3582 u_int channels; 3583 3584 track = arg->context; 3585 KASSERT(track); 3586 src = &track->freq.srcbuf; 3587 dst = track->freq.dst; 3588 3589 DIAGNOSTIC_ring(dst); 3590 DIAGNOSTIC_ring(src); 3591 KASSERT(src->used > 0); 3592 KASSERT(src->fmt.channels == dst->fmt.channels); 3593 KASSERTMSG(src->head % track->mixer->frames_per_block == 0, 3594 "src->head=%d fpb=%d", 3595 src->head, track->mixer->frames_per_block); 3596 3597 s0 = arg->src; 3598 d = arg->dst; 3599 t = track->freq_current; 3600 step = track->freq_step; 3601 channels = dst->fmt.channels; 3602 PRINTF("downstart step=%d leap=%d", step, track->freq_leap); 3603 PRINTF(" srcused=%d arg->count=%u", src->used, arg->count); 3604 PRINTF(" t=%d\n", t); 3605 3606 for (i = 0; i < arg->count && t / 65536 < src->used; i++) { 3607 const aint_t *s; 3608 PRINTF("i=%4d t=%10d", i, t); 3609 s = s0 + (t / 65536) * channels; 3610 PRINTF(" s=%5ld", (s - s0) / channels); 3611 for (ch = 0; ch < channels; ch++) { 3612 if (ch == 0) PRINTF(" *s=%d", s[ch]); 3613 *d++ = s[ch]; 3614 } 3615 PRINTF("\n"); 3616 t += step; 3617 } 3618 t += track->freq_leap; 3619 PRINTF("end t=%d\n", t); 3620 auring_take(src, src->used); 3621 auring_push(dst, i); 3622 track->freq_current = t % 65536; 3623 } 3624 3625 /* 3626 * Creates track and returns it. 3627 */ 3628 audio_track_t * 3629 audio_track_create(struct audio_softc *sc, audio_trackmixer_t *mixer) 3630 { 3631 audio_track_t *track; 3632 static int newid = 0; 3633 3634 track = kmem_zalloc(sizeof(*track), KM_SLEEP); 3635 3636 track->id = newid++; 3637 track->mixer = mixer; 3638 track->mode = mixer->mode; 3639 3640 /* Do TRACE after id is assigned. */ 3641 TRACET(3, track, "for %s", 3642 mixer->mode == AUMODE_PLAY ? "playback" : "recording"); 3643 3644 #if defined(AUDIO_SUPPORT_TRACK_VOLUME) 3645 track->volume = 256; 3646 #endif 3647 for (int i = 0; i < AUDIO_MAX_CHANNELS; i++) { 3648 track->ch_volume[i] = 256; 3649 } 3650 3651 return track; 3652 } 3653 3654 /* 3655 * Release all resources of the track and track itself. 3656 * track must not be NULL. Don't specify the track within the file 3657 * structure linked from sc->sc_files. 3658 */ 3659 static void 3660 audio_track_destroy(audio_track_t *track) 3661 { 3662 3663 KASSERT(track); 3664 3665 audio_free_usrbuf(track); 3666 audio_free(track->codec.srcbuf.mem); 3667 audio_free(track->chvol.srcbuf.mem); 3668 audio_free(track->chmix.srcbuf.mem); 3669 audio_free(track->freq.srcbuf.mem); 3670 audio_free(track->outbuf.mem); 3671 3672 kmem_free(track, sizeof(*track)); 3673 } 3674 3675 /* 3676 * It returns encoding conversion filter according to src and dst format. 3677 * If it is not a convertible pair, it returns NULL. Either src or dst 3678 * must be internal format. 3679 */ 3680 static audio_filter_t 3681 audio_track_get_codec(audio_track_t *track, const audio_format2_t *src, 3682 const audio_format2_t *dst) 3683 { 3684 3685 if (audio_format2_is_internal(src)) { 3686 if (dst->encoding == AUDIO_ENCODING_ULAW) { 3687 return audio_internal_to_mulaw; 3688 } else if (dst->encoding == AUDIO_ENCODING_ALAW) { 3689 return audio_internal_to_alaw; 3690 } else if (audio_format2_is_linear(dst)) { 3691 switch (dst->stride) { 3692 case 8: 3693 return audio_internal_to_linear8; 3694 case 16: 3695 return audio_internal_to_linear16; 3696 #if defined(AUDIO_SUPPORT_LINEAR24) 3697 case 24: 3698 return audio_internal_to_linear24; 3699 #endif 3700 case 32: 3701 return audio_internal_to_linear32; 3702 default: 3703 TRACET(1, track, "unsupported %s stride %d", 3704 "dst", dst->stride); 3705 goto abort; 3706 } 3707 } 3708 } else if (audio_format2_is_internal(dst)) { 3709 if (src->encoding == AUDIO_ENCODING_ULAW) { 3710 return audio_mulaw_to_internal; 3711 } else if (src->encoding == AUDIO_ENCODING_ALAW) { 3712 return audio_alaw_to_internal; 3713 } else if (audio_format2_is_linear(src)) { 3714 switch (src->stride) { 3715 case 8: 3716 return audio_linear8_to_internal; 3717 case 16: 3718 return audio_linear16_to_internal; 3719 #if defined(AUDIO_SUPPORT_LINEAR24) 3720 case 24: 3721 return audio_linear24_to_internal; 3722 #endif 3723 case 32: 3724 return audio_linear32_to_internal; 3725 default: 3726 TRACET(1, track, "unsupported %s stride %d", 3727 "src", src->stride); 3728 goto abort; 3729 } 3730 } 3731 } 3732 3733 TRACET(1, track, "unsupported encoding"); 3734 abort: 3735 #if defined(AUDIO_DEBUG) 3736 if (audiodebug >= 2) { 3737 char buf[100]; 3738 audio_format2_tostr(buf, sizeof(buf), src); 3739 TRACET(2, track, "src %s", buf); 3740 audio_format2_tostr(buf, sizeof(buf), dst); 3741 TRACET(2, track, "dst %s", buf); 3742 } 3743 #endif 3744 return NULL; 3745 } 3746 3747 /* 3748 * Initialize the codec stage of this track as necessary. 3749 * If successful, it initializes the codec stage as necessary, stores updated 3750 * last_dst in *last_dstp in any case, and returns 0. 3751 * Otherwise, it returns errno without modifying *last_dstp. 3752 */ 3753 static int 3754 audio_track_init_codec(audio_track_t *track, audio_ring_t **last_dstp) 3755 { 3756 struct audio_softc *sc; 3757 audio_ring_t *last_dst; 3758 audio_ring_t *srcbuf; 3759 audio_format2_t *srcfmt; 3760 audio_format2_t *dstfmt; 3761 audio_filter_arg_t *arg; 3762 u_int len; 3763 int error; 3764 3765 KASSERT(track); 3766 3767 sc = track->mixer->sc; 3768 last_dst = *last_dstp; 3769 dstfmt = &last_dst->fmt; 3770 srcfmt = &track->inputfmt; 3771 srcbuf = &track->codec.srcbuf; 3772 error = 0; 3773 3774 if (srcfmt->encoding != dstfmt->encoding 3775 || srcfmt->precision != dstfmt->precision 3776 || srcfmt->stride != dstfmt->stride) { 3777 track->codec.dst = last_dst; 3778 3779 srcbuf->fmt = *dstfmt; 3780 srcbuf->fmt.encoding = srcfmt->encoding; 3781 srcbuf->fmt.precision = srcfmt->precision; 3782 srcbuf->fmt.stride = srcfmt->stride; 3783 3784 track->codec.filter = audio_track_get_codec(track, 3785 &srcbuf->fmt, dstfmt); 3786 if (track->codec.filter == NULL) { 3787 error = EINVAL; 3788 goto abort; 3789 } 3790 3791 srcbuf->head = 0; 3792 srcbuf->used = 0; 3793 srcbuf->capacity = frame_per_block(track->mixer, &srcbuf->fmt); 3794 len = auring_bytelen(srcbuf); 3795 srcbuf->mem = audio_realloc(srcbuf->mem, len); 3796 if (srcbuf->mem == NULL) { 3797 device_printf(sc->sc_dev, "%s: malloc(%d) failed\n", 3798 __func__, len); 3799 error = ENOMEM; 3800 goto abort; 3801 } 3802 3803 arg = &track->codec.arg; 3804 arg->srcfmt = &srcbuf->fmt; 3805 arg->dstfmt = dstfmt; 3806 arg->context = NULL; 3807 3808 *last_dstp = srcbuf; 3809 return 0; 3810 } 3811 3812 abort: 3813 track->codec.filter = NULL; 3814 audio_free(srcbuf->mem); 3815 return error; 3816 } 3817 3818 /* 3819 * Initialize the chvol stage of this track as necessary. 3820 * If successful, it initializes the chvol stage as necessary, stores updated 3821 * last_dst in *last_dstp in any case, and returns 0. 3822 * Otherwise, it returns errno without modifying *last_dstp. 3823 */ 3824 static int 3825 audio_track_init_chvol(audio_track_t *track, audio_ring_t **last_dstp) 3826 { 3827 struct audio_softc *sc; 3828 audio_ring_t *last_dst; 3829 audio_ring_t *srcbuf; 3830 audio_format2_t *srcfmt; 3831 audio_format2_t *dstfmt; 3832 audio_filter_arg_t *arg; 3833 u_int len; 3834 int error; 3835 3836 KASSERT(track); 3837 3838 sc = track->mixer->sc; 3839 last_dst = *last_dstp; 3840 dstfmt = &last_dst->fmt; 3841 srcfmt = &track->inputfmt; 3842 srcbuf = &track->chvol.srcbuf; 3843 error = 0; 3844 3845 /* Check whether channel volume conversion is necessary. */ 3846 bool use_chvol = false; 3847 for (int ch = 0; ch < srcfmt->channels; ch++) { 3848 if (track->ch_volume[ch] != 256) { 3849 use_chvol = true; 3850 break; 3851 } 3852 } 3853 3854 if (use_chvol == true) { 3855 track->chvol.dst = last_dst; 3856 track->chvol.filter = audio_track_chvol; 3857 3858 srcbuf->fmt = *dstfmt; 3859 /* no format conversion occurs */ 3860 3861 srcbuf->head = 0; 3862 srcbuf->used = 0; 3863 srcbuf->capacity = frame_per_block(track->mixer, &srcbuf->fmt); 3864 len = auring_bytelen(srcbuf); 3865 srcbuf->mem = audio_realloc(srcbuf->mem, len); 3866 if (srcbuf->mem == NULL) { 3867 device_printf(sc->sc_dev, "%s: malloc(%d) failed\n", 3868 __func__, len); 3869 error = ENOMEM; 3870 goto abort; 3871 } 3872 3873 arg = &track->chvol.arg; 3874 arg->srcfmt = &srcbuf->fmt; 3875 arg->dstfmt = dstfmt; 3876 arg->context = track->ch_volume; 3877 3878 *last_dstp = srcbuf; 3879 return 0; 3880 } 3881 3882 abort: 3883 track->chvol.filter = NULL; 3884 audio_free(srcbuf->mem); 3885 return error; 3886 } 3887 3888 /* 3889 * Initialize the chmix stage of this track as necessary. 3890 * If successful, it initializes the chmix stage as necessary, stores updated 3891 * last_dst in *last_dstp in any case, and returns 0. 3892 * Otherwise, it returns errno without modifying *last_dstp. 3893 */ 3894 static int 3895 audio_track_init_chmix(audio_track_t *track, audio_ring_t **last_dstp) 3896 { 3897 struct audio_softc *sc; 3898 audio_ring_t *last_dst; 3899 audio_ring_t *srcbuf; 3900 audio_format2_t *srcfmt; 3901 audio_format2_t *dstfmt; 3902 audio_filter_arg_t *arg; 3903 u_int srcch; 3904 u_int dstch; 3905 u_int len; 3906 int error; 3907 3908 KASSERT(track); 3909 3910 sc = track->mixer->sc; 3911 last_dst = *last_dstp; 3912 dstfmt = &last_dst->fmt; 3913 srcfmt = &track->inputfmt; 3914 srcbuf = &track->chmix.srcbuf; 3915 error = 0; 3916 3917 srcch = srcfmt->channels; 3918 dstch = dstfmt->channels; 3919 if (srcch != dstch) { 3920 track->chmix.dst = last_dst; 3921 3922 if (srcch >= 2 && dstch == 1) { 3923 track->chmix.filter = audio_track_chmix_mixLR; 3924 } else if (srcch == 1 && dstch >= 2) { 3925 track->chmix.filter = audio_track_chmix_dupLR; 3926 } else if (srcch > dstch) { 3927 track->chmix.filter = audio_track_chmix_shrink; 3928 } else { 3929 track->chmix.filter = audio_track_chmix_expand; 3930 } 3931 3932 srcbuf->fmt = *dstfmt; 3933 srcbuf->fmt.channels = srcch; 3934 3935 srcbuf->head = 0; 3936 srcbuf->used = 0; 3937 /* XXX The buffer size should be able to calculate. */ 3938 srcbuf->capacity = frame_per_block(track->mixer, &srcbuf->fmt); 3939 len = auring_bytelen(srcbuf); 3940 srcbuf->mem = audio_realloc(srcbuf->mem, len); 3941 if (srcbuf->mem == NULL) { 3942 device_printf(sc->sc_dev, "%s: malloc(%d) failed\n", 3943 __func__, len); 3944 error = ENOMEM; 3945 goto abort; 3946 } 3947 3948 arg = &track->chmix.arg; 3949 arg->srcfmt = &srcbuf->fmt; 3950 arg->dstfmt = dstfmt; 3951 arg->context = NULL; 3952 3953 *last_dstp = srcbuf; 3954 return 0; 3955 } 3956 3957 abort: 3958 track->chmix.filter = NULL; 3959 audio_free(srcbuf->mem); 3960 return error; 3961 } 3962 3963 /* 3964 * Initialize the freq stage of this track as necessary. 3965 * If successful, it initializes the freq stage as necessary, stores updated 3966 * last_dst in *last_dstp in any case, and returns 0. 3967 * Otherwise, it returns errno without modifying *last_dstp. 3968 */ 3969 static int 3970 audio_track_init_freq(audio_track_t *track, audio_ring_t **last_dstp) 3971 { 3972 struct audio_softc *sc; 3973 audio_ring_t *last_dst; 3974 audio_ring_t *srcbuf; 3975 audio_format2_t *srcfmt; 3976 audio_format2_t *dstfmt; 3977 audio_filter_arg_t *arg; 3978 uint32_t srcfreq; 3979 uint32_t dstfreq; 3980 u_int dst_capacity; 3981 u_int mod; 3982 u_int len; 3983 int error; 3984 3985 KASSERT(track); 3986 3987 sc = track->mixer->sc; 3988 last_dst = *last_dstp; 3989 dstfmt = &last_dst->fmt; 3990 srcfmt = &track->inputfmt; 3991 srcbuf = &track->freq.srcbuf; 3992 error = 0; 3993 3994 srcfreq = srcfmt->sample_rate; 3995 dstfreq = dstfmt->sample_rate; 3996 if (srcfreq != dstfreq) { 3997 track->freq.dst = last_dst; 3998 3999 memset(track->freq_prev, 0, sizeof(track->freq_prev)); 4000 memset(track->freq_curr, 0, sizeof(track->freq_curr)); 4001 4002 /* freq_step is the ratio of src/dst when let dst 65536. */ 4003 track->freq_step = (uint64_t)srcfreq * 65536 / dstfreq; 4004 4005 dst_capacity = frame_per_block(track->mixer, dstfmt); 4006 mod = (uint64_t)srcfreq * 65536 % dstfreq; 4007 track->freq_leap = (mod * dst_capacity + dstfreq / 2) / dstfreq; 4008 4009 if (track->freq_step < 65536) { 4010 track->freq.filter = audio_track_freq_up; 4011 /* In order to carry at the first time. */ 4012 track->freq_current = 65536; 4013 } else { 4014 track->freq.filter = audio_track_freq_down; 4015 track->freq_current = 0; 4016 } 4017 4018 srcbuf->fmt = *dstfmt; 4019 srcbuf->fmt.sample_rate = srcfreq; 4020 4021 srcbuf->head = 0; 4022 srcbuf->used = 0; 4023 srcbuf->capacity = frame_per_block(track->mixer, &srcbuf->fmt); 4024 len = auring_bytelen(srcbuf); 4025 srcbuf->mem = audio_realloc(srcbuf->mem, len); 4026 if (srcbuf->mem == NULL) { 4027 device_printf(sc->sc_dev, "%s: malloc(%d) failed\n", 4028 __func__, len); 4029 error = ENOMEM; 4030 goto abort; 4031 } 4032 4033 arg = &track->freq.arg; 4034 arg->srcfmt = &srcbuf->fmt; 4035 arg->dstfmt = dstfmt;/*&last_dst->fmt;*/ 4036 arg->context = track; 4037 4038 *last_dstp = srcbuf; 4039 return 0; 4040 } 4041 4042 abort: 4043 track->freq.filter = NULL; 4044 audio_free(srcbuf->mem); 4045 return error; 4046 } 4047 4048 /* 4049 * When playing back: (e.g. if codec and freq stage are valid) 4050 * 4051 * write 4052 * | uiomove 4053 * v 4054 * usrbuf [...............] byte ring buffer (mmap-able) 4055 * | memcpy 4056 * v 4057 * codec.srcbuf[....] 1 block (ring) buffer <-- stage input 4058 * .dst ----+ 4059 * | convert 4060 * v 4061 * freq.srcbuf [....] 1 block (ring) buffer 4062 * .dst ----+ 4063 * | convert 4064 * v 4065 * outbuf [...............] NBLKOUT blocks ring buffer 4066 * 4067 * 4068 * When recording: 4069 * 4070 * freq.srcbuf [...............] NBLKOUT blocks ring buffer <-- stage input 4071 * .dst ----+ 4072 * | convert 4073 * v 4074 * codec.srcbuf[.....] 1 block (ring) buffer 4075 * .dst ----+ 4076 * | convert 4077 * v 4078 * outbuf [.....] 1 block (ring) buffer 4079 * | memcpy 4080 * v 4081 * usrbuf [...............] byte ring buffer (mmap-able *) 4082 * | uiomove 4083 * v 4084 * read 4085 * 4086 * *: usrbuf for recording is also mmap-able due to symmetry with 4087 * playback buffer, but for now mmap will never happen for recording. 4088 */ 4089 4090 /* 4091 * Set the userland format of this track. 4092 * usrfmt argument should be parameter verified with audio_check_params(). 4093 * It will release and reallocate all internal conversion buffers. 4094 * It returns 0 if successful. Otherwise it returns errno with clearing all 4095 * internal buffers. 4096 * It must be called without sc_intr_lock since uvm_* routines require non 4097 * intr_lock state. 4098 * It must be called with track lock held since it may release and reallocate 4099 * outbuf. 4100 */ 4101 static int 4102 audio_track_set_format(audio_track_t *track, audio_format2_t *usrfmt) 4103 { 4104 struct audio_softc *sc; 4105 u_int newbufsize; 4106 u_int oldblksize; 4107 u_int len; 4108 int error; 4109 4110 KASSERT(track); 4111 sc = track->mixer->sc; 4112 4113 /* usrbuf is the closest buffer to the userland. */ 4114 track->usrbuf.fmt = *usrfmt; 4115 4116 /* 4117 * For references, one block size (in 40msec) is: 4118 * 320 bytes = 204 blocks/64KB for mulaw/8kHz/1ch 4119 * 7680 bytes = 8 blocks/64KB for s16/48kHz/2ch 4120 * 30720 bytes = 90 KB/3blocks for s16/48kHz/8ch 4121 * 61440 bytes = 180 KB/3blocks for s16/96kHz/8ch 4122 * 245760 bytes = 720 KB/3blocks for s32/192kHz/8ch 4123 * 4124 * For example, 4125 * 1) If usrbuf_blksize = 7056 (s16/44.1k/2ch) and PAGE_SIZE = 8192, 4126 * newbufsize = rounddown(65536 / 7056) = 63504 4127 * newvsize = roundup2(63504, PAGE_SIZE) = 65536 4128 * Therefore it maps 8 * 8K pages and usrbuf->capacity = 63504. 4129 * 4130 * 2) If usrbuf_blksize = 7680 (s16/48k/2ch) and PAGE_SIZE = 4096, 4131 * newbufsize = rounddown(65536 / 7680) = 61440 4132 * newvsize = roundup2(61440, PAGE_SIZE) = 61440 (= 15 pages) 4133 * Therefore it maps 15 * 4K pages and usrbuf->capacity = 61440. 4134 */ 4135 oldblksize = track->usrbuf_blksize; 4136 track->usrbuf_blksize = frametobyte(&track->usrbuf.fmt, 4137 frame_per_block(track->mixer, &track->usrbuf.fmt)); 4138 track->usrbuf.head = 0; 4139 track->usrbuf.used = 0; 4140 newbufsize = MAX(track->usrbuf_blksize * AUMINNOBLK, 65536); 4141 newbufsize = rounddown(newbufsize, track->usrbuf_blksize); 4142 error = audio_realloc_usrbuf(track, newbufsize); 4143 if (error) { 4144 device_printf(sc->sc_dev, "malloc usrbuf(%d) failed\n", 4145 newbufsize); 4146 goto error; 4147 } 4148 4149 /* Recalc water mark. */ 4150 if (track->usrbuf_blksize != oldblksize) { 4151 if (audio_track_is_playback(track)) { 4152 /* Set high at 100%, low at 75%. */ 4153 track->usrbuf_usedhigh = track->usrbuf.capacity; 4154 track->usrbuf_usedlow = track->usrbuf.capacity * 3 / 4; 4155 } else { 4156 /* Set high at 100% minus 1block(?), low at 0% */ 4157 track->usrbuf_usedhigh = track->usrbuf.capacity - 4158 track->usrbuf_blksize; 4159 track->usrbuf_usedlow = 0; 4160 } 4161 } 4162 4163 /* Stage buffer */ 4164 audio_ring_t *last_dst = &track->outbuf; 4165 if (audio_track_is_playback(track)) { 4166 /* On playback, initialize from the mixer side in order. */ 4167 track->inputfmt = *usrfmt; 4168 track->outbuf.fmt = track->mixer->track_fmt; 4169 4170 if ((error = audio_track_init_freq(track, &last_dst)) != 0) 4171 goto error; 4172 if ((error = audio_track_init_chmix(track, &last_dst)) != 0) 4173 goto error; 4174 if ((error = audio_track_init_chvol(track, &last_dst)) != 0) 4175 goto error; 4176 if ((error = audio_track_init_codec(track, &last_dst)) != 0) 4177 goto error; 4178 } else { 4179 /* On recording, initialize from userland side in order. */ 4180 track->inputfmt = track->mixer->track_fmt; 4181 track->outbuf.fmt = *usrfmt; 4182 4183 if ((error = audio_track_init_codec(track, &last_dst)) != 0) 4184 goto error; 4185 if ((error = audio_track_init_chvol(track, &last_dst)) != 0) 4186 goto error; 4187 if ((error = audio_track_init_chmix(track, &last_dst)) != 0) 4188 goto error; 4189 if ((error = audio_track_init_freq(track, &last_dst)) != 0) 4190 goto error; 4191 } 4192 #if 0 4193 /* debug */ 4194 if (track->freq.filter) { 4195 audio_print_format2("freq src", &track->freq.srcbuf.fmt); 4196 audio_print_format2("freq dst", &track->freq.dst->fmt); 4197 } 4198 if (track->chmix.filter) { 4199 audio_print_format2("chmix src", &track->chmix.srcbuf.fmt); 4200 audio_print_format2("chmix dst", &track->chmix.dst->fmt); 4201 } 4202 if (track->chvol.filter) { 4203 audio_print_format2("chvol src", &track->chvol.srcbuf.fmt); 4204 audio_print_format2("chvol dst", &track->chvol.dst->fmt); 4205 } 4206 if (track->codec.filter) { 4207 audio_print_format2("codec src", &track->codec.srcbuf.fmt); 4208 audio_print_format2("codec dst", &track->codec.dst->fmt); 4209 } 4210 #endif 4211 4212 /* Stage input buffer */ 4213 track->input = last_dst; 4214 4215 /* 4216 * On the recording track, make the first stage a ring buffer. 4217 * XXX is there a better way? 4218 */ 4219 if (audio_track_is_record(track)) { 4220 track->input->capacity = NBLKOUT * 4221 frame_per_block(track->mixer, &track->input->fmt); 4222 len = auring_bytelen(track->input); 4223 track->input->mem = audio_realloc(track->input->mem, len); 4224 if (track->input->mem == NULL) { 4225 device_printf(sc->sc_dev, "malloc input(%d) failed\n", 4226 len); 4227 error = ENOMEM; 4228 goto error; 4229 } 4230 } 4231 4232 /* 4233 * Output buffer. 4234 * On the playback track, its capacity is NBLKOUT blocks. 4235 * On the recording track, its capacity is 1 block. 4236 */ 4237 track->outbuf.head = 0; 4238 track->outbuf.used = 0; 4239 track->outbuf.capacity = frame_per_block(track->mixer, 4240 &track->outbuf.fmt); 4241 if (audio_track_is_playback(track)) 4242 track->outbuf.capacity *= NBLKOUT; 4243 len = auring_bytelen(&track->outbuf); 4244 track->outbuf.mem = audio_realloc(track->outbuf.mem, len); 4245 if (track->outbuf.mem == NULL) { 4246 device_printf(sc->sc_dev, "malloc outbuf(%d) failed\n", len); 4247 error = ENOMEM; 4248 goto error; 4249 } 4250 4251 #if defined(AUDIO_DEBUG) 4252 if (audiodebug >= 3) { 4253 struct audio_track_debugbuf m; 4254 4255 memset(&m, 0, sizeof(m)); 4256 snprintf(m.outbuf, sizeof(m.outbuf), " out=%d", 4257 track->outbuf.capacity * frametobyte(&track->outbuf.fmt,1)); 4258 if (track->freq.filter) 4259 snprintf(m.freq, sizeof(m.freq), " freq=%d", 4260 track->freq.srcbuf.capacity * 4261 frametobyte(&track->freq.srcbuf.fmt, 1)); 4262 if (track->chmix.filter) 4263 snprintf(m.chmix, sizeof(m.chmix), " chmix=%d", 4264 track->chmix.srcbuf.capacity * 4265 frametobyte(&track->chmix.srcbuf.fmt, 1)); 4266 if (track->chvol.filter) 4267 snprintf(m.chvol, sizeof(m.chvol), " chvol=%d", 4268 track->chvol.srcbuf.capacity * 4269 frametobyte(&track->chvol.srcbuf.fmt, 1)); 4270 if (track->codec.filter) 4271 snprintf(m.codec, sizeof(m.codec), " codec=%d", 4272 track->codec.srcbuf.capacity * 4273 frametobyte(&track->codec.srcbuf.fmt, 1)); 4274 snprintf(m.usrbuf, sizeof(m.usrbuf), 4275 " usr=%d", track->usrbuf.capacity); 4276 4277 if (audio_track_is_playback(track)) { 4278 TRACET(0, track, "bufsize%s%s%s%s%s%s", 4279 m.outbuf, m.freq, m.chmix, 4280 m.chvol, m.codec, m.usrbuf); 4281 } else { 4282 TRACET(0, track, "bufsize%s%s%s%s%s%s", 4283 m.freq, m.chmix, m.chvol, 4284 m.codec, m.outbuf, m.usrbuf); 4285 } 4286 } 4287 #endif 4288 return 0; 4289 4290 error: 4291 audio_free_usrbuf(track); 4292 audio_free(track->codec.srcbuf.mem); 4293 audio_free(track->chvol.srcbuf.mem); 4294 audio_free(track->chmix.srcbuf.mem); 4295 audio_free(track->freq.srcbuf.mem); 4296 audio_free(track->outbuf.mem); 4297 return error; 4298 } 4299 4300 /* 4301 * Fill silence frames (as the internal format) up to 1 block 4302 * if the ring is not empty and less than 1 block. 4303 * It returns the number of appended frames. 4304 */ 4305 static int 4306 audio_append_silence(audio_track_t *track, audio_ring_t *ring) 4307 { 4308 int fpb; 4309 int n; 4310 4311 KASSERT(track); 4312 KASSERT(audio_format2_is_internal(&ring->fmt)); 4313 4314 /* XXX is n correct? */ 4315 /* XXX memset uses frametobyte()? */ 4316 4317 if (ring->used == 0) 4318 return 0; 4319 4320 fpb = frame_per_block(track->mixer, &ring->fmt); 4321 if (ring->used >= fpb) 4322 return 0; 4323 4324 n = (ring->capacity - ring->used) % fpb; 4325 4326 KASSERT(auring_get_contig_free(ring) >= n); 4327 4328 memset(auring_tailptr_aint(ring), 0, 4329 n * ring->fmt.channels * sizeof(aint_t)); 4330 auring_push(ring, n); 4331 return n; 4332 } 4333 4334 /* 4335 * Execute the conversion stage. 4336 * It prepares arg from this stage and executes stage->filter. 4337 * It must be called only if stage->filter is not NULL. 4338 * 4339 * For stages other than frequency conversion, the function increments 4340 * src and dst counters here. For frequency conversion stage, on the 4341 * other hand, the function does not touch src and dst counters and 4342 * filter side has to increment them. 4343 */ 4344 static void 4345 audio_apply_stage(audio_track_t *track, audio_stage_t *stage, bool isfreq) 4346 { 4347 audio_filter_arg_t *arg; 4348 int srccount; 4349 int dstcount; 4350 int count; 4351 4352 KASSERT(track); 4353 KASSERT(stage->filter); 4354 4355 srccount = auring_get_contig_used(&stage->srcbuf); 4356 dstcount = auring_get_contig_free(stage->dst); 4357 4358 if (isfreq) { 4359 KASSERTMSG(srccount > 0, "freq but srccount == %d", srccount); 4360 count = uimin(dstcount, track->mixer->frames_per_block); 4361 } else { 4362 count = uimin(srccount, dstcount); 4363 } 4364 4365 if (count > 0) { 4366 arg = &stage->arg; 4367 arg->src = auring_headptr(&stage->srcbuf); 4368 arg->dst = auring_tailptr(stage->dst); 4369 arg->count = count; 4370 4371 stage->filter(arg); 4372 4373 if (!isfreq) { 4374 auring_take(&stage->srcbuf, count); 4375 auring_push(stage->dst, count); 4376 } 4377 } 4378 } 4379 4380 /* 4381 * Produce output buffer for playback from user input buffer. 4382 * It must be called only if usrbuf is not empty and outbuf is 4383 * available at least one free block. 4384 */ 4385 static void 4386 audio_track_play(audio_track_t *track) 4387 { 4388 audio_ring_t *usrbuf; 4389 audio_ring_t *input; 4390 int count; 4391 int framesize; 4392 int bytes; 4393 u_int dropcount; 4394 4395 KASSERT(track); 4396 KASSERT(track->lock); 4397 TRACET(4, track, "start pstate=%d", track->pstate); 4398 4399 /* At this point usrbuf must not be empty. */ 4400 KASSERT(track->usrbuf.used > 0); 4401 /* Also, outbuf must be available at least one block. */ 4402 count = auring_get_contig_free(&track->outbuf); 4403 KASSERTMSG(count >= frame_per_block(track->mixer, &track->outbuf.fmt), 4404 "count=%d fpb=%d", 4405 count, frame_per_block(track->mixer, &track->outbuf.fmt)); 4406 4407 /* XXX TODO: is this necessary for now? */ 4408 int track_count_0 = track->outbuf.used; 4409 4410 usrbuf = &track->usrbuf; 4411 input = track->input; 4412 dropcount = 0; 4413 4414 /* 4415 * framesize is always 1 byte or more since all formats supported as 4416 * usrfmt(=input) have 8bit or more stride. 4417 */ 4418 framesize = frametobyte(&input->fmt, 1); 4419 KASSERT(framesize >= 1); 4420 4421 /* The next stage of usrbuf (=input) must be available. */ 4422 KASSERT(auring_get_contig_free(input) > 0); 4423 4424 /* 4425 * Copy usrbuf up to 1block to input buffer. 4426 * count is the number of frames to copy from usrbuf. 4427 * bytes is the number of bytes to copy from usrbuf. However it is 4428 * not copied less than one frame. 4429 */ 4430 count = uimin(usrbuf->used, track->usrbuf_blksize) / framesize; 4431 bytes = count * framesize; 4432 4433 /* 4434 * If bytes is less than one block, 4435 * if not draining, buffer is not filled so return. 4436 * if draining, fall through. 4437 */ 4438 if (count < track->usrbuf_blksize / framesize) { 4439 dropcount = track->usrbuf_blksize / framesize - count; 4440 4441 if (track->pstate != AUDIO_STATE_DRAINING) { 4442 /* Wait until filled. */ 4443 TRACET(4, track, "not enough; return"); 4444 return; 4445 } 4446 } 4447 4448 track->usrbuf_stamp += bytes; 4449 4450 if (usrbuf->head + bytes < usrbuf->capacity) { 4451 memcpy((uint8_t *)input->mem + auring_tail(input) * framesize, 4452 (uint8_t *)usrbuf->mem + usrbuf->head, 4453 bytes); 4454 auring_push(input, count); 4455 auring_take(usrbuf, bytes); 4456 } else { 4457 int bytes1; 4458 int bytes2; 4459 4460 bytes1 = auring_get_contig_used(usrbuf); 4461 KASSERT(bytes1 % framesize == 0); 4462 memcpy((uint8_t *)input->mem + auring_tail(input) * framesize, 4463 (uint8_t *)usrbuf->mem + usrbuf->head, 4464 bytes1); 4465 auring_push(input, bytes1 / framesize); 4466 auring_take(usrbuf, bytes1); 4467 4468 bytes2 = bytes - bytes1; 4469 memcpy((uint8_t *)input->mem + auring_tail(input) * framesize, 4470 (uint8_t *)usrbuf->mem + usrbuf->head, 4471 bytes2); 4472 auring_push(input, bytes2 / framesize); 4473 auring_take(usrbuf, bytes2); 4474 } 4475 4476 /* Encoding conversion */ 4477 if (track->codec.filter) 4478 audio_apply_stage(track, &track->codec, false); 4479 4480 /* Channel volume */ 4481 if (track->chvol.filter) 4482 audio_apply_stage(track, &track->chvol, false); 4483 4484 /* Channel mix */ 4485 if (track->chmix.filter) 4486 audio_apply_stage(track, &track->chmix, false); 4487 4488 /* Frequency conversion */ 4489 /* 4490 * Since the frequency conversion needs correction for each block, 4491 * it rounds up to 1 block. 4492 */ 4493 if (track->freq.filter) { 4494 int n; 4495 n = audio_append_silence(track, &track->freq.srcbuf); 4496 if (n > 0) { 4497 TRACET(4, track, 4498 "freq.srcbuf add silence %d -> %d/%d/%d", 4499 n, 4500 track->freq.srcbuf.head, 4501 track->freq.srcbuf.used, 4502 track->freq.srcbuf.capacity); 4503 } 4504 if (track->freq.srcbuf.used > 0) { 4505 audio_apply_stage(track, &track->freq, true); 4506 } 4507 } 4508 4509 if (dropcount != 0) { 4510 /* 4511 * Clear all conversion buffer pointer if the conversion was 4512 * not exactly one block. These conversion stage buffers are 4513 * certainly circular buffers because of symmetry with the 4514 * previous and next stage buffer. However, since they are 4515 * treated as simple contiguous buffers in operation, so head 4516 * always should point 0. This may happen during drain-age. 4517 */ 4518 TRACET(4, track, "reset stage"); 4519 if (track->codec.filter) { 4520 KASSERT(track->codec.srcbuf.used == 0); 4521 track->codec.srcbuf.head = 0; 4522 } 4523 if (track->chvol.filter) { 4524 KASSERT(track->chvol.srcbuf.used == 0); 4525 track->chvol.srcbuf.head = 0; 4526 } 4527 if (track->chmix.filter) { 4528 KASSERT(track->chmix.srcbuf.used == 0); 4529 track->chmix.srcbuf.head = 0; 4530 } 4531 if (track->freq.filter) { 4532 KASSERT(track->freq.srcbuf.used == 0); 4533 track->freq.srcbuf.head = 0; 4534 } 4535 } 4536 4537 if (track->input == &track->outbuf) { 4538 track->outputcounter = track->inputcounter; 4539 } else { 4540 track->outputcounter += track->outbuf.used - track_count_0; 4541 } 4542 4543 #if defined(AUDIO_DEBUG) 4544 if (audiodebug >= 3) { 4545 struct audio_track_debugbuf m; 4546 audio_track_bufstat(track, &m); 4547 TRACET(0, track, "end%s%s%s%s%s%s", 4548 m.outbuf, m.freq, m.chvol, m.chmix, m.codec, m.usrbuf); 4549 } 4550 #endif 4551 } 4552 4553 /* 4554 * Produce user output buffer for recording from input buffer. 4555 */ 4556 static void 4557 audio_track_record(audio_track_t *track) 4558 { 4559 audio_ring_t *outbuf; 4560 audio_ring_t *usrbuf; 4561 int count; 4562 int bytes; 4563 int framesize; 4564 4565 KASSERT(track); 4566 KASSERT(track->lock); 4567 4568 /* Number of frames to process */ 4569 count = auring_get_contig_used(track->input); 4570 count = uimin(count, track->mixer->frames_per_block); 4571 if (count == 0) { 4572 TRACET(4, track, "count == 0"); 4573 return; 4574 } 4575 4576 /* Frequency conversion */ 4577 if (track->freq.filter) { 4578 if (track->freq.srcbuf.used > 0) { 4579 audio_apply_stage(track, &track->freq, true); 4580 /* XXX should input of freq be from beginning of buf? */ 4581 } 4582 } 4583 4584 /* Channel mix */ 4585 if (track->chmix.filter) 4586 audio_apply_stage(track, &track->chmix, false); 4587 4588 /* Channel volume */ 4589 if (track->chvol.filter) 4590 audio_apply_stage(track, &track->chvol, false); 4591 4592 /* Encoding conversion */ 4593 if (track->codec.filter) 4594 audio_apply_stage(track, &track->codec, false); 4595 4596 /* Copy outbuf to usrbuf */ 4597 outbuf = &track->outbuf; 4598 usrbuf = &track->usrbuf; 4599 /* 4600 * framesize is always 1 byte or more since all formats supported 4601 * as usrfmt(=output) have 8bit or more stride. 4602 */ 4603 framesize = frametobyte(&outbuf->fmt, 1); 4604 KASSERT(framesize >= 1); 4605 /* 4606 * count is the number of frames to copy to usrbuf. 4607 * bytes is the number of bytes to copy to usrbuf. 4608 */ 4609 count = outbuf->used; 4610 count = uimin(count, 4611 (track->usrbuf_usedhigh - usrbuf->used) / framesize); 4612 bytes = count * framesize; 4613 if (auring_tail(usrbuf) + bytes < usrbuf->capacity) { 4614 memcpy((uint8_t *)usrbuf->mem + auring_tail(usrbuf), 4615 (uint8_t *)outbuf->mem + outbuf->head * framesize, 4616 bytes); 4617 auring_push(usrbuf, bytes); 4618 auring_take(outbuf, count); 4619 } else { 4620 int bytes1; 4621 int bytes2; 4622 4623 bytes1 = auring_get_contig_used(usrbuf); 4624 KASSERT(bytes1 % framesize == 0); 4625 memcpy((uint8_t *)usrbuf->mem + auring_tail(usrbuf), 4626 (uint8_t *)outbuf->mem + outbuf->head * framesize, 4627 bytes1); 4628 auring_push(usrbuf, bytes1); 4629 auring_take(outbuf, bytes1 / framesize); 4630 4631 bytes2 = bytes - bytes1; 4632 memcpy((uint8_t *)usrbuf->mem + auring_tail(usrbuf), 4633 (uint8_t *)outbuf->mem + outbuf->head * framesize, 4634 bytes2); 4635 auring_push(usrbuf, bytes2); 4636 auring_take(outbuf, bytes2 / framesize); 4637 } 4638 4639 /* XXX TODO: any counters here? */ 4640 4641 #if defined(AUDIO_DEBUG) 4642 if (audiodebug >= 3) { 4643 struct audio_track_debugbuf m; 4644 audio_track_bufstat(track, &m); 4645 TRACET(0, track, "end%s%s%s%s%s%s", 4646 m.freq, m.chvol, m.chmix, m.codec, m.outbuf, m.usrbuf); 4647 } 4648 #endif 4649 } 4650 4651 /* 4652 * Calcurate blktime [msec] from mixer(.hwbuf.fmt). 4653 * Must be called with sc_lock held. 4654 */ 4655 static u_int 4656 audio_mixer_calc_blktime(struct audio_softc *sc, audio_trackmixer_t *mixer) 4657 { 4658 audio_format2_t *fmt; 4659 u_int blktime; 4660 u_int frames_per_block; 4661 4662 KASSERT(mutex_owned(sc->sc_lock)); 4663 4664 fmt = &mixer->hwbuf.fmt; 4665 blktime = sc->sc_blk_ms; 4666 4667 /* 4668 * If stride is not multiples of 8, special treatment is necessary. 4669 * For now, it is only x68k's vs(4), 4 bit/sample ADPCM. 4670 */ 4671 if (fmt->stride == 4) { 4672 frames_per_block = fmt->sample_rate * blktime / 1000; 4673 if ((frames_per_block & 1) != 0) 4674 blktime *= 2; 4675 } 4676 #ifdef DIAGNOSTIC 4677 else if (fmt->stride % NBBY != 0) { 4678 panic("unsupported HW stride %d", fmt->stride); 4679 } 4680 #endif 4681 4682 return blktime; 4683 } 4684 4685 /* 4686 * Initialize the mixer corresponding to the mode. 4687 * Set AUMODE_PLAY to the 'mode' for playback or AUMODE_RECORD for recording. 4688 * sc->sc_[pr]mixer (corresponding to the 'mode') must be zero-filled. 4689 * This function returns 0 on sucessful. Otherwise returns errno. 4690 * Must be called with sc_lock held. 4691 */ 4692 static int 4693 audio_mixer_init(struct audio_softc *sc, int mode, 4694 const audio_format2_t *hwfmt, const audio_filter_reg_t *reg) 4695 { 4696 char codecbuf[64]; 4697 audio_trackmixer_t *mixer; 4698 void (*softint_handler)(void *); 4699 int len; 4700 int blksize; 4701 int capacity; 4702 size_t bufsize; 4703 int hwblks; 4704 int blkms; 4705 int error; 4706 4707 KASSERT(hwfmt != NULL); 4708 KASSERT(reg != NULL); 4709 KASSERT(mutex_owned(sc->sc_lock)); 4710 4711 error = 0; 4712 if (mode == AUMODE_PLAY) 4713 mixer = sc->sc_pmixer; 4714 else 4715 mixer = sc->sc_rmixer; 4716 4717 mixer->sc = sc; 4718 mixer->mode = mode; 4719 4720 mixer->hwbuf.fmt = *hwfmt; 4721 mixer->volume = 256; 4722 mixer->blktime_d = 1000; 4723 mixer->blktime_n = audio_mixer_calc_blktime(sc, mixer); 4724 sc->sc_blk_ms = mixer->blktime_n; 4725 hwblks = NBLKHW; 4726 4727 mixer->frames_per_block = frame_per_block(mixer, &mixer->hwbuf.fmt); 4728 blksize = frametobyte(&mixer->hwbuf.fmt, mixer->frames_per_block); 4729 if (sc->hw_if->round_blocksize) { 4730 int rounded; 4731 audio_params_t p = format2_to_params(&mixer->hwbuf.fmt); 4732 rounded = sc->hw_if->round_blocksize(sc->hw_hdl, blksize, 4733 mode, &p); 4734 TRACE(2, "round_blocksize %d -> %d", blksize, rounded); 4735 if (rounded != blksize) { 4736 if ((rounded * NBBY) % (mixer->hwbuf.fmt.stride * 4737 mixer->hwbuf.fmt.channels) != 0) { 4738 device_printf(sc->sc_dev, 4739 "blksize not configured %d -> %d\n", 4740 blksize, rounded); 4741 return EINVAL; 4742 } 4743 /* Recalculation */ 4744 blksize = rounded; 4745 mixer->frames_per_block = blksize * NBBY / 4746 (mixer->hwbuf.fmt.stride * 4747 mixer->hwbuf.fmt.channels); 4748 } 4749 } 4750 mixer->blktime_n = mixer->frames_per_block; 4751 mixer->blktime_d = mixer->hwbuf.fmt.sample_rate; 4752 4753 capacity = mixer->frames_per_block * hwblks; 4754 bufsize = frametobyte(&mixer->hwbuf.fmt, capacity); 4755 if (sc->hw_if->round_buffersize) { 4756 size_t rounded; 4757 rounded = sc->hw_if->round_buffersize(sc->hw_hdl, mode, 4758 bufsize); 4759 TRACE(2, "round_buffersize %zd -> %zd", bufsize, rounded); 4760 if (rounded < bufsize) { 4761 /* buffersize needs NBLKHW blocks at least. */ 4762 device_printf(sc->sc_dev, 4763 "buffersize too small: buffersize=%zd blksize=%d\n", 4764 rounded, blksize); 4765 return EINVAL; 4766 } 4767 if (rounded % blksize != 0) { 4768 /* buffersize/blksize constraint mismatch? */ 4769 device_printf(sc->sc_dev, 4770 "buffersize must be multiple of blksize: " 4771 "buffersize=%zu blksize=%d\n", 4772 rounded, blksize); 4773 return EINVAL; 4774 } 4775 if (rounded != bufsize) { 4776 /* Recalcuration */ 4777 bufsize = rounded; 4778 hwblks = bufsize / blksize; 4779 capacity = mixer->frames_per_block * hwblks; 4780 } 4781 } 4782 TRACE(2, "buffersize for %s = %zu", 4783 (mode == AUMODE_PLAY) ? "playback" : "recording", 4784 bufsize); 4785 mixer->hwbuf.capacity = capacity; 4786 4787 /* 4788 * XXX need to release sc_lock for compatibility? 4789 */ 4790 if (sc->hw_if->allocm) { 4791 mixer->hwbuf.mem = sc->hw_if->allocm(sc->hw_hdl, mode, bufsize); 4792 if (mixer->hwbuf.mem == NULL) { 4793 device_printf(sc->sc_dev, "%s: allocm(%zu) failed\n", 4794 __func__, bufsize); 4795 return ENOMEM; 4796 } 4797 } else { 4798 mixer->hwbuf.mem = kern_malloc(bufsize, M_NOWAIT); 4799 if (mixer->hwbuf.mem == NULL) { 4800 device_printf(sc->sc_dev, 4801 "%s: malloc hwbuf(%zu) failed\n", 4802 __func__, bufsize); 4803 return ENOMEM; 4804 } 4805 } 4806 4807 /* From here, audio_mixer_destroy is necessary to exit. */ 4808 if (mode == AUMODE_PLAY) { 4809 cv_init(&mixer->outcv, "audiowr"); 4810 } else { 4811 cv_init(&mixer->outcv, "audiord"); 4812 } 4813 4814 if (mode == AUMODE_PLAY) { 4815 softint_handler = audio_softintr_wr; 4816 } else { 4817 softint_handler = audio_softintr_rd; 4818 } 4819 mixer->sih = softint_establish(SOFTINT_SERIAL | SOFTINT_MPSAFE, 4820 softint_handler, sc); 4821 if (mixer->sih == NULL) { 4822 device_printf(sc->sc_dev, "softint_establish failed\n"); 4823 goto abort; 4824 } 4825 4826 mixer->track_fmt.encoding = AUDIO_ENCODING_SLINEAR_NE; 4827 mixer->track_fmt.precision = AUDIO_INTERNAL_BITS; 4828 mixer->track_fmt.stride = AUDIO_INTERNAL_BITS; 4829 mixer->track_fmt.channels = mixer->hwbuf.fmt.channels; 4830 mixer->track_fmt.sample_rate = mixer->hwbuf.fmt.sample_rate; 4831 4832 if (mixer->hwbuf.fmt.encoding == AUDIO_ENCODING_SLINEAR_OE && 4833 mixer->hwbuf.fmt.precision == AUDIO_INTERNAL_BITS) { 4834 mixer->swap_endian = true; 4835 TRACE(1, "swap_endian"); 4836 } 4837 4838 if (mode == AUMODE_PLAY) { 4839 /* Mixing buffer */ 4840 mixer->mixfmt = mixer->track_fmt; 4841 mixer->mixfmt.precision *= 2; 4842 mixer->mixfmt.stride *= 2; 4843 /* XXX TODO: use some macros? */ 4844 len = mixer->frames_per_block * mixer->mixfmt.channels * 4845 mixer->mixfmt.stride / NBBY; 4846 mixer->mixsample = audio_realloc(mixer->mixsample, len); 4847 if (mixer->mixsample == NULL) { 4848 device_printf(sc->sc_dev, 4849 "%s: malloc mixsample(%d) failed\n", 4850 __func__, len); 4851 error = ENOMEM; 4852 goto abort; 4853 } 4854 } else { 4855 /* No mixing buffer for recording */ 4856 } 4857 4858 if (reg->codec) { 4859 mixer->codec = reg->codec; 4860 mixer->codecarg.context = reg->context; 4861 if (mode == AUMODE_PLAY) { 4862 mixer->codecarg.srcfmt = &mixer->track_fmt; 4863 mixer->codecarg.dstfmt = &mixer->hwbuf.fmt; 4864 } else { 4865 mixer->codecarg.srcfmt = &mixer->hwbuf.fmt; 4866 mixer->codecarg.dstfmt = &mixer->track_fmt; 4867 } 4868 mixer->codecbuf.fmt = mixer->track_fmt; 4869 mixer->codecbuf.capacity = mixer->frames_per_block; 4870 len = auring_bytelen(&mixer->codecbuf); 4871 mixer->codecbuf.mem = audio_realloc(mixer->codecbuf.mem, len); 4872 if (mixer->codecbuf.mem == NULL) { 4873 device_printf(sc->sc_dev, 4874 "%s: malloc codecbuf(%d) failed\n", 4875 __func__, len); 4876 error = ENOMEM; 4877 goto abort; 4878 } 4879 } 4880 4881 /* Succeeded so display it. */ 4882 codecbuf[0] = '\0'; 4883 if (mixer->codec || mixer->swap_endian) { 4884 snprintf(codecbuf, sizeof(codecbuf), " %s %s:%d", 4885 (mode == AUMODE_PLAY) ? "->" : "<-", 4886 audio_encoding_name(mixer->hwbuf.fmt.encoding), 4887 mixer->hwbuf.fmt.precision); 4888 } 4889 blkms = mixer->blktime_n * 1000 / mixer->blktime_d; 4890 aprint_normal_dev(sc->sc_dev, "%s:%d%s %dch %dHz, blk %dms for %s\n", 4891 audio_encoding_name(mixer->track_fmt.encoding), 4892 mixer->track_fmt.precision, 4893 codecbuf, 4894 mixer->track_fmt.channels, 4895 mixer->track_fmt.sample_rate, 4896 blkms, 4897 (mode == AUMODE_PLAY) ? "playback" : "recording"); 4898 4899 return 0; 4900 4901 abort: 4902 audio_mixer_destroy(sc, mixer); 4903 return error; 4904 } 4905 4906 /* 4907 * Releases all resources of 'mixer'. 4908 * Note that it does not release the memory area of 'mixer' itself. 4909 * Must be called with sc_lock held. 4910 */ 4911 static void 4912 audio_mixer_destroy(struct audio_softc *sc, audio_trackmixer_t *mixer) 4913 { 4914 int mode; 4915 4916 KASSERT(mutex_owned(sc->sc_lock)); 4917 4918 mode = mixer->mode; 4919 KASSERT(mode == AUMODE_PLAY || mode == AUMODE_RECORD); 4920 4921 if (mixer->hwbuf.mem != NULL) { 4922 if (sc->hw_if->freem) { 4923 sc->hw_if->freem(sc->hw_hdl, mixer->hwbuf.mem, mode); 4924 } else { 4925 kern_free(mixer->hwbuf.mem); 4926 } 4927 mixer->hwbuf.mem = NULL; 4928 } 4929 4930 audio_free(mixer->codecbuf.mem); 4931 audio_free(mixer->mixsample); 4932 4933 cv_destroy(&mixer->outcv); 4934 4935 if (mixer->sih) { 4936 softint_disestablish(mixer->sih); 4937 mixer->sih = NULL; 4938 } 4939 } 4940 4941 /* 4942 * Starts playback mixer. 4943 * Must be called only if sc_pbusy is false. 4944 * Must be called with sc_lock held. 4945 * Must not be called from the interrupt context. 4946 */ 4947 static void 4948 audio_pmixer_start(struct audio_softc *sc, bool force) 4949 { 4950 audio_trackmixer_t *mixer; 4951 int minimum; 4952 4953 KASSERT(mutex_owned(sc->sc_lock)); 4954 KASSERT(sc->sc_pbusy == false); 4955 4956 mutex_enter(sc->sc_intr_lock); 4957 4958 mixer = sc->sc_pmixer; 4959 TRACE(2, "%smixseq=%d hwseq=%d hwbuf=%d/%d/%d%s", 4960 (audiodebug >= 3) ? "begin " : "", 4961 (int)mixer->mixseq, (int)mixer->hwseq, 4962 mixer->hwbuf.head, mixer->hwbuf.used, mixer->hwbuf.capacity, 4963 force ? " force" : ""); 4964 4965 /* Need two blocks to start normally. */ 4966 minimum = (force) ? 1 : 2; 4967 while (mixer->hwbuf.used < mixer->frames_per_block * minimum) { 4968 audio_pmixer_process(sc); 4969 } 4970 4971 /* Start output */ 4972 audio_pmixer_output(sc); 4973 sc->sc_pbusy = true; 4974 4975 TRACE(3, "end mixseq=%d hwseq=%d hwbuf=%d/%d/%d", 4976 (int)mixer->mixseq, (int)mixer->hwseq, 4977 mixer->hwbuf.head, mixer->hwbuf.used, mixer->hwbuf.capacity); 4978 4979 mutex_exit(sc->sc_intr_lock); 4980 } 4981 4982 /* 4983 * When playing back with MD filter: 4984 * 4985 * track track ... 4986 * v v 4987 * + mix (with aint2_t) 4988 * | master volume (with aint2_t) 4989 * v 4990 * mixsample [::::] wide-int 1 block (ring) buffer 4991 * | 4992 * | convert aint2_t -> aint_t 4993 * v 4994 * codecbuf [....] 1 block (ring) buffer 4995 * | 4996 * | convert to hw format 4997 * v 4998 * hwbuf [............] NBLKHW blocks ring buffer 4999 * 5000 * When playing back without MD filter: 5001 * 5002 * mixsample [::::] wide-int 1 block (ring) buffer 5003 * | 5004 * | convert aint2_t -> aint_t 5005 * | (with byte swap if necessary) 5006 * v 5007 * hwbuf [............] NBLKHW blocks ring buffer 5008 * 5009 * mixsample: slinear_NE, wide internal precision, HW ch, HW freq. 5010 * codecbuf: slinear_NE, internal precision, HW ch, HW freq. 5011 * hwbuf: HW encoding, HW precision, HW ch, HW freq. 5012 */ 5013 5014 /* 5015 * Performs track mixing and converts it to hwbuf. 5016 * Note that this function doesn't transfer hwbuf to hardware. 5017 * Must be called with sc_intr_lock held. 5018 */ 5019 static void 5020 audio_pmixer_process(struct audio_softc *sc) 5021 { 5022 audio_trackmixer_t *mixer; 5023 audio_file_t *f; 5024 int frame_count; 5025 int sample_count; 5026 int mixed; 5027 int i; 5028 aint2_t *m; 5029 aint_t *h; 5030 5031 mixer = sc->sc_pmixer; 5032 5033 frame_count = mixer->frames_per_block; 5034 KASSERT(auring_get_contig_free(&mixer->hwbuf) >= frame_count); 5035 sample_count = frame_count * mixer->mixfmt.channels; 5036 5037 mixer->mixseq++; 5038 5039 /* Mix all tracks */ 5040 mixed = 0; 5041 SLIST_FOREACH(f, &sc->sc_files, entry) { 5042 audio_track_t *track = f->ptrack; 5043 5044 if (track == NULL) 5045 continue; 5046 5047 if (track->is_pause) { 5048 TRACET(4, track, "skip; paused"); 5049 continue; 5050 } 5051 5052 /* Skip if the track is used by process context. */ 5053 if (audio_track_lock_tryenter(track) == false) { 5054 TRACET(4, track, "skip; in use"); 5055 continue; 5056 } 5057 5058 /* Emulate mmap'ped track */ 5059 if (track->mmapped) { 5060 auring_push(&track->usrbuf, track->usrbuf_blksize); 5061 TRACET(4, track, "mmap; usr=%d/%d/C%d", 5062 track->usrbuf.head, 5063 track->usrbuf.used, 5064 track->usrbuf.capacity); 5065 } 5066 5067 if (track->outbuf.used < mixer->frames_per_block && 5068 track->usrbuf.used > 0) { 5069 TRACET(4, track, "process"); 5070 audio_track_play(track); 5071 } 5072 5073 if (track->outbuf.used > 0) { 5074 mixed = audio_pmixer_mix_track(mixer, track, mixed); 5075 } else { 5076 TRACET(4, track, "skip; empty"); 5077 } 5078 5079 audio_track_lock_exit(track); 5080 } 5081 5082 if (mixed == 0) { 5083 /* Silence */ 5084 memset(mixer->mixsample, 0, 5085 frametobyte(&mixer->mixfmt, frame_count)); 5086 } else { 5087 aint2_t ovf_plus; 5088 aint2_t ovf_minus; 5089 int vol; 5090 5091 /* Overflow detection */ 5092 ovf_plus = AINT_T_MAX; 5093 ovf_minus = AINT_T_MIN; 5094 m = mixer->mixsample; 5095 for (i = 0; i < sample_count; i++) { 5096 aint2_t val; 5097 5098 val = *m++; 5099 if (val > ovf_plus) 5100 ovf_plus = val; 5101 else if (val < ovf_minus) 5102 ovf_minus = val; 5103 } 5104 5105 /* Master Volume Auto Adjust */ 5106 vol = mixer->volume; 5107 if (ovf_plus > (aint2_t)AINT_T_MAX 5108 || ovf_minus < (aint2_t)AINT_T_MIN) { 5109 aint2_t ovf; 5110 int vol2; 5111 5112 /* XXX TODO: Check AINT2_T_MIN ? */ 5113 ovf = ovf_plus; 5114 if (ovf < -ovf_minus) 5115 ovf = -ovf_minus; 5116 5117 /* Turn down the volume if overflow occured. */ 5118 vol2 = (int)((aint2_t)AINT_T_MAX * 256 / ovf); 5119 if (vol2 < vol) 5120 vol = vol2; 5121 5122 if (vol < mixer->volume) { 5123 /* Turn down gradually to 128. */ 5124 if (mixer->volume > 128) { 5125 mixer->volume = 5126 (mixer->volume * 95) / 100; 5127 device_printf(sc->sc_dev, 5128 "auto volume adjust: volume %d\n", 5129 mixer->volume); 5130 } 5131 } 5132 } 5133 5134 /* Apply Master Volume. */ 5135 if (vol != 256) { 5136 m = mixer->mixsample; 5137 for (i = 0; i < sample_count; i++) { 5138 #if defined(AUDIO_USE_C_IMPLEMENTATION_DEFINED_BEHAVIOR) && defined(__GNUC__) 5139 *m = *m * vol >> 8; 5140 #else 5141 *m = *m * vol / 256; 5142 #endif 5143 m++; 5144 } 5145 } 5146 } 5147 5148 /* 5149 * The rest is the hardware part. 5150 */ 5151 5152 if (mixer->codec) { 5153 h = auring_tailptr_aint(&mixer->codecbuf); 5154 } else { 5155 h = auring_tailptr_aint(&mixer->hwbuf); 5156 } 5157 5158 m = mixer->mixsample; 5159 if (mixer->swap_endian) { 5160 for (i = 0; i < sample_count; i++) { 5161 *h++ = bswap16(*m++); 5162 } 5163 } else { 5164 for (i = 0; i < sample_count; i++) { 5165 *h++ = *m++; 5166 } 5167 } 5168 5169 /* Hardware driver's codec */ 5170 if (mixer->codec) { 5171 auring_push(&mixer->codecbuf, frame_count); 5172 mixer->codecarg.src = auring_headptr(&mixer->codecbuf); 5173 mixer->codecarg.dst = auring_tailptr(&mixer->hwbuf); 5174 mixer->codecarg.count = frame_count; 5175 mixer->codec(&mixer->codecarg); 5176 auring_take(&mixer->codecbuf, mixer->codecarg.count); 5177 } 5178 5179 auring_push(&mixer->hwbuf, frame_count); 5180 5181 TRACE(4, "done mixseq=%d hwbuf=%d/%d/%d%s", 5182 (int)mixer->mixseq, 5183 mixer->hwbuf.head, mixer->hwbuf.used, mixer->hwbuf.capacity, 5184 (mixed == 0) ? " silent" : ""); 5185 } 5186 5187 /* 5188 * Mix one track. 5189 * 'mixed' specifies the number of tracks mixed so far. 5190 * It returns the number of tracks mixed. In other words, it returns 5191 * mixed + 1 if this track is mixed. 5192 */ 5193 static int 5194 audio_pmixer_mix_track(audio_trackmixer_t *mixer, audio_track_t *track, 5195 int mixed) 5196 { 5197 int count; 5198 int sample_count; 5199 int remain; 5200 int i; 5201 const aint_t *s; 5202 aint2_t *d; 5203 5204 /* XXX TODO: Is this necessary for now? */ 5205 if (mixer->mixseq < track->seq) 5206 return mixed; 5207 5208 count = auring_get_contig_used(&track->outbuf); 5209 count = uimin(count, mixer->frames_per_block); 5210 5211 s = auring_headptr_aint(&track->outbuf); 5212 d = mixer->mixsample; 5213 5214 /* 5215 * Apply track volume with double-sized integer and perform 5216 * additive synthesis. 5217 * 5218 * XXX If you limit the track volume to 1.0 or less (<= 256), 5219 * it would be better to do this in the track conversion stage 5220 * rather than here. However, if you accept the volume to 5221 * be greater than 1.0 (> 256), it's better to do it here. 5222 * Because the operation here is done by double-sized integer. 5223 */ 5224 sample_count = count * mixer->mixfmt.channels; 5225 if (mixed == 0) { 5226 /* If this is the first track, assignment can be used. */ 5227 #if defined(AUDIO_SUPPORT_TRACK_VOLUME) 5228 if (track->volume != 256) { 5229 for (i = 0; i < sample_count; i++) { 5230 #if defined(AUDIO_USE_C_IMPLEMENTATION_DEFINED_BEHAVIOR) && defined(__GNUC__) 5231 *d++ = ((aint2_t)*s++) * track->volume >> 8; 5232 #else 5233 *d++ = ((aint2_t)*s++) * track->volume / 256; 5234 #endif 5235 } 5236 } else 5237 #endif 5238 { 5239 for (i = 0; i < sample_count; i++) { 5240 *d++ = ((aint2_t)*s++); 5241 } 5242 } 5243 } else { 5244 /* If this is the second or later, add it. */ 5245 #if defined(AUDIO_SUPPORT_TRACK_VOLUME) 5246 if (track->volume != 256) { 5247 for (i = 0; i < sample_count; i++) { 5248 #if defined(AUDIO_USE_C_IMPLEMENTATION_DEFINED_BEHAVIOR) && defined(__GNUC__) 5249 *d++ += ((aint2_t)*s++) * track->volume >> 8; 5250 #else 5251 *d++ += ((aint2_t)*s++) * track->volume / 256; 5252 #endif 5253 } 5254 } else 5255 #endif 5256 { 5257 for (i = 0; i < sample_count; i++) { 5258 *d++ += ((aint2_t)*s++); 5259 } 5260 } 5261 } 5262 5263 auring_take(&track->outbuf, count); 5264 /* 5265 * The counters have to align block even if outbuf is less than 5266 * one block. XXX Is this still necessary? 5267 */ 5268 remain = mixer->frames_per_block - count; 5269 if (__predict_false(remain != 0)) { 5270 auring_push(&track->outbuf, remain); 5271 auring_take(&track->outbuf, remain); 5272 } 5273 5274 /* 5275 * Update track sequence. 5276 * mixseq has previous value yet at this point. 5277 */ 5278 track->seq = mixer->mixseq + 1; 5279 5280 return mixed + 1; 5281 } 5282 5283 /* 5284 * Output one block from hwbuf to HW. 5285 * Must be called with sc_intr_lock held. 5286 */ 5287 static void 5288 audio_pmixer_output(struct audio_softc *sc) 5289 { 5290 audio_trackmixer_t *mixer; 5291 audio_params_t params; 5292 void *start; 5293 void *end; 5294 int blksize; 5295 int error; 5296 5297 mixer = sc->sc_pmixer; 5298 TRACE(4, "pbusy=%d hwbuf=%d/%d/%d", 5299 sc->sc_pbusy, 5300 mixer->hwbuf.head, mixer->hwbuf.used, mixer->hwbuf.capacity); 5301 KASSERT(mixer->hwbuf.used >= mixer->frames_per_block); 5302 5303 blksize = frametobyte(&mixer->hwbuf.fmt, mixer->frames_per_block); 5304 5305 if (sc->hw_if->trigger_output) { 5306 /* trigger (at once) */ 5307 if (!sc->sc_pbusy) { 5308 start = mixer->hwbuf.mem; 5309 end = (uint8_t *)start + auring_bytelen(&mixer->hwbuf); 5310 params = format2_to_params(&mixer->hwbuf.fmt); 5311 5312 error = sc->hw_if->trigger_output(sc->hw_hdl, 5313 start, end, blksize, audio_pintr, sc, ¶ms); 5314 if (error) { 5315 device_printf(sc->sc_dev, 5316 "trigger_output failed with %d", error); 5317 return; 5318 } 5319 } 5320 } else { 5321 /* start (everytime) */ 5322 start = auring_headptr(&mixer->hwbuf); 5323 5324 error = sc->hw_if->start_output(sc->hw_hdl, 5325 start, blksize, audio_pintr, sc); 5326 if (error) { 5327 device_printf(sc->sc_dev, 5328 "start_output failed with %d", error); 5329 return; 5330 } 5331 } 5332 } 5333 5334 /* 5335 * This is an interrupt handler for playback. 5336 * It is called with sc_intr_lock held. 5337 * 5338 * It is usually called from hardware interrupt. However, note that 5339 * for some drivers (e.g. uaudio) it is called from software interrupt. 5340 */ 5341 static void 5342 audio_pintr(void *arg) 5343 { 5344 struct audio_softc *sc; 5345 audio_trackmixer_t *mixer; 5346 5347 sc = arg; 5348 KASSERT(mutex_owned(sc->sc_intr_lock)); 5349 5350 if (sc->sc_dying) 5351 return; 5352 #if defined(DIAGNOSTIC) 5353 if (sc->sc_pbusy == false) { 5354 device_printf(sc->sc_dev, "stray interrupt\n"); 5355 return; 5356 } 5357 #endif 5358 5359 mixer = sc->sc_pmixer; 5360 mixer->hw_complete_counter += mixer->frames_per_block; 5361 mixer->hwseq++; 5362 5363 auring_take(&mixer->hwbuf, mixer->frames_per_block); 5364 5365 TRACE(4, 5366 "HW_INT ++hwseq=%" PRIu64 " cmplcnt=%" PRIu64 " hwbuf=%d/%d/%d", 5367 mixer->hwseq, mixer->hw_complete_counter, 5368 mixer->hwbuf.head, mixer->hwbuf.used, mixer->hwbuf.capacity); 5369 5370 #if !defined(_KERNEL) 5371 /* This is a debug code for userland test. */ 5372 return; 5373 #endif 5374 5375 #if defined(AUDIO_HW_SINGLE_BUFFER) 5376 /* 5377 * Create a new block here and output it immediately. 5378 * It makes a latency lower but needs machine power. 5379 */ 5380 audio_pmixer_process(sc); 5381 audio_pmixer_output(sc); 5382 #else 5383 /* 5384 * It is called when block N output is done. 5385 * Output immediately block N+1 created by the last interrupt. 5386 * And then create block N+2 for the next interrupt. 5387 * This method makes playback robust even on slower machines. 5388 * Instead the latency is increased by one block. 5389 */ 5390 5391 /* At first, output ready block. */ 5392 if (mixer->hwbuf.used >= mixer->frames_per_block) { 5393 audio_pmixer_output(sc); 5394 } 5395 5396 bool later = false; 5397 5398 if (mixer->hwbuf.used < mixer->frames_per_block) { 5399 later = true; 5400 } 5401 5402 /* Then, process next block. */ 5403 audio_pmixer_process(sc); 5404 5405 if (later) { 5406 audio_pmixer_output(sc); 5407 } 5408 #endif 5409 5410 /* 5411 * When this interrupt is the real hardware interrupt, disabling 5412 * preemption here is not necessary. But some drivers (e.g. uaudio) 5413 * emulate it by software interrupt, so kpreempt_disable is necessary. 5414 */ 5415 kpreempt_disable(); 5416 softint_schedule(mixer->sih); 5417 kpreempt_enable(); 5418 } 5419 5420 /* 5421 * Starts record mixer. 5422 * Must be called only if sc_rbusy is false. 5423 * Must be called with sc_lock held. 5424 * Must not be called from the interrupt context. 5425 */ 5426 static void 5427 audio_rmixer_start(struct audio_softc *sc) 5428 { 5429 5430 KASSERT(mutex_owned(sc->sc_lock)); 5431 KASSERT(sc->sc_rbusy == false); 5432 5433 mutex_enter(sc->sc_intr_lock); 5434 5435 TRACE(2, "%s", (audiodebug >= 3) ? "begin" : ""); 5436 audio_rmixer_input(sc); 5437 sc->sc_rbusy = true; 5438 TRACE(3, "end"); 5439 5440 mutex_exit(sc->sc_intr_lock); 5441 } 5442 5443 /* 5444 * When recording with MD filter: 5445 * 5446 * hwbuf [............] NBLKHW blocks ring buffer 5447 * | 5448 * | convert from hw format 5449 * v 5450 * codecbuf [....] 1 block (ring) buffer 5451 * | | 5452 * v v 5453 * track track ... 5454 * 5455 * When recording without MD filter: 5456 * 5457 * hwbuf [............] NBLKHW blocks ring buffer 5458 * | | 5459 * v v 5460 * track track ... 5461 * 5462 * hwbuf: HW encoding, HW precision, HW ch, HW freq. 5463 * codecbuf: slinear_NE, internal precision, HW ch, HW freq. 5464 */ 5465 5466 /* 5467 * Distribute a recorded block to all recording tracks. 5468 */ 5469 static void 5470 audio_rmixer_process(struct audio_softc *sc) 5471 { 5472 audio_trackmixer_t *mixer; 5473 audio_ring_t *mixersrc; 5474 audio_file_t *f; 5475 aint_t *p; 5476 int count; 5477 int bytes; 5478 int i; 5479 5480 mixer = sc->sc_rmixer; 5481 5482 /* 5483 * count is the number of frames to be retrieved this time. 5484 * count should be one block. 5485 */ 5486 count = auring_get_contig_used(&mixer->hwbuf); 5487 count = uimin(count, mixer->frames_per_block); 5488 if (count <= 0) { 5489 TRACE(4, "count %d: too short", count); 5490 return; 5491 } 5492 bytes = frametobyte(&mixer->track_fmt, count); 5493 5494 /* Hardware driver's codec */ 5495 if (mixer->codec) { 5496 mixer->codecarg.src = auring_headptr(&mixer->hwbuf); 5497 mixer->codecarg.dst = auring_tailptr(&mixer->codecbuf); 5498 mixer->codecarg.count = count; 5499 mixer->codec(&mixer->codecarg); 5500 auring_take(&mixer->hwbuf, mixer->codecarg.count); 5501 auring_push(&mixer->codecbuf, mixer->codecarg.count); 5502 mixersrc = &mixer->codecbuf; 5503 } else { 5504 mixersrc = &mixer->hwbuf; 5505 } 5506 5507 if (mixer->swap_endian) { 5508 /* inplace conversion */ 5509 p = auring_headptr_aint(mixersrc); 5510 for (i = 0; i < count * mixer->track_fmt.channels; i++, p++) { 5511 *p = bswap16(*p); 5512 } 5513 } 5514 5515 /* Distribute to all tracks. */ 5516 SLIST_FOREACH(f, &sc->sc_files, entry) { 5517 audio_track_t *track = f->rtrack; 5518 audio_ring_t *input; 5519 5520 if (track == NULL) 5521 continue; 5522 5523 if (track->is_pause) { 5524 TRACET(4, track, "skip; paused"); 5525 continue; 5526 } 5527 5528 if (audio_track_lock_tryenter(track) == false) { 5529 TRACET(4, track, "skip; in use"); 5530 continue; 5531 } 5532 5533 /* If the track buffer is full, discard the oldest one? */ 5534 input = track->input; 5535 if (input->capacity - input->used < mixer->frames_per_block) { 5536 int drops = mixer->frames_per_block - 5537 (input->capacity - input->used); 5538 track->dropframes += drops; 5539 TRACET(4, track, "drop %d frames: inp=%d/%d/%d", 5540 drops, 5541 input->head, input->used, input->capacity); 5542 auring_take(input, drops); 5543 } 5544 KASSERT(input->used % mixer->frames_per_block == 0); 5545 5546 memcpy(auring_tailptr_aint(input), 5547 auring_headptr_aint(mixersrc), 5548 bytes); 5549 auring_push(input, count); 5550 5551 /* XXX sequence counter? */ 5552 5553 audio_track_lock_exit(track); 5554 } 5555 5556 auring_take(mixersrc, count); 5557 } 5558 5559 /* 5560 * Input one block from HW to hwbuf. 5561 * Must be called with sc_intr_lock held. 5562 */ 5563 static void 5564 audio_rmixer_input(struct audio_softc *sc) 5565 { 5566 audio_trackmixer_t *mixer; 5567 audio_params_t params; 5568 void *start; 5569 void *end; 5570 int blksize; 5571 int error; 5572 5573 mixer = sc->sc_rmixer; 5574 blksize = frametobyte(&mixer->hwbuf.fmt, mixer->frames_per_block); 5575 5576 if (sc->hw_if->trigger_input) { 5577 /* trigger (at once) */ 5578 if (!sc->sc_rbusy) { 5579 start = mixer->hwbuf.mem; 5580 end = (uint8_t *)start + auring_bytelen(&mixer->hwbuf); 5581 params = format2_to_params(&mixer->hwbuf.fmt); 5582 5583 error = sc->hw_if->trigger_input(sc->hw_hdl, 5584 start, end, blksize, audio_rintr, sc, ¶ms); 5585 if (error) { 5586 device_printf(sc->sc_dev, 5587 "trigger_input failed with %d", error); 5588 return; 5589 } 5590 } 5591 } else { 5592 /* start (everytime) */ 5593 start = auring_tailptr(&mixer->hwbuf); 5594 5595 error = sc->hw_if->start_input(sc->hw_hdl, 5596 start, blksize, audio_rintr, sc); 5597 if (error) { 5598 device_printf(sc->sc_dev, 5599 "start_input failed with %d", error); 5600 return; 5601 } 5602 } 5603 } 5604 5605 /* 5606 * This is an interrupt handler for recording. 5607 * It is called with sc_intr_lock. 5608 * 5609 * It is usually called from hardware interrupt. However, note that 5610 * for some drivers (e.g. uaudio) it is called from software interrupt. 5611 */ 5612 static void 5613 audio_rintr(void *arg) 5614 { 5615 struct audio_softc *sc; 5616 audio_trackmixer_t *mixer; 5617 5618 sc = arg; 5619 KASSERT(mutex_owned(sc->sc_intr_lock)); 5620 5621 if (sc->sc_dying) 5622 return; 5623 #if defined(DIAGNOSTIC) 5624 if (sc->sc_rbusy == false) { 5625 device_printf(sc->sc_dev, "stray interrupt\n"); 5626 return; 5627 } 5628 #endif 5629 5630 mixer = sc->sc_rmixer; 5631 mixer->hw_complete_counter += mixer->frames_per_block; 5632 mixer->hwseq++; 5633 5634 auring_push(&mixer->hwbuf, mixer->frames_per_block); 5635 5636 TRACE(4, 5637 "HW_INT ++hwseq=%" PRIu64 " cmplcnt=%" PRIu64 " hwbuf=%d/%d/%d", 5638 mixer->hwseq, mixer->hw_complete_counter, 5639 mixer->hwbuf.head, mixer->hwbuf.used, mixer->hwbuf.capacity); 5640 5641 /* Distrubute recorded block */ 5642 audio_rmixer_process(sc); 5643 5644 /* Request next block */ 5645 audio_rmixer_input(sc); 5646 5647 /* 5648 * When this interrupt is the real hardware interrupt, disabling 5649 * preemption here is not necessary. But some drivers (e.g. uaudio) 5650 * emulate it by software interrupt, so kpreempt_disable is necessary. 5651 */ 5652 kpreempt_disable(); 5653 softint_schedule(mixer->sih); 5654 kpreempt_enable(); 5655 } 5656 5657 /* 5658 * Halts playback mixer. 5659 * This function also clears related parameters, so call this function 5660 * instead of calling halt_output directly. 5661 * Must be called only if sc_pbusy is true. 5662 * Must be called with sc_lock && sc_exlock held. 5663 */ 5664 static int 5665 audio_pmixer_halt(struct audio_softc *sc) 5666 { 5667 int error; 5668 5669 TRACE(2, ""); 5670 KASSERT(mutex_owned(sc->sc_lock)); 5671 KASSERT(sc->sc_exlock); 5672 5673 mutex_enter(sc->sc_intr_lock); 5674 error = sc->hw_if->halt_output(sc->hw_hdl); 5675 mutex_exit(sc->sc_intr_lock); 5676 5677 /* Halts anyway even if some error has occurred. */ 5678 sc->sc_pbusy = false; 5679 sc->sc_pmixer->hwbuf.head = 0; 5680 sc->sc_pmixer->hwbuf.used = 0; 5681 sc->sc_pmixer->mixseq = 0; 5682 sc->sc_pmixer->hwseq = 0; 5683 5684 return error; 5685 } 5686 5687 /* 5688 * Halts recording mixer. 5689 * This function also clears related parameters, so call this function 5690 * instead of calling halt_input directly. 5691 * Must be called only if sc_rbusy is true. 5692 * Must be called with sc_lock && sc_exlock held. 5693 */ 5694 static int 5695 audio_rmixer_halt(struct audio_softc *sc) 5696 { 5697 int error; 5698 5699 TRACE(2, ""); 5700 KASSERT(mutex_owned(sc->sc_lock)); 5701 KASSERT(sc->sc_exlock); 5702 5703 mutex_enter(sc->sc_intr_lock); 5704 error = sc->hw_if->halt_input(sc->hw_hdl); 5705 mutex_exit(sc->sc_intr_lock); 5706 5707 /* Halts anyway even if some error has occurred. */ 5708 sc->sc_rbusy = false; 5709 sc->sc_rmixer->hwbuf.head = 0; 5710 sc->sc_rmixer->hwbuf.used = 0; 5711 sc->sc_rmixer->mixseq = 0; 5712 sc->sc_rmixer->hwseq = 0; 5713 5714 return error; 5715 } 5716 5717 /* 5718 * Flush this track. 5719 * Halts all operations, clears all buffers, reset error counters. 5720 * XXX I'm not sure... 5721 */ 5722 static void 5723 audio_track_clear(struct audio_softc *sc, audio_track_t *track) 5724 { 5725 5726 KASSERT(track); 5727 TRACET(3, track, "clear"); 5728 5729 audio_track_lock_enter(track); 5730 5731 track->usrbuf.used = 0; 5732 /* Clear all internal parameters. */ 5733 if (track->codec.filter) { 5734 track->codec.srcbuf.used = 0; 5735 track->codec.srcbuf.head = 0; 5736 } 5737 if (track->chvol.filter) { 5738 track->chvol.srcbuf.used = 0; 5739 track->chvol.srcbuf.head = 0; 5740 } 5741 if (track->chmix.filter) { 5742 track->chmix.srcbuf.used = 0; 5743 track->chmix.srcbuf.head = 0; 5744 } 5745 if (track->freq.filter) { 5746 track->freq.srcbuf.used = 0; 5747 track->freq.srcbuf.head = 0; 5748 if (track->freq_step < 65536) 5749 track->freq_current = 65536; 5750 else 5751 track->freq_current = 0; 5752 memset(track->freq_prev, 0, sizeof(track->freq_prev)); 5753 memset(track->freq_curr, 0, sizeof(track->freq_curr)); 5754 } 5755 /* Clear buffer, then operation halts naturally. */ 5756 track->outbuf.used = 0; 5757 5758 /* Clear counters. */ 5759 track->dropframes = 0; 5760 5761 audio_track_lock_exit(track); 5762 } 5763 5764 /* 5765 * Drain the track. 5766 * track must be present and for playback. 5767 * If successful, it returns 0. Otherwise returns errno. 5768 * Must be called with sc_lock held. 5769 */ 5770 static int 5771 audio_track_drain(struct audio_softc *sc, audio_track_t *track) 5772 { 5773 audio_trackmixer_t *mixer; 5774 int done; 5775 int error; 5776 5777 KASSERT(track); 5778 TRACET(3, track, "start"); 5779 mixer = track->mixer; 5780 KASSERT(mutex_owned(sc->sc_lock)); 5781 5782 /* Ignore them if pause. */ 5783 if (track->is_pause) { 5784 TRACET(3, track, "pause -> clear"); 5785 track->pstate = AUDIO_STATE_CLEAR; 5786 } 5787 /* Terminate early here if there is no data in the track. */ 5788 if (track->pstate == AUDIO_STATE_CLEAR) { 5789 TRACET(3, track, "no need to drain"); 5790 return 0; 5791 } 5792 track->pstate = AUDIO_STATE_DRAINING; 5793 5794 for (;;) { 5795 /* I want to display it bofore condition evaluation. */ 5796 TRACET(3, track, "pid=%d.%d trkseq=%d hwseq=%d out=%d/%d/%d", 5797 (int)curproc->p_pid, (int)curlwp->l_lid, 5798 (int)track->seq, (int)mixer->hwseq, 5799 track->outbuf.head, track->outbuf.used, 5800 track->outbuf.capacity); 5801 5802 /* Condition to terminate */ 5803 audio_track_lock_enter(track); 5804 done = (track->usrbuf.used < frametobyte(&track->inputfmt, 1) && 5805 track->outbuf.used == 0 && 5806 track->seq <= mixer->hwseq); 5807 audio_track_lock_exit(track); 5808 if (done) 5809 break; 5810 5811 TRACET(3, track, "sleep"); 5812 error = audio_track_waitio(sc, track); 5813 if (error) 5814 return error; 5815 5816 /* XXX call audio_track_play here ? */ 5817 } 5818 5819 track->pstate = AUDIO_STATE_CLEAR; 5820 TRACET(3, track, "done trk_inp=%d trk_out=%d", 5821 (int)track->inputcounter, (int)track->outputcounter); 5822 return 0; 5823 } 5824 5825 /* 5826 * This is software interrupt handler for record. 5827 * It is called from recording hardware interrupt everytime. 5828 * It does: 5829 * - Deliver SIGIO for all async processes. 5830 * - Notify to audio_read() that data has arrived. 5831 * - selnotify() for select/poll-ing processes. 5832 */ 5833 /* 5834 * XXX If a process issues FIOASYNC between hardware interrupt and 5835 * software interrupt, (stray) SIGIO will be sent to the process 5836 * despite the fact that it has not receive recorded data yet. 5837 */ 5838 static void 5839 audio_softintr_rd(void *cookie) 5840 { 5841 struct audio_softc *sc = cookie; 5842 audio_file_t *f; 5843 proc_t *p; 5844 pid_t pid; 5845 5846 mutex_enter(sc->sc_lock); 5847 mutex_enter(sc->sc_intr_lock); 5848 5849 SLIST_FOREACH(f, &sc->sc_files, entry) { 5850 audio_track_t *track = f->rtrack; 5851 5852 if (track == NULL) 5853 continue; 5854 5855 TRACET(4, track, "broadcast; inp=%d/%d/%d", 5856 track->input->head, 5857 track->input->used, 5858 track->input->capacity); 5859 5860 pid = f->async_audio; 5861 if (pid != 0) { 5862 TRACEF(4, f, "sending SIGIO %d", pid); 5863 mutex_enter(proc_lock); 5864 if ((p = proc_find(pid)) != NULL) 5865 psignal(p, SIGIO); 5866 mutex_exit(proc_lock); 5867 } 5868 } 5869 mutex_exit(sc->sc_intr_lock); 5870 5871 /* Notify that data has arrived. */ 5872 selnotify(&sc->sc_rsel, 0, NOTE_SUBMIT); 5873 KNOTE(&sc->sc_rsel.sel_klist, 0); 5874 cv_broadcast(&sc->sc_rmixer->outcv); 5875 5876 mutex_exit(sc->sc_lock); 5877 } 5878 5879 /* 5880 * This is software interrupt handler for playback. 5881 * It is called from playback hardware interrupt everytime. 5882 * It does: 5883 * - Deliver SIGIO for all async and writable (used < lowat) processes. 5884 * - Notify to audio_write() that outbuf block available. 5885 * - selnotify() for select/poll-ing processes if there are any writable 5886 * (used < lowat) processes. Checking each descriptor will be done by 5887 * filt_audiowrite_event(). 5888 */ 5889 static void 5890 audio_softintr_wr(void *cookie) 5891 { 5892 struct audio_softc *sc = cookie; 5893 audio_file_t *f; 5894 bool found; 5895 proc_t *p; 5896 pid_t pid; 5897 5898 TRACE(4, "called"); 5899 found = false; 5900 5901 mutex_enter(sc->sc_lock); 5902 mutex_enter(sc->sc_intr_lock); 5903 5904 SLIST_FOREACH(f, &sc->sc_files, entry) { 5905 audio_track_t *track = f->ptrack; 5906 5907 if (track == NULL) 5908 continue; 5909 5910 TRACET(4, track, "broadcast; trseq=%d out=%d/%d/%d", 5911 (int)track->seq, 5912 track->outbuf.head, 5913 track->outbuf.used, 5914 track->outbuf.capacity); 5915 5916 /* 5917 * Send a signal if the process is async mode and 5918 * used is lower than lowat. 5919 */ 5920 if (track->usrbuf.used <= track->usrbuf_usedlow && 5921 !track->is_pause) { 5922 found = true; 5923 pid = f->async_audio; 5924 if (pid != 0) { 5925 TRACEF(4, f, "sending SIGIO %d", pid); 5926 mutex_enter(proc_lock); 5927 if ((p = proc_find(pid)) != NULL) 5928 psignal(p, SIGIO); 5929 mutex_exit(proc_lock); 5930 } 5931 } 5932 } 5933 mutex_exit(sc->sc_intr_lock); 5934 5935 /* 5936 * Notify for select/poll when someone become writable. 5937 * It needs sc_lock (and not sc_intr_lock). 5938 */ 5939 if (found) { 5940 TRACE(4, "selnotify"); 5941 selnotify(&sc->sc_wsel, 0, NOTE_SUBMIT); 5942 KNOTE(&sc->sc_wsel.sel_klist, 0); 5943 } 5944 5945 /* Notify to audio_write() that outbuf available. */ 5946 cv_broadcast(&sc->sc_pmixer->outcv); 5947 5948 mutex_exit(sc->sc_lock); 5949 } 5950 5951 /* 5952 * Check (and convert) the format *p came from userland. 5953 * If successful, it writes back the converted format to *p if necessary 5954 * and returns 0. Otherwise returns errno (*p may change even this case). 5955 */ 5956 static int 5957 audio_check_params(audio_format2_t *p) 5958 { 5959 5960 /* Convert obsoleted AUDIO_ENCODING_PCM* */ 5961 /* XXX Is this conversion right? */ 5962 if (p->encoding == AUDIO_ENCODING_PCM16) { 5963 if (p->precision == 8) 5964 p->encoding = AUDIO_ENCODING_ULINEAR; 5965 else 5966 p->encoding = AUDIO_ENCODING_SLINEAR; 5967 } else if (p->encoding == AUDIO_ENCODING_PCM8) { 5968 if (p->precision == 8) 5969 p->encoding = AUDIO_ENCODING_ULINEAR; 5970 else 5971 return EINVAL; 5972 } 5973 5974 /* 5975 * Convert obsoleted AUDIO_ENCODING_[SU]LINEAR without endianness 5976 * suffix. 5977 */ 5978 if (p->encoding == AUDIO_ENCODING_SLINEAR) 5979 p->encoding = AUDIO_ENCODING_SLINEAR_NE; 5980 if (p->encoding == AUDIO_ENCODING_ULINEAR) 5981 p->encoding = AUDIO_ENCODING_ULINEAR_NE; 5982 5983 switch (p->encoding) { 5984 case AUDIO_ENCODING_ULAW: 5985 case AUDIO_ENCODING_ALAW: 5986 if (p->precision != 8) 5987 return EINVAL; 5988 break; 5989 case AUDIO_ENCODING_ADPCM: 5990 if (p->precision != 4 && p->precision != 8) 5991 return EINVAL; 5992 break; 5993 case AUDIO_ENCODING_SLINEAR_LE: 5994 case AUDIO_ENCODING_SLINEAR_BE: 5995 case AUDIO_ENCODING_ULINEAR_LE: 5996 case AUDIO_ENCODING_ULINEAR_BE: 5997 if (p->precision != 8 && p->precision != 16 && 5998 p->precision != 24 && p->precision != 32) 5999 return EINVAL; 6000 6001 /* 8bit format does not have endianness. */ 6002 if (p->precision == 8) { 6003 if (p->encoding == AUDIO_ENCODING_SLINEAR_OE) 6004 p->encoding = AUDIO_ENCODING_SLINEAR_NE; 6005 if (p->encoding == AUDIO_ENCODING_ULINEAR_OE) 6006 p->encoding = AUDIO_ENCODING_ULINEAR_NE; 6007 } 6008 6009 if (p->precision > p->stride) 6010 return EINVAL; 6011 break; 6012 case AUDIO_ENCODING_MPEG_L1_STREAM: 6013 case AUDIO_ENCODING_MPEG_L1_PACKETS: 6014 case AUDIO_ENCODING_MPEG_L1_SYSTEM: 6015 case AUDIO_ENCODING_MPEG_L2_STREAM: 6016 case AUDIO_ENCODING_MPEG_L2_PACKETS: 6017 case AUDIO_ENCODING_MPEG_L2_SYSTEM: 6018 case AUDIO_ENCODING_AC3: 6019 break; 6020 default: 6021 return EINVAL; 6022 } 6023 6024 /* sanity check # of channels*/ 6025 if (p->channels < 1 || p->channels > AUDIO_MAX_CHANNELS) 6026 return EINVAL; 6027 6028 return 0; 6029 } 6030 6031 /* 6032 * Initialize playback and record mixers. 6033 * mode (AUMODE_{PLAY,RECORD}) indicates the mixer to be initalized. 6034 * phwfmt and rhwfmt indicate the hardware format. pfil and rfil indicate 6035 * the filter registration information. These four must not be NULL. 6036 * If successful returns 0. Otherwise returns errno. 6037 * Must be called with sc_lock held. 6038 * Must not be called if there are any tracks. 6039 * Caller should check that the initialization succeed by whether 6040 * sc_[pr]mixer is not NULL. 6041 */ 6042 static int 6043 audio_mixers_init(struct audio_softc *sc, int mode, 6044 const audio_format2_t *phwfmt, const audio_format2_t *rhwfmt, 6045 const audio_filter_reg_t *pfil, const audio_filter_reg_t *rfil) 6046 { 6047 int error; 6048 6049 KASSERT(phwfmt != NULL); 6050 KASSERT(rhwfmt != NULL); 6051 KASSERT(pfil != NULL); 6052 KASSERT(rfil != NULL); 6053 KASSERT(mutex_owned(sc->sc_lock)); 6054 6055 if ((mode & AUMODE_PLAY)) { 6056 if (sc->sc_pmixer) { 6057 audio_mixer_destroy(sc, sc->sc_pmixer); 6058 kmem_free(sc->sc_pmixer, sizeof(*sc->sc_pmixer)); 6059 } 6060 sc->sc_pmixer = kmem_zalloc(sizeof(*sc->sc_pmixer), KM_SLEEP); 6061 error = audio_mixer_init(sc, AUMODE_PLAY, phwfmt, pfil); 6062 if (error) { 6063 aprint_error_dev(sc->sc_dev, 6064 "configuring playback mode failed\n"); 6065 kmem_free(sc->sc_pmixer, sizeof(*sc->sc_pmixer)); 6066 sc->sc_pmixer = NULL; 6067 return error; 6068 } 6069 } 6070 if ((mode & AUMODE_RECORD)) { 6071 if (sc->sc_rmixer) { 6072 audio_mixer_destroy(sc, sc->sc_rmixer); 6073 kmem_free(sc->sc_rmixer, sizeof(*sc->sc_rmixer)); 6074 } 6075 sc->sc_rmixer = kmem_zalloc(sizeof(*sc->sc_rmixer), KM_SLEEP); 6076 error = audio_mixer_init(sc, AUMODE_RECORD, rhwfmt, rfil); 6077 if (error) { 6078 aprint_error_dev(sc->sc_dev, 6079 "configuring record mode failed\n"); 6080 kmem_free(sc->sc_rmixer, sizeof(*sc->sc_rmixer)); 6081 sc->sc_rmixer = NULL; 6082 return error; 6083 } 6084 } 6085 6086 return 0; 6087 } 6088 6089 /* 6090 * Select a frequency. 6091 * Prioritize 48kHz and 44.1kHz. Otherwise choose the highest one. 6092 * XXX Better algorithm? 6093 */ 6094 static int 6095 audio_select_freq(const struct audio_format *fmt) 6096 { 6097 int freq; 6098 int high; 6099 int low; 6100 int j; 6101 6102 if (fmt->frequency_type == 0) { 6103 low = fmt->frequency[0]; 6104 high = fmt->frequency[1]; 6105 freq = 48000; 6106 if (low <= freq && freq <= high) { 6107 return freq; 6108 } 6109 freq = 44100; 6110 if (low <= freq && freq <= high) { 6111 return freq; 6112 } 6113 return high; 6114 } else { 6115 for (j = 0; j < fmt->frequency_type; j++) { 6116 if (fmt->frequency[j] == 48000) { 6117 return fmt->frequency[j]; 6118 } 6119 } 6120 high = 0; 6121 for (j = 0; j < fmt->frequency_type; j++) { 6122 if (fmt->frequency[j] == 44100) { 6123 return fmt->frequency[j]; 6124 } 6125 if (fmt->frequency[j] > high) { 6126 high = fmt->frequency[j]; 6127 } 6128 } 6129 return high; 6130 } 6131 } 6132 6133 /* 6134 * Probe playback and/or recording format (depending on *modep). 6135 * *modep is an in-out parameter. It indicates the direction to configure 6136 * as an argument, and the direction configured is written back as out 6137 * parameter. 6138 * If successful, probed hardware format is stored into *phwfmt, *rhwfmt 6139 * depending on *modep, and return 0. Otherwise it returns errno. 6140 * Must be called with sc_lock held. 6141 */ 6142 static int 6143 audio_hw_probe(struct audio_softc *sc, int is_indep, int *modep, 6144 audio_format2_t *phwfmt, audio_format2_t *rhwfmt) 6145 { 6146 audio_format2_t fmt; 6147 int mode; 6148 int error = 0; 6149 6150 KASSERT(mutex_owned(sc->sc_lock)); 6151 6152 mode = *modep; 6153 KASSERTMSG((mode & (AUMODE_PLAY | AUMODE_RECORD)) != 0, 6154 "invalid mode = %x", mode); 6155 6156 if (is_indep) { 6157 int errorp = 0, errorr = 0; 6158 6159 /* On independent devices, probe separately. */ 6160 if ((mode & AUMODE_PLAY) != 0) { 6161 errorp = audio_hw_probe_fmt(sc, phwfmt, AUMODE_PLAY); 6162 if (errorp) 6163 mode &= ~AUMODE_PLAY; 6164 } 6165 if ((mode & AUMODE_RECORD) != 0) { 6166 errorr = audio_hw_probe_fmt(sc, rhwfmt, AUMODE_RECORD); 6167 if (errorr) 6168 mode &= ~AUMODE_RECORD; 6169 } 6170 6171 /* Return error if both play and record probes failed. */ 6172 if (errorp && errorr) 6173 error = errorp; 6174 } else { 6175 /* On non independent devices, probe simultaneously. */ 6176 error = audio_hw_probe_fmt(sc, &fmt, mode); 6177 if (error) { 6178 mode = 0; 6179 } else { 6180 *phwfmt = fmt; 6181 *rhwfmt = fmt; 6182 } 6183 } 6184 6185 *modep = mode; 6186 return error; 6187 } 6188 6189 /* 6190 * Choose the most preferred hardware format. 6191 * If successful, it will store the chosen format into *cand and return 0. 6192 * Otherwise, return errno. 6193 * Must be called with sc_lock held. 6194 */ 6195 static int 6196 audio_hw_probe_fmt(struct audio_softc *sc, audio_format2_t *cand, int mode) 6197 { 6198 audio_format_query_t query; 6199 int cand_score; 6200 int score; 6201 int i; 6202 int error; 6203 6204 KASSERT(mutex_owned(sc->sc_lock)); 6205 6206 /* 6207 * Score each formats and choose the highest one. 6208 * 6209 * +---- priority(0-3) 6210 * |+--- encoding/precision 6211 * ||+-- channels 6212 * score = 0x000000PEC 6213 */ 6214 6215 cand_score = 0; 6216 for (i = 0; ; i++) { 6217 memset(&query, 0, sizeof(query)); 6218 query.index = i; 6219 6220 error = sc->hw_if->query_format(sc->hw_hdl, &query); 6221 if (error == EINVAL) 6222 break; 6223 if (error) 6224 return error; 6225 6226 #if defined(AUDIO_DEBUG) 6227 DPRINTF(1, "fmt[%d] %c%c pri=%d %s,%d/%dbit,%dch,", i, 6228 (query.fmt.mode & AUMODE_PLAY) ? 'P' : '-', 6229 (query.fmt.mode & AUMODE_RECORD) ? 'R' : '-', 6230 query.fmt.priority, 6231 audio_encoding_name(query.fmt.encoding), 6232 query.fmt.validbits, 6233 query.fmt.precision, 6234 query.fmt.channels); 6235 if (query.fmt.frequency_type == 0) { 6236 DPRINTF(1, "{%d-%d", 6237 query.fmt.frequency[0], query.fmt.frequency[1]); 6238 } else { 6239 int j; 6240 for (j = 0; j < query.fmt.frequency_type; j++) { 6241 DPRINTF(1, "%c%d", 6242 (j == 0) ? '{' : ',', 6243 query.fmt.frequency[j]); 6244 } 6245 } 6246 DPRINTF(1, "}\n"); 6247 #endif 6248 6249 if ((query.fmt.mode & mode) == 0) { 6250 DPRINTF(1, "fmt[%d] skip; mode not match %d\n", i, 6251 mode); 6252 continue; 6253 } 6254 6255 if (query.fmt.priority < 0) { 6256 DPRINTF(1, "fmt[%d] skip; unsupported encoding\n", i); 6257 continue; 6258 } 6259 6260 /* Score */ 6261 score = (query.fmt.priority & 3) * 0x100; 6262 if (query.fmt.encoding == AUDIO_ENCODING_SLINEAR_NE && 6263 query.fmt.validbits == AUDIO_INTERNAL_BITS && 6264 query.fmt.precision == AUDIO_INTERNAL_BITS) { 6265 score += 0x20; 6266 } else if (query.fmt.encoding == AUDIO_ENCODING_SLINEAR_OE && 6267 query.fmt.validbits == AUDIO_INTERNAL_BITS && 6268 query.fmt.precision == AUDIO_INTERNAL_BITS) { 6269 score += 0x10; 6270 } 6271 score += query.fmt.channels; 6272 6273 if (score < cand_score) { 6274 DPRINTF(1, "fmt[%d] skip; score 0x%x < 0x%x\n", i, 6275 score, cand_score); 6276 continue; 6277 } 6278 6279 /* Update candidate */ 6280 cand_score = score; 6281 cand->encoding = query.fmt.encoding; 6282 cand->precision = query.fmt.validbits; 6283 cand->stride = query.fmt.precision; 6284 cand->channels = query.fmt.channels; 6285 cand->sample_rate = audio_select_freq(&query.fmt); 6286 DPRINTF(1, "fmt[%d] candidate (score=0x%x)" 6287 " pri=%d %s,%d/%d,%dch,%dHz\n", i, 6288 cand_score, query.fmt.priority, 6289 audio_encoding_name(query.fmt.encoding), 6290 cand->precision, cand->stride, 6291 cand->channels, cand->sample_rate); 6292 } 6293 6294 if (cand_score == 0) { 6295 DPRINTF(1, "%s no fmt\n", __func__); 6296 return ENXIO; 6297 } 6298 DPRINTF(1, "%s selected: %s,%d/%d,%dch,%dHz\n", __func__, 6299 audio_encoding_name(cand->encoding), 6300 cand->precision, cand->stride, cand->channels, cand->sample_rate); 6301 return 0; 6302 } 6303 6304 /* 6305 * Validate fmt with query_format. 6306 * If fmt is included in the result of query_format, returns 0. 6307 * Otherwise returns EINVAL. 6308 * Must be called with sc_lock held. 6309 */ 6310 static int 6311 audio_hw_validate_format(struct audio_softc *sc, int mode, 6312 const audio_format2_t *fmt) 6313 { 6314 audio_format_query_t query; 6315 struct audio_format *q; 6316 int index; 6317 int error; 6318 int j; 6319 6320 KASSERT(mutex_owned(sc->sc_lock)); 6321 6322 /* 6323 * If query_format is not supported by hardware driver, 6324 * a rough check instead will be performed. 6325 * XXX This will gone in the future. 6326 */ 6327 if (sc->hw_if->query_format == NULL) { 6328 if (fmt->encoding != AUDIO_ENCODING_SLINEAR_NE) 6329 return EINVAL; 6330 if (fmt->precision != AUDIO_INTERNAL_BITS) 6331 return EINVAL; 6332 if (fmt->stride != AUDIO_INTERNAL_BITS) 6333 return EINVAL; 6334 return 0; 6335 } 6336 6337 for (index = 0; ; index++) { 6338 query.index = index; 6339 error = sc->hw_if->query_format(sc->hw_hdl, &query); 6340 if (error == EINVAL) 6341 break; 6342 if (error) 6343 return error; 6344 6345 q = &query.fmt; 6346 /* 6347 * Note that fmt is audio_format2_t (precision/stride) but 6348 * q is audio_format_t (validbits/precision). 6349 */ 6350 if ((q->mode & mode) == 0) { 6351 continue; 6352 } 6353 if (fmt->encoding != q->encoding) { 6354 continue; 6355 } 6356 if (fmt->precision != q->validbits) { 6357 continue; 6358 } 6359 if (fmt->stride != q->precision) { 6360 continue; 6361 } 6362 if (fmt->channels != q->channels) { 6363 continue; 6364 } 6365 if (q->frequency_type == 0) { 6366 if (fmt->sample_rate < q->frequency[0] || 6367 fmt->sample_rate > q->frequency[1]) { 6368 continue; 6369 } 6370 } else { 6371 for (j = 0; j < q->frequency_type; j++) { 6372 if (fmt->sample_rate == q->frequency[j]) 6373 break; 6374 } 6375 if (j == query.fmt.frequency_type) { 6376 continue; 6377 } 6378 } 6379 6380 /* Matched. */ 6381 return 0; 6382 } 6383 6384 return EINVAL; 6385 } 6386 6387 /* 6388 * Set track mixer's format depending on ai->mode. 6389 * If AUMODE_PLAY is set in ai->mode, it set up the playback mixer 6390 * with ai.play.{channels, sample_rate}. 6391 * If AUMODE_RECORD is set in ai->mode, it set up the recording mixer 6392 * with ai.record.{channels, sample_rate}. 6393 * All other fields in ai are ignored. 6394 * If successful returns 0. Otherwise returns errno. 6395 * This function does not roll back even if it fails. 6396 * Must be called with sc_lock held. 6397 */ 6398 static int 6399 audio_mixers_set_format(struct audio_softc *sc, const struct audio_info *ai) 6400 { 6401 audio_format2_t phwfmt; 6402 audio_format2_t rhwfmt; 6403 audio_filter_reg_t pfil; 6404 audio_filter_reg_t rfil; 6405 int mode; 6406 int props; 6407 int error; 6408 6409 KASSERT(mutex_owned(sc->sc_lock)); 6410 6411 /* 6412 * Even when setting either one of playback and recording, 6413 * both must be halted. 6414 */ 6415 if (sc->sc_popens + sc->sc_ropens > 0) 6416 return EBUSY; 6417 6418 if (!SPECIFIED(ai->mode) || ai->mode == 0) 6419 return ENOTTY; 6420 6421 /* Only channels and sample_rate are changeable. */ 6422 mode = ai->mode; 6423 if ((mode & AUMODE_PLAY)) { 6424 phwfmt.encoding = ai->play.encoding; 6425 phwfmt.precision = ai->play.precision; 6426 phwfmt.stride = ai->play.precision; 6427 phwfmt.channels = ai->play.channels; 6428 phwfmt.sample_rate = ai->play.sample_rate; 6429 } 6430 if ((mode & AUMODE_RECORD)) { 6431 rhwfmt.encoding = ai->record.encoding; 6432 rhwfmt.precision = ai->record.precision; 6433 rhwfmt.stride = ai->record.precision; 6434 rhwfmt.channels = ai->record.channels; 6435 rhwfmt.sample_rate = ai->record.sample_rate; 6436 } 6437 6438 /* On non-independent devices, use the same format for both. */ 6439 props = audio_get_props(sc); 6440 if ((props & AUDIO_PROP_INDEPENDENT) == 0) { 6441 if (mode == AUMODE_RECORD) { 6442 phwfmt = rhwfmt; 6443 } else { 6444 rhwfmt = phwfmt; 6445 } 6446 mode = AUMODE_PLAY | AUMODE_RECORD; 6447 } 6448 6449 /* Then, unset the direction not exist on the hardware. */ 6450 if ((props & AUDIO_PROP_PLAYBACK) == 0) 6451 mode &= ~AUMODE_PLAY; 6452 if ((props & AUDIO_PROP_CAPTURE) == 0) 6453 mode &= ~AUMODE_RECORD; 6454 6455 /* debug */ 6456 if ((mode & AUMODE_PLAY)) { 6457 TRACE(1, "play=%s/%d/%d/%dch/%dHz", 6458 audio_encoding_name(phwfmt.encoding), 6459 phwfmt.precision, 6460 phwfmt.stride, 6461 phwfmt.channels, 6462 phwfmt.sample_rate); 6463 } 6464 if ((mode & AUMODE_RECORD)) { 6465 TRACE(1, "rec =%s/%d/%d/%dch/%dHz", 6466 audio_encoding_name(rhwfmt.encoding), 6467 rhwfmt.precision, 6468 rhwfmt.stride, 6469 rhwfmt.channels, 6470 rhwfmt.sample_rate); 6471 } 6472 6473 /* Check the format */ 6474 if ((mode & AUMODE_PLAY)) { 6475 if (audio_hw_validate_format(sc, AUMODE_PLAY, &phwfmt)) { 6476 TRACE(1, "invalid format"); 6477 return EINVAL; 6478 } 6479 } 6480 if ((mode & AUMODE_RECORD)) { 6481 if (audio_hw_validate_format(sc, AUMODE_RECORD, &rhwfmt)) { 6482 TRACE(1, "invalid format"); 6483 return EINVAL; 6484 } 6485 } 6486 6487 /* Configure the mixers. */ 6488 memset(&pfil, 0, sizeof(pfil)); 6489 memset(&rfil, 0, sizeof(rfil)); 6490 error = audio_hw_set_format(sc, mode, &phwfmt, &rhwfmt, &pfil, &rfil); 6491 if (error) 6492 return error; 6493 6494 error = audio_mixers_init(sc, mode, &phwfmt, &rhwfmt, &pfil, &rfil); 6495 if (error) 6496 return error; 6497 6498 return 0; 6499 } 6500 6501 /* 6502 * Store current mixers format into *ai. 6503 */ 6504 static void 6505 audio_mixers_get_format(struct audio_softc *sc, struct audio_info *ai) 6506 { 6507 /* 6508 * There is no stride information in audio_info but it doesn't matter. 6509 * trackmixer always treats stride and precision as the same. 6510 */ 6511 AUDIO_INITINFO(ai); 6512 ai->mode = 0; 6513 if (sc->sc_pmixer) { 6514 audio_format2_t *fmt = &sc->sc_pmixer->track_fmt; 6515 ai->play.encoding = fmt->encoding; 6516 ai->play.precision = fmt->precision; 6517 ai->play.channels = fmt->channels; 6518 ai->play.sample_rate = fmt->sample_rate; 6519 ai->mode |= AUMODE_PLAY; 6520 } 6521 if (sc->sc_rmixer) { 6522 audio_format2_t *fmt = &sc->sc_rmixer->track_fmt; 6523 ai->record.encoding = fmt->encoding; 6524 ai->record.precision = fmt->precision; 6525 ai->record.channels = fmt->channels; 6526 ai->record.sample_rate = fmt->sample_rate; 6527 ai->mode |= AUMODE_RECORD; 6528 } 6529 } 6530 6531 /* 6532 * audio_info details: 6533 * 6534 * ai.{play,record}.sample_rate (R/W) 6535 * ai.{play,record}.encoding (R/W) 6536 * ai.{play,record}.precision (R/W) 6537 * ai.{play,record}.channels (R/W) 6538 * These specify the playback or recording format. 6539 * Ignore members within an inactive track. 6540 * 6541 * ai.mode (R/W) 6542 * It specifies the playback or recording mode, AUMODE_*. 6543 * Currently, a mode change operation by ai.mode after opening is 6544 * prohibited. In addition, AUMODE_PLAY_ALL no longer makes sense. 6545 * However, it's possible to get or to set for backward compatibility. 6546 * 6547 * ai.{hiwat,lowat} (R/W) 6548 * These specify the high water mark and low water mark for playback 6549 * track. The unit is block. 6550 * 6551 * ai.{play,record}.gain (R/W) 6552 * It specifies the HW mixer volume in 0-255. 6553 * It is historical reason that the gain is connected to HW mixer. 6554 * 6555 * ai.{play,record}.balance (R/W) 6556 * It specifies the left-right balance of HW mixer in 0-64. 6557 * 32 means the center. 6558 * It is historical reason that the balance is connected to HW mixer. 6559 * 6560 * ai.{play,record}.port (R/W) 6561 * It specifies the input/output port of HW mixer. 6562 * 6563 * ai.monitor_gain (R/W) 6564 * It specifies the recording monitor gain(?) of HW mixer. 6565 * 6566 * ai.{play,record}.pause (R/W) 6567 * Non-zero means the track is paused. 6568 * 6569 * ai.play.seek (R/-) 6570 * It indicates the number of bytes written but not processed. 6571 * ai.record.seek (R/-) 6572 * It indicates the number of bytes to be able to read. 6573 * 6574 * ai.{play,record}.avail_ports (R/-) 6575 * Mixer info. 6576 * 6577 * ai.{play,record}.buffer_size (R/-) 6578 * It indicates the buffer size in bytes. Internally it means usrbuf. 6579 * 6580 * ai.{play,record}.samples (R/-) 6581 * It indicates the total number of bytes played or recorded. 6582 * 6583 * ai.{play,record}.eof (R/-) 6584 * It indicates the number of times reached EOF(?). 6585 * 6586 * ai.{play,record}.error (R/-) 6587 * Non-zero indicates overflow/underflow has occured. 6588 * 6589 * ai.{play,record}.waiting (R/-) 6590 * Non-zero indicates that other process waits to open. 6591 * It will never happen anymore. 6592 * 6593 * ai.{play,record}.open (R/-) 6594 * Non-zero indicates the direction is opened by this process(?). 6595 * XXX Is this better to indicate that "the device is opened by 6596 * at least one process"? 6597 * 6598 * ai.{play,record}.active (R/-) 6599 * Non-zero indicates that I/O is currently active. 6600 * 6601 * ai.blocksize (R/-) 6602 * It indicates the block size in bytes. 6603 * XXX The blocksize of playback and recording may be different. 6604 */ 6605 6606 /* 6607 * Pause consideration: 6608 * 6609 * The introduction of these two behavior makes pause/unpause operation 6610 * simple. 6611 * 1. The first read/write access of the first track makes mixer start. 6612 * 2. A pause of the last track doesn't make mixer stop. 6613 */ 6614 6615 /* 6616 * Set both track's parameters within a file depending on ai. 6617 * Update sc_sound_[pr]* if set. 6618 * Must be called with sc_lock and sc_exlock held. 6619 */ 6620 static int 6621 audio_file_setinfo(struct audio_softc *sc, audio_file_t *file, 6622 const struct audio_info *ai) 6623 { 6624 const struct audio_prinfo *pi; 6625 const struct audio_prinfo *ri; 6626 audio_track_t *ptrack; 6627 audio_track_t *rtrack; 6628 audio_format2_t pfmt; 6629 audio_format2_t rfmt; 6630 int pchanges; 6631 int rchanges; 6632 int mode; 6633 struct audio_info saved_ai; 6634 audio_format2_t saved_pfmt; 6635 audio_format2_t saved_rfmt; 6636 int error; 6637 6638 KASSERT(mutex_owned(sc->sc_lock)); 6639 KASSERT(sc->sc_exlock); 6640 6641 pi = &ai->play; 6642 ri = &ai->record; 6643 pchanges = 0; 6644 rchanges = 0; 6645 6646 ptrack = file->ptrack; 6647 rtrack = file->rtrack; 6648 6649 #if defined(AUDIO_DEBUG) 6650 if (audiodebug >= 2) { 6651 char buf[256]; 6652 char p[64]; 6653 int buflen; 6654 int plen; 6655 #define SPRINTF(var, fmt...) do { \ 6656 var##len += snprintf(var + var##len, sizeof(var) - var##len, fmt); \ 6657 } while (0) 6658 6659 buflen = 0; 6660 plen = 0; 6661 if (SPECIFIED(pi->encoding)) 6662 SPRINTF(p, "/%s", audio_encoding_name(pi->encoding)); 6663 if (SPECIFIED(pi->precision)) 6664 SPRINTF(p, "/%dbit", pi->precision); 6665 if (SPECIFIED(pi->channels)) 6666 SPRINTF(p, "/%dch", pi->channels); 6667 if (SPECIFIED(pi->sample_rate)) 6668 SPRINTF(p, "/%dHz", pi->sample_rate); 6669 if (plen > 0) 6670 SPRINTF(buf, ",play.param=%s", p + 1); 6671 6672 plen = 0; 6673 if (SPECIFIED(ri->encoding)) 6674 SPRINTF(p, "/%s", audio_encoding_name(ri->encoding)); 6675 if (SPECIFIED(ri->precision)) 6676 SPRINTF(p, "/%dbit", ri->precision); 6677 if (SPECIFIED(ri->channels)) 6678 SPRINTF(p, "/%dch", ri->channels); 6679 if (SPECIFIED(ri->sample_rate)) 6680 SPRINTF(p, "/%dHz", ri->sample_rate); 6681 if (plen > 0) 6682 SPRINTF(buf, ",record.param=%s", p + 1); 6683 6684 if (SPECIFIED(ai->mode)) 6685 SPRINTF(buf, ",mode=%d", ai->mode); 6686 if (SPECIFIED(ai->hiwat)) 6687 SPRINTF(buf, ",hiwat=%d", ai->hiwat); 6688 if (SPECIFIED(ai->lowat)) 6689 SPRINTF(buf, ",lowat=%d", ai->lowat); 6690 if (SPECIFIED(ai->play.gain)) 6691 SPRINTF(buf, ",play.gain=%d", ai->play.gain); 6692 if (SPECIFIED(ai->record.gain)) 6693 SPRINTF(buf, ",record.gain=%d", ai->record.gain); 6694 if (SPECIFIED_CH(ai->play.balance)) 6695 SPRINTF(buf, ",play.balance=%d", ai->play.balance); 6696 if (SPECIFIED_CH(ai->record.balance)) 6697 SPRINTF(buf, ",record.balance=%d", ai->record.balance); 6698 if (SPECIFIED(ai->play.port)) 6699 SPRINTF(buf, ",play.port=%d", ai->play.port); 6700 if (SPECIFIED(ai->record.port)) 6701 SPRINTF(buf, ",record.port=%d", ai->record.port); 6702 if (SPECIFIED(ai->monitor_gain)) 6703 SPRINTF(buf, ",monitor_gain=%d", ai->monitor_gain); 6704 if (SPECIFIED_CH(ai->play.pause)) 6705 SPRINTF(buf, ",play.pause=%d", ai->play.pause); 6706 if (SPECIFIED_CH(ai->record.pause)) 6707 SPRINTF(buf, ",record.pause=%d", ai->record.pause); 6708 6709 if (buflen > 0) 6710 TRACE(2, "specified %s", buf + 1); 6711 } 6712 #endif 6713 6714 AUDIO_INITINFO(&saved_ai); 6715 /* XXX shut up gcc */ 6716 memset(&saved_pfmt, 0, sizeof(saved_pfmt)); 6717 memset(&saved_rfmt, 0, sizeof(saved_rfmt)); 6718 6719 /* Set default value and save current parameters */ 6720 if (ptrack) { 6721 pfmt = ptrack->usrbuf.fmt; 6722 saved_pfmt = ptrack->usrbuf.fmt; 6723 saved_ai.play.pause = ptrack->is_pause; 6724 } 6725 if (rtrack) { 6726 rfmt = rtrack->usrbuf.fmt; 6727 saved_rfmt = rtrack->usrbuf.fmt; 6728 saved_ai.record.pause = rtrack->is_pause; 6729 } 6730 saved_ai.mode = file->mode; 6731 6732 /* Overwrite if specified */ 6733 mode = file->mode; 6734 if (SPECIFIED(ai->mode)) { 6735 /* 6736 * Setting ai->mode no longer does anything because it's 6737 * prohibited to change playback/recording mode after open 6738 * and AUMODE_PLAY_ALL is obsoleted. However, it still 6739 * keeps the state of AUMODE_PLAY_ALL itself for backward 6740 * compatibility. 6741 * In the internal, only file->mode has the state of 6742 * AUMODE_PLAY_ALL flag and track->mode in both track does 6743 * not have. 6744 */ 6745 if ((file->mode & AUMODE_PLAY)) { 6746 mode = (file->mode & (AUMODE_PLAY | AUMODE_RECORD)) 6747 | (ai->mode & AUMODE_PLAY_ALL); 6748 } 6749 } 6750 6751 if (ptrack) { 6752 pchanges = audio_track_setinfo_check(&pfmt, pi); 6753 if (pchanges == -1) { 6754 TRACET(1, ptrack, "check play.params failed"); 6755 return EINVAL; 6756 } 6757 if (SPECIFIED(ai->mode)) 6758 pchanges = 1; 6759 } 6760 if (rtrack) { 6761 rchanges = audio_track_setinfo_check(&rfmt, ri); 6762 if (rchanges == -1) { 6763 TRACET(1, rtrack, "check record.params failed"); 6764 return EINVAL; 6765 } 6766 if (SPECIFIED(ai->mode)) 6767 rchanges = 1; 6768 } 6769 6770 /* 6771 * Even when setting either one of playback and recording, 6772 * both track must be halted. 6773 */ 6774 if (pchanges || rchanges) { 6775 audio_file_clear(sc, file); 6776 #if defined(AUDIO_DEBUG) 6777 char fmtbuf[64]; 6778 if (pchanges) { 6779 audio_format2_tostr(fmtbuf, sizeof(fmtbuf), &pfmt); 6780 DPRINTF(1, "audio track#%d play mode: %s\n", 6781 ptrack->id, fmtbuf); 6782 } 6783 if (rchanges) { 6784 audio_format2_tostr(fmtbuf, sizeof(fmtbuf), &rfmt); 6785 DPRINTF(1, "audio track#%d rec mode: %s\n", 6786 rtrack->id, fmtbuf); 6787 } 6788 #endif 6789 } 6790 6791 /* Set mixer parameters */ 6792 error = audio_hw_setinfo(sc, ai, &saved_ai); 6793 if (error) 6794 goto abort1; 6795 6796 /* Set to track and update sticky parameters */ 6797 error = 0; 6798 file->mode = mode; 6799 if (ptrack) { 6800 if (SPECIFIED_CH(pi->pause)) { 6801 ptrack->is_pause = pi->pause; 6802 sc->sc_sound_ppause = pi->pause; 6803 } 6804 if (pchanges) { 6805 audio_track_lock_enter(ptrack); 6806 error = audio_track_set_format(ptrack, &pfmt); 6807 audio_track_lock_exit(ptrack); 6808 if (error) { 6809 TRACET(1, ptrack, "set play.params failed"); 6810 goto abort2; 6811 } 6812 sc->sc_sound_pparams = pfmt; 6813 } 6814 /* Change water marks after initializing the buffers. */ 6815 if (SPECIFIED(ai->hiwat) || SPECIFIED(ai->lowat)) 6816 audio_track_setinfo_water(ptrack, ai); 6817 } 6818 if (rtrack) { 6819 if (SPECIFIED_CH(ri->pause)) { 6820 rtrack->is_pause = ri->pause; 6821 sc->sc_sound_rpause = ri->pause; 6822 } 6823 if (rchanges) { 6824 audio_track_lock_enter(rtrack); 6825 error = audio_track_set_format(rtrack, &rfmt); 6826 audio_track_lock_exit(rtrack); 6827 if (error) { 6828 TRACET(1, rtrack, "set record.params failed"); 6829 goto abort3; 6830 } 6831 sc->sc_sound_rparams = rfmt; 6832 } 6833 } 6834 6835 return 0; 6836 6837 /* Rollback */ 6838 abort3: 6839 if (error != ENOMEM) { 6840 rtrack->is_pause = saved_ai.record.pause; 6841 audio_track_lock_enter(rtrack); 6842 audio_track_set_format(rtrack, &saved_rfmt); 6843 audio_track_lock_exit(rtrack); 6844 } 6845 abort2: 6846 if (ptrack && error != ENOMEM) { 6847 ptrack->is_pause = saved_ai.play.pause; 6848 audio_track_lock_enter(ptrack); 6849 audio_track_set_format(ptrack, &saved_pfmt); 6850 audio_track_lock_exit(ptrack); 6851 sc->sc_sound_pparams = saved_pfmt; 6852 sc->sc_sound_ppause = saved_ai.play.pause; 6853 } 6854 file->mode = saved_ai.mode; 6855 abort1: 6856 audio_hw_setinfo(sc, &saved_ai, NULL); 6857 6858 return error; 6859 } 6860 6861 /* 6862 * Write SPECIFIED() parameters within info back to fmt. 6863 * Return value of 1 indicates that fmt is modified. 6864 * Return value of 0 indicates that fmt is not modified. 6865 * Return value of -1 indicates that error EINVAL has occurred. 6866 */ 6867 static int 6868 audio_track_setinfo_check(audio_format2_t *fmt, const struct audio_prinfo *info) 6869 { 6870 int changes; 6871 6872 changes = 0; 6873 if (SPECIFIED(info->sample_rate)) { 6874 if (info->sample_rate < AUDIO_MIN_FREQUENCY) 6875 return -1; 6876 if (info->sample_rate > AUDIO_MAX_FREQUENCY) 6877 return -1; 6878 fmt->sample_rate = info->sample_rate; 6879 changes = 1; 6880 } 6881 if (SPECIFIED(info->encoding)) { 6882 fmt->encoding = info->encoding; 6883 changes = 1; 6884 } 6885 if (SPECIFIED(info->precision)) { 6886 fmt->precision = info->precision; 6887 /* we don't have API to specify stride */ 6888 fmt->stride = info->precision; 6889 changes = 1; 6890 } 6891 if (SPECIFIED(info->channels)) { 6892 fmt->channels = info->channels; 6893 changes = 1; 6894 } 6895 6896 if (changes) { 6897 if (audio_check_params(fmt) != 0) { 6898 #ifdef DIAGNOSTIC 6899 char fmtbuf[64]; 6900 audio_format2_tostr(fmtbuf, sizeof(fmtbuf), fmt); 6901 printf("%s failed: %s\n", __func__, fmtbuf); 6902 #endif 6903 return -1; 6904 } 6905 } 6906 6907 return changes; 6908 } 6909 6910 /* 6911 * Change water marks for playback track if specfied. 6912 */ 6913 static void 6914 audio_track_setinfo_water(audio_track_t *track, const struct audio_info *ai) 6915 { 6916 u_int blks; 6917 u_int maxblks; 6918 u_int blksize; 6919 6920 KASSERT(audio_track_is_playback(track)); 6921 6922 blksize = track->usrbuf_blksize; 6923 maxblks = track->usrbuf.capacity / blksize; 6924 6925 if (SPECIFIED(ai->hiwat)) { 6926 blks = ai->hiwat; 6927 if (blks > maxblks) 6928 blks = maxblks; 6929 if (blks < 2) 6930 blks = 2; 6931 track->usrbuf_usedhigh = blks * blksize; 6932 } 6933 if (SPECIFIED(ai->lowat)) { 6934 blks = ai->lowat; 6935 if (blks > maxblks - 1) 6936 blks = maxblks - 1; 6937 track->usrbuf_usedlow = blks * blksize; 6938 } 6939 if (SPECIFIED(ai->hiwat) || SPECIFIED(ai->lowat)) { 6940 if (track->usrbuf_usedlow > track->usrbuf_usedhigh - blksize) { 6941 track->usrbuf_usedlow = track->usrbuf_usedhigh - 6942 blksize; 6943 } 6944 } 6945 } 6946 6947 /* 6948 * Set hardware part of *ai. 6949 * The parameters handled here are *.port, *.gain, *.balance and monitor_gain. 6950 * If oldai is specified, previous parameters are stored. 6951 * This function itself does not roll back if error occurred. 6952 * Must be called with sc_lock and sc_exlock held. 6953 */ 6954 static int 6955 audio_hw_setinfo(struct audio_softc *sc, const struct audio_info *newai, 6956 struct audio_info *oldai) 6957 { 6958 const struct audio_prinfo *newpi; 6959 const struct audio_prinfo *newri; 6960 struct audio_prinfo *oldpi; 6961 struct audio_prinfo *oldri; 6962 u_int pgain; 6963 u_int rgain; 6964 u_char pbalance; 6965 u_char rbalance; 6966 int error; 6967 6968 KASSERT(mutex_owned(sc->sc_lock)); 6969 KASSERT(sc->sc_exlock); 6970 6971 /* XXX shut up gcc */ 6972 oldpi = NULL; 6973 oldri = NULL; 6974 6975 newpi = &newai->play; 6976 newri = &newai->record; 6977 if (oldai) { 6978 oldpi = &oldai->play; 6979 oldri = &oldai->record; 6980 } 6981 error = 0; 6982 6983 /* 6984 * It looks like unnecessary to halt HW mixers to set HW mixers. 6985 * mixer_ioctl(MIXER_WRITE) also doesn't halt. 6986 */ 6987 6988 if (SPECIFIED(newpi->port)) { 6989 if (oldai) 6990 oldpi->port = au_get_port(sc, &sc->sc_outports); 6991 error = au_set_port(sc, &sc->sc_outports, newpi->port); 6992 if (error) { 6993 device_printf(sc->sc_dev, 6994 "setting play.port=%d failed with %d\n", 6995 newpi->port, error); 6996 goto abort; 6997 } 6998 } 6999 if (SPECIFIED(newri->port)) { 7000 if (oldai) 7001 oldri->port = au_get_port(sc, &sc->sc_inports); 7002 error = au_set_port(sc, &sc->sc_inports, newri->port); 7003 if (error) { 7004 device_printf(sc->sc_dev, 7005 "setting record.port=%d failed with %d\n", 7006 newri->port, error); 7007 goto abort; 7008 } 7009 } 7010 7011 /* Backup play.{gain,balance} */ 7012 if (SPECIFIED(newpi->gain) || SPECIFIED_CH(newpi->balance)) { 7013 au_get_gain(sc, &sc->sc_outports, &pgain, &pbalance); 7014 if (oldai) { 7015 oldpi->gain = pgain; 7016 oldpi->balance = pbalance; 7017 } 7018 } 7019 /* Backup record.{gain,balance} */ 7020 if (SPECIFIED(newri->gain) || SPECIFIED_CH(newri->balance)) { 7021 au_get_gain(sc, &sc->sc_inports, &rgain, &rbalance); 7022 if (oldai) { 7023 oldri->gain = rgain; 7024 oldri->balance = rbalance; 7025 } 7026 } 7027 if (SPECIFIED(newpi->gain)) { 7028 error = au_set_gain(sc, &sc->sc_outports, 7029 newpi->gain, pbalance); 7030 if (error) { 7031 device_printf(sc->sc_dev, 7032 "setting play.gain=%d failed with %d\n", 7033 newpi->gain, error); 7034 goto abort; 7035 } 7036 } 7037 if (SPECIFIED(newri->gain)) { 7038 error = au_set_gain(sc, &sc->sc_inports, 7039 newri->gain, rbalance); 7040 if (error) { 7041 device_printf(sc->sc_dev, 7042 "setting record.gain=%d failed with %d\n", 7043 newri->gain, error); 7044 goto abort; 7045 } 7046 } 7047 if (SPECIFIED_CH(newpi->balance)) { 7048 error = au_set_gain(sc, &sc->sc_outports, 7049 pgain, newpi->balance); 7050 if (error) { 7051 device_printf(sc->sc_dev, 7052 "setting play.balance=%d failed with %d\n", 7053 newpi->balance, error); 7054 goto abort; 7055 } 7056 } 7057 if (SPECIFIED_CH(newri->balance)) { 7058 error = au_set_gain(sc, &sc->sc_inports, 7059 rgain, newri->balance); 7060 if (error) { 7061 device_printf(sc->sc_dev, 7062 "setting record.balance=%d failed with %d\n", 7063 newri->balance, error); 7064 goto abort; 7065 } 7066 } 7067 7068 if (SPECIFIED(newai->monitor_gain) && sc->sc_monitor_port != -1) { 7069 if (oldai) 7070 oldai->monitor_gain = au_get_monitor_gain(sc); 7071 error = au_set_monitor_gain(sc, newai->monitor_gain); 7072 if (error) { 7073 device_printf(sc->sc_dev, 7074 "setting monitor_gain=%d failed with %d\n", 7075 newai->monitor_gain, error); 7076 goto abort; 7077 } 7078 } 7079 7080 /* XXX TODO */ 7081 /* sc->sc_ai = *ai; */ 7082 7083 error = 0; 7084 abort: 7085 return error; 7086 } 7087 7088 /* 7089 * Setup the hardware with mixer format phwfmt, rhwfmt. 7090 * The arguments have following restrictions: 7091 * - setmode is the direction you want to set, AUMODE_PLAY or AUMODE_RECORD, 7092 * or both. 7093 * - phwfmt and rhwfmt must not be NULL regardless of setmode. 7094 * - On non-independent devices, phwfmt and rhwfmt must have the same 7095 * parameters. 7096 * - pfil and rfil must be zero-filled. 7097 * If successful, 7098 * - phwfmt, rhwfmt will be overwritten by hardware format. 7099 * - pfil, rfil will be filled with filter information specified by the 7100 * hardware driver. 7101 * and then returns 0. Otherwise returns errno. 7102 * Must be called with sc_lock held. 7103 */ 7104 static int 7105 audio_hw_set_format(struct audio_softc *sc, int setmode, 7106 audio_format2_t *phwfmt, audio_format2_t *rhwfmt, 7107 audio_filter_reg_t *pfil, audio_filter_reg_t *rfil) 7108 { 7109 audio_params_t pp, rp; 7110 int error; 7111 7112 KASSERT(mutex_owned(sc->sc_lock)); 7113 KASSERT(phwfmt != NULL); 7114 KASSERT(rhwfmt != NULL); 7115 7116 pp = format2_to_params(phwfmt); 7117 rp = format2_to_params(rhwfmt); 7118 7119 error = sc->hw_if->set_format(sc->hw_hdl, setmode, 7120 &pp, &rp, pfil, rfil); 7121 if (error) { 7122 device_printf(sc->sc_dev, 7123 "set_format failed with %d\n", error); 7124 return error; 7125 } 7126 7127 if (sc->hw_if->commit_settings) { 7128 error = sc->hw_if->commit_settings(sc->hw_hdl); 7129 if (error) { 7130 device_printf(sc->sc_dev, 7131 "commit_settings failed with %d\n", error); 7132 return error; 7133 } 7134 } 7135 7136 return 0; 7137 } 7138 7139 /* 7140 * Fill audio_info structure. If need_mixerinfo is true, it will also 7141 * fill the hardware mixer information. 7142 * Must be called with sc_lock held. 7143 * Must be called with sc_exlock held, in addition, if need_mixerinfo is 7144 * true. 7145 */ 7146 static int 7147 audiogetinfo(struct audio_softc *sc, struct audio_info *ai, int need_mixerinfo, 7148 audio_file_t *file) 7149 { 7150 struct audio_prinfo *ri, *pi; 7151 audio_track_t *track; 7152 audio_track_t *ptrack; 7153 audio_track_t *rtrack; 7154 int gain; 7155 7156 KASSERT(mutex_owned(sc->sc_lock)); 7157 7158 ri = &ai->record; 7159 pi = &ai->play; 7160 ptrack = file->ptrack; 7161 rtrack = file->rtrack; 7162 7163 memset(ai, 0, sizeof(*ai)); 7164 7165 if (ptrack) { 7166 pi->sample_rate = ptrack->usrbuf.fmt.sample_rate; 7167 pi->channels = ptrack->usrbuf.fmt.channels; 7168 pi->precision = ptrack->usrbuf.fmt.precision; 7169 pi->encoding = ptrack->usrbuf.fmt.encoding; 7170 } else { 7171 /* Set default parameters if the track is not available. */ 7172 if (ISDEVAUDIO(file->dev)) { 7173 pi->sample_rate = audio_default.sample_rate; 7174 pi->channels = audio_default.channels; 7175 pi->precision = audio_default.precision; 7176 pi->encoding = audio_default.encoding; 7177 } else { 7178 pi->sample_rate = sc->sc_sound_pparams.sample_rate; 7179 pi->channels = sc->sc_sound_pparams.channels; 7180 pi->precision = sc->sc_sound_pparams.precision; 7181 pi->encoding = sc->sc_sound_pparams.encoding; 7182 } 7183 } 7184 if (rtrack) { 7185 ri->sample_rate = rtrack->usrbuf.fmt.sample_rate; 7186 ri->channels = rtrack->usrbuf.fmt.channels; 7187 ri->precision = rtrack->usrbuf.fmt.precision; 7188 ri->encoding = rtrack->usrbuf.fmt.encoding; 7189 } else { 7190 /* Set default parameters if the track is not available. */ 7191 if (ISDEVAUDIO(file->dev)) { 7192 ri->sample_rate = audio_default.sample_rate; 7193 ri->channels = audio_default.channels; 7194 ri->precision = audio_default.precision; 7195 ri->encoding = audio_default.encoding; 7196 } else { 7197 ri->sample_rate = sc->sc_sound_rparams.sample_rate; 7198 ri->channels = sc->sc_sound_rparams.channels; 7199 ri->precision = sc->sc_sound_rparams.precision; 7200 ri->encoding = sc->sc_sound_rparams.encoding; 7201 } 7202 } 7203 7204 if (ptrack) { 7205 pi->seek = ptrack->usrbuf.used; 7206 pi->samples = ptrack->usrbuf_stamp; 7207 pi->eof = ptrack->eofcounter; 7208 pi->pause = ptrack->is_pause; 7209 pi->error = (ptrack->dropframes != 0) ? 1 : 0; 7210 pi->waiting = 0; /* open never hangs */ 7211 pi->open = 1; 7212 pi->active = sc->sc_pbusy; 7213 pi->buffer_size = ptrack->usrbuf.capacity; 7214 } 7215 if (rtrack) { 7216 ri->seek = rtrack->usrbuf.used; 7217 ri->samples = rtrack->usrbuf_stamp; 7218 ri->eof = 0; 7219 ri->pause = rtrack->is_pause; 7220 ri->error = (rtrack->dropframes != 0) ? 1 : 0; 7221 ri->waiting = 0; /* open never hangs */ 7222 ri->open = 1; 7223 ri->active = sc->sc_rbusy; 7224 ri->buffer_size = rtrack->usrbuf.capacity; 7225 } 7226 7227 /* 7228 * XXX There may be different number of channels between playback 7229 * and recording, so that blocksize also may be different. 7230 * But struct audio_info has an united blocksize... 7231 * Here, I use play info precedencely if ptrack is available, 7232 * otherwise record info. 7233 * 7234 * XXX hiwat/lowat is a playback-only parameter. What should I 7235 * return for a record-only descriptor? 7236 */ 7237 track = ptrack ? ptrack : rtrack; 7238 if (track) { 7239 ai->blocksize = track->usrbuf_blksize; 7240 ai->hiwat = track->usrbuf_usedhigh / track->usrbuf_blksize; 7241 ai->lowat = track->usrbuf_usedlow / track->usrbuf_blksize; 7242 } 7243 ai->mode = file->mode; 7244 7245 if (need_mixerinfo) { 7246 KASSERT(sc->sc_exlock); 7247 7248 pi->port = au_get_port(sc, &sc->sc_outports); 7249 ri->port = au_get_port(sc, &sc->sc_inports); 7250 7251 pi->avail_ports = sc->sc_outports.allports; 7252 ri->avail_ports = sc->sc_inports.allports; 7253 7254 au_get_gain(sc, &sc->sc_outports, &pi->gain, &pi->balance); 7255 au_get_gain(sc, &sc->sc_inports, &ri->gain, &ri->balance); 7256 7257 if (sc->sc_monitor_port != -1) { 7258 gain = au_get_monitor_gain(sc); 7259 if (gain != -1) 7260 ai->monitor_gain = gain; 7261 } 7262 } 7263 7264 return 0; 7265 } 7266 7267 /* 7268 * Must be called with sc_lock held. 7269 */ 7270 static int 7271 audio_get_props(struct audio_softc *sc) 7272 { 7273 const struct audio_hw_if *hw; 7274 int props; 7275 7276 KASSERT(mutex_owned(sc->sc_lock)); 7277 7278 hw = sc->hw_if; 7279 props = hw->get_props(sc->hw_hdl); 7280 7281 /* 7282 * For historical reasons, if neither playback nor capture 7283 * properties are reported, assume both are supported. 7284 * XXX Ideally (all) hardware driver should be updated... 7285 */ 7286 if ((props & (AUDIO_PROP_PLAYBACK|AUDIO_PROP_CAPTURE)) == 0) 7287 props |= (AUDIO_PROP_PLAYBACK | AUDIO_PROP_CAPTURE); 7288 7289 /* MMAP is now supported by upper layer. */ 7290 props |= AUDIO_PROP_MMAP; 7291 7292 return props; 7293 } 7294 7295 /* 7296 * Return true if playback is configured. 7297 * This function can be used after audioattach. 7298 */ 7299 static bool 7300 audio_can_playback(struct audio_softc *sc) 7301 { 7302 7303 return (sc->sc_pmixer != NULL); 7304 } 7305 7306 /* 7307 * Return true if recording is configured. 7308 * This function can be used after audioattach. 7309 */ 7310 static bool 7311 audio_can_capture(struct audio_softc *sc) 7312 { 7313 7314 return (sc->sc_rmixer != NULL); 7315 } 7316 7317 /* 7318 * Get the afp->index'th item from the valid one of format[]. 7319 * If found, stores it to afp->fmt and returns 0. Otherwise return EINVAL. 7320 * 7321 * This is common routines for query_format. 7322 * If your hardware driver has struct audio_format[], the simplest case 7323 * you can write your query_format interface as follows: 7324 * 7325 * struct audio_format foo_format[] = { ... }; 7326 * 7327 * int 7328 * foo_query_format(void *hdl, audio_format_query_t *afp) 7329 * { 7330 * return audio_query_format(foo_format, __arraycount(foo_format), afp); 7331 * } 7332 */ 7333 int 7334 audio_query_format(const struct audio_format *format, int nformats, 7335 audio_format_query_t *afp) 7336 { 7337 const struct audio_format *f; 7338 int idx; 7339 int i; 7340 7341 idx = 0; 7342 for (i = 0; i < nformats; i++) { 7343 f = &format[i]; 7344 if (!AUFMT_IS_VALID(f)) 7345 continue; 7346 if (afp->index == idx) { 7347 afp->fmt = *f; 7348 return 0; 7349 } 7350 idx++; 7351 } 7352 return EINVAL; 7353 } 7354 7355 /* 7356 * This function is provided for the hardware driver's set_format() to 7357 * find index matches with 'param' from array of audio_format_t 'formats'. 7358 * 'mode' is either of AUMODE_PLAY or AUMODE_RECORD. 7359 * It returns the matched index and never fails. Because param passed to 7360 * set_format() is selected from query_format(). 7361 * This function will be an alternative to auconv_set_converter() to 7362 * find index. 7363 */ 7364 int 7365 audio_indexof_format(const struct audio_format *formats, int nformats, 7366 int mode, const audio_params_t *param) 7367 { 7368 const struct audio_format *f; 7369 int index; 7370 int j; 7371 7372 for (index = 0; index < nformats; index++) { 7373 f = &formats[index]; 7374 7375 if (!AUFMT_IS_VALID(f)) 7376 continue; 7377 if ((f->mode & mode) == 0) 7378 continue; 7379 if (f->encoding != param->encoding) 7380 continue; 7381 if (f->validbits != param->precision) 7382 continue; 7383 if (f->channels != param->channels) 7384 continue; 7385 7386 if (f->frequency_type == 0) { 7387 if (param->sample_rate < f->frequency[0] || 7388 param->sample_rate > f->frequency[1]) 7389 continue; 7390 } else { 7391 for (j = 0; j < f->frequency_type; j++) { 7392 if (param->sample_rate == f->frequency[j]) 7393 break; 7394 } 7395 if (j == f->frequency_type) 7396 continue; 7397 } 7398 7399 /* Then, matched */ 7400 return index; 7401 } 7402 7403 /* Not matched. This should not be happened. */ 7404 panic("%s: cannot find matched format\n", __func__); 7405 } 7406 7407 /* 7408 * Get or set software master volume: 0..256 7409 * XXX It's for debug. 7410 */ 7411 static int 7412 audio_sysctl_volume(SYSCTLFN_ARGS) 7413 { 7414 struct sysctlnode node; 7415 struct audio_softc *sc; 7416 int t, error; 7417 7418 node = *rnode; 7419 sc = node.sysctl_data; 7420 7421 if (sc->sc_pmixer) 7422 t = sc->sc_pmixer->volume; 7423 else 7424 t = -1; 7425 node.sysctl_data = &t; 7426 error = sysctl_lookup(SYSCTLFN_CALL(&node)); 7427 if (error || newp == NULL) 7428 return error; 7429 7430 if (sc->sc_pmixer == NULL) 7431 return EINVAL; 7432 if (t < 0) 7433 return EINVAL; 7434 7435 sc->sc_pmixer->volume = t; 7436 return 0; 7437 } 7438 7439 /* 7440 * Get or set hardware blocksize in msec. 7441 * XXX It's for debug. 7442 */ 7443 static int 7444 audio_sysctl_blk_ms(SYSCTLFN_ARGS) 7445 { 7446 struct sysctlnode node; 7447 struct audio_softc *sc; 7448 audio_format2_t phwfmt; 7449 audio_format2_t rhwfmt; 7450 audio_filter_reg_t pfil; 7451 audio_filter_reg_t rfil; 7452 int t; 7453 int old_blk_ms; 7454 int mode; 7455 int error; 7456 7457 node = *rnode; 7458 sc = node.sysctl_data; 7459 7460 mutex_enter(sc->sc_lock); 7461 7462 old_blk_ms = sc->sc_blk_ms; 7463 t = old_blk_ms; 7464 node.sysctl_data = &t; 7465 error = sysctl_lookup(SYSCTLFN_CALL(&node)); 7466 if (error || newp == NULL) 7467 goto abort; 7468 7469 if (t < 0) { 7470 error = EINVAL; 7471 goto abort; 7472 } 7473 7474 if (sc->sc_popens + sc->sc_ropens > 0) { 7475 error = EBUSY; 7476 goto abort; 7477 } 7478 sc->sc_blk_ms = t; 7479 mode = 0; 7480 if (sc->sc_pmixer) { 7481 mode |= AUMODE_PLAY; 7482 phwfmt = sc->sc_pmixer->hwbuf.fmt; 7483 } 7484 if (sc->sc_rmixer) { 7485 mode |= AUMODE_RECORD; 7486 rhwfmt = sc->sc_rmixer->hwbuf.fmt; 7487 } 7488 7489 /* re-init hardware */ 7490 memset(&pfil, 0, sizeof(pfil)); 7491 memset(&rfil, 0, sizeof(rfil)); 7492 error = audio_hw_set_format(sc, mode, &phwfmt, &rhwfmt, &pfil, &rfil); 7493 if (error) { 7494 goto abort; 7495 } 7496 7497 /* re-init track mixer */ 7498 error = audio_mixers_init(sc, mode, &phwfmt, &rhwfmt, &pfil, &rfil); 7499 if (error) { 7500 /* Rollback */ 7501 sc->sc_blk_ms = old_blk_ms; 7502 audio_mixers_init(sc, mode, &phwfmt, &rhwfmt, &pfil, &rfil); 7503 goto abort; 7504 } 7505 error = 0; 7506 abort: 7507 mutex_exit(sc->sc_lock); 7508 return error; 7509 } 7510 7511 /* 7512 * Get or set multiuser mode. 7513 */ 7514 static int 7515 audio_sysctl_multiuser(SYSCTLFN_ARGS) 7516 { 7517 struct sysctlnode node; 7518 struct audio_softc *sc; 7519 bool t; 7520 int error; 7521 7522 node = *rnode; 7523 sc = node.sysctl_data; 7524 7525 mutex_enter(sc->sc_lock); 7526 7527 t = sc->sc_multiuser; 7528 node.sysctl_data = &t; 7529 error = sysctl_lookup(SYSCTLFN_CALL(&node)); 7530 if (error || newp == NULL) 7531 goto abort; 7532 7533 sc->sc_multiuser = t; 7534 error = 0; 7535 abort: 7536 mutex_exit(sc->sc_lock); 7537 return error; 7538 } 7539 7540 #if defined(AUDIO_DEBUG) 7541 /* 7542 * Get or set debug verbose level. (0..4) 7543 * XXX It's for debug. 7544 * XXX It is not separated per device. 7545 */ 7546 static int 7547 audio_sysctl_debug(SYSCTLFN_ARGS) 7548 { 7549 struct sysctlnode node; 7550 int t; 7551 int error; 7552 7553 node = *rnode; 7554 t = audiodebug; 7555 node.sysctl_data = &t; 7556 error = sysctl_lookup(SYSCTLFN_CALL(&node)); 7557 if (error || newp == NULL) 7558 return error; 7559 7560 if (t < 0 || t > 4) 7561 return EINVAL; 7562 audiodebug = t; 7563 printf("audio: audiodebug = %d\n", audiodebug); 7564 return 0; 7565 } 7566 #endif /* AUDIO_DEBUG */ 7567 7568 #ifdef AUDIO_PM_IDLE 7569 static void 7570 audio_idle(void *arg) 7571 { 7572 device_t dv = arg; 7573 struct audio_softc *sc = device_private(dv); 7574 7575 #ifdef PNP_DEBUG 7576 extern int pnp_debug_idle; 7577 if (pnp_debug_idle) 7578 printf("%s: idle handler called\n", device_xname(dv)); 7579 #endif 7580 7581 sc->sc_idle = true; 7582 7583 /* XXX joerg Make pmf_device_suspend handle children? */ 7584 if (!pmf_device_suspend(dv, PMF_Q_SELF)) 7585 return; 7586 7587 if (!pmf_device_suspend(sc->hw_dev, PMF_Q_SELF)) 7588 pmf_device_resume(dv, PMF_Q_SELF); 7589 } 7590 7591 static void 7592 audio_activity(device_t dv, devactive_t type) 7593 { 7594 struct audio_softc *sc = device_private(dv); 7595 7596 if (type != DVA_SYSTEM) 7597 return; 7598 7599 callout_schedule(&sc->sc_idle_counter, audio_idle_timeout * hz); 7600 7601 sc->sc_idle = false; 7602 if (!device_is_active(dv)) { 7603 /* XXX joerg How to deal with a failing resume... */ 7604 pmf_device_resume(sc->hw_dev, PMF_Q_SELF); 7605 pmf_device_resume(dv, PMF_Q_SELF); 7606 } 7607 } 7608 #endif 7609 7610 static bool 7611 audio_suspend(device_t dv, const pmf_qual_t *qual) 7612 { 7613 struct audio_softc *sc = device_private(dv); 7614 int error; 7615 7616 error = audio_enter_exclusive(sc); 7617 if (error) 7618 return error; 7619 audio_mixer_capture(sc); 7620 7621 /* Halts mixers but don't clear busy flag for resume */ 7622 if (sc->sc_pbusy) { 7623 audio_pmixer_halt(sc); 7624 sc->sc_pbusy = true; 7625 } 7626 if (sc->sc_rbusy) { 7627 audio_rmixer_halt(sc); 7628 sc->sc_rbusy = true; 7629 } 7630 7631 #ifdef AUDIO_PM_IDLE 7632 callout_halt(&sc->sc_idle_counter, sc->sc_lock); 7633 #endif 7634 audio_exit_exclusive(sc); 7635 7636 return true; 7637 } 7638 7639 static bool 7640 audio_resume(device_t dv, const pmf_qual_t *qual) 7641 { 7642 struct audio_softc *sc = device_private(dv); 7643 struct audio_info ai; 7644 int error; 7645 7646 error = audio_enter_exclusive(sc); 7647 if (error) 7648 return error; 7649 7650 audio_mixer_restore(sc); 7651 /* XXX ? */ 7652 AUDIO_INITINFO(&ai); 7653 audio_hw_setinfo(sc, &ai, NULL); 7654 7655 if (sc->sc_pbusy) 7656 audio_pmixer_start(sc, true); 7657 if (sc->sc_rbusy) 7658 audio_rmixer_start(sc); 7659 7660 audio_exit_exclusive(sc); 7661 7662 return true; 7663 } 7664 7665 #if defined(DIAGNOSTIC) || defined(AUDIO_DEBUG) 7666 static void 7667 audio_format2_tostr(char *buf, size_t bufsize, const audio_format2_t *fmt) 7668 { 7669 int n; 7670 7671 n = 0; 7672 n += snprintf(buf + n, bufsize - n, "%s", 7673 audio_encoding_name(fmt->encoding)); 7674 if (fmt->precision == fmt->stride) { 7675 n += snprintf(buf + n, bufsize - n, " %dbit", fmt->precision); 7676 } else { 7677 n += snprintf(buf + n, bufsize - n, " %d/%dbit", 7678 fmt->precision, fmt->stride); 7679 } 7680 7681 snprintf(buf + n, bufsize - n, " %uch %uHz", 7682 fmt->channels, fmt->sample_rate); 7683 } 7684 #endif 7685 7686 #if defined(AUDIO_DEBUG) 7687 static void 7688 audio_print_format2(const char *s, const audio_format2_t *fmt) 7689 { 7690 char fmtstr[64]; 7691 7692 audio_format2_tostr(fmtstr, sizeof(fmtstr), fmt); 7693 printf("%s %s\n", s, fmtstr); 7694 } 7695 #endif 7696 7697 #ifdef DIAGNOSTIC 7698 void 7699 audio_diagnostic_format2(const char *func, const audio_format2_t *fmt) 7700 { 7701 7702 KASSERTMSG(fmt, "%s: fmt == NULL", func); 7703 7704 /* XXX MSM6258 vs(4) only has 4bit stride format. */ 7705 if (fmt->encoding == AUDIO_ENCODING_ADPCM) { 7706 KASSERTMSG(fmt->stride == 4 || fmt->stride == 8, 7707 "%s: stride(%d) is invalid", func, fmt->stride); 7708 } else { 7709 KASSERTMSG(fmt->stride % NBBY == 0, 7710 "%s: stride(%d) is invalid", func, fmt->stride); 7711 } 7712 KASSERTMSG(fmt->precision <= fmt->stride, 7713 "%s: precision(%d) <= stride(%d)", 7714 func, fmt->precision, fmt->stride); 7715 KASSERTMSG(1 <= fmt->channels && fmt->channels <= AUDIO_MAX_CHANNELS, 7716 "%s: channels(%d) is out of range", 7717 func, fmt->channels); 7718 7719 /* XXX No check for encodings? */ 7720 } 7721 7722 void 7723 audio_diagnostic_filter_arg(const char *func, const audio_filter_arg_t *arg) 7724 { 7725 7726 KASSERT(arg != NULL); 7727 KASSERT(arg->src != NULL); 7728 KASSERT(arg->dst != NULL); 7729 DIAGNOSTIC_format2(arg->srcfmt); 7730 DIAGNOSTIC_format2(arg->dstfmt); 7731 KASSERTMSG(arg->count > 0, 7732 "%s: count(%d) is out of range", func, arg->count); 7733 } 7734 7735 void 7736 audio_diagnostic_ring(const char *func, const audio_ring_t *ring) 7737 { 7738 7739 KASSERTMSG(ring, "%s: ring == NULL", func); 7740 DIAGNOSTIC_format2(&ring->fmt); 7741 KASSERTMSG(0 <= ring->capacity && ring->capacity < INT_MAX / 2, 7742 "%s: capacity(%d) is out of range", func, ring->capacity); 7743 KASSERTMSG(0 <= ring->used && ring->used <= ring->capacity, 7744 "%s: used(%d) is out of range (capacity:%d)", 7745 func, ring->used, ring->capacity); 7746 if (ring->capacity == 0) { 7747 KASSERTMSG(ring->mem == NULL, 7748 "%s: capacity == 0 but mem != NULL", func); 7749 } else { 7750 KASSERTMSG(ring->mem != NULL, 7751 "%s: capacity != 0 but mem == NULL", func); 7752 KASSERTMSG(0 <= ring->head && ring->head < ring->capacity, 7753 "%s: head(%d) is out of range (capacity:%d)", 7754 func, ring->head, ring->capacity); 7755 } 7756 } 7757 #endif /* DIAGNOSTIC */ 7758 7759 7760 /* 7761 * Mixer driver 7762 */ 7763 int 7764 mixer_open(dev_t dev, struct audio_softc *sc, int flags, int ifmt, 7765 struct lwp *l) 7766 { 7767 struct file *fp; 7768 audio_file_t *af; 7769 int error, fd; 7770 7771 KASSERT(mutex_owned(sc->sc_lock)); 7772 7773 TRACE(1, "flags=0x%x", flags); 7774 7775 error = fd_allocfile(&fp, &fd); 7776 if (error) 7777 return error; 7778 7779 af = kmem_zalloc(sizeof(*af), KM_SLEEP); 7780 af->sc = sc; 7781 af->dev = dev; 7782 7783 error = fd_clone(fp, fd, flags, &audio_fileops, af); 7784 KASSERT(error == EMOVEFD); 7785 7786 return error; 7787 } 7788 7789 /* 7790 * Remove a process from those to be signalled on mixer activity. 7791 * Must be called with sc_lock held. 7792 */ 7793 static void 7794 mixer_remove(struct audio_softc *sc) 7795 { 7796 struct mixer_asyncs **pm, *m; 7797 pid_t pid; 7798 7799 KASSERT(mutex_owned(sc->sc_lock)); 7800 7801 pid = curproc->p_pid; 7802 for (pm = &sc->sc_async_mixer; *pm; pm = &(*pm)->next) { 7803 if ((*pm)->pid == pid) { 7804 m = *pm; 7805 *pm = m->next; 7806 kmem_free(m, sizeof(*m)); 7807 return; 7808 } 7809 } 7810 } 7811 7812 /* 7813 * Signal all processes waiting for the mixer. 7814 * Must be called with sc_lock held. 7815 */ 7816 static void 7817 mixer_signal(struct audio_softc *sc) 7818 { 7819 struct mixer_asyncs *m; 7820 proc_t *p; 7821 7822 for (m = sc->sc_async_mixer; m; m = m->next) { 7823 mutex_enter(proc_lock); 7824 if ((p = proc_find(m->pid)) != NULL) 7825 psignal(p, SIGIO); 7826 mutex_exit(proc_lock); 7827 } 7828 } 7829 7830 /* 7831 * Close a mixer device 7832 */ 7833 int 7834 mixer_close(struct audio_softc *sc, audio_file_t *file) 7835 { 7836 7837 mutex_enter(sc->sc_lock); 7838 TRACE(1, ""); 7839 mixer_remove(sc); 7840 mutex_exit(sc->sc_lock); 7841 7842 return 0; 7843 } 7844 7845 int 7846 mixer_ioctl(struct audio_softc *sc, u_long cmd, void *addr, int flag, 7847 struct lwp *l) 7848 { 7849 struct mixer_asyncs *ma; 7850 mixer_devinfo_t *mi; 7851 mixer_ctrl_t *mc; 7852 int error; 7853 7854 KASSERT(!mutex_owned(sc->sc_lock)); 7855 7856 TRACE(2, "(%lu,'%c',%lu)", 7857 IOCPARM_LEN(cmd), (char)IOCGROUP(cmd), cmd & 0xff); 7858 error = EINVAL; 7859 7860 /* we can return cached values if we are sleeping */ 7861 if (cmd != AUDIO_MIXER_READ) { 7862 mutex_enter(sc->sc_lock); 7863 device_active(sc->sc_dev, DVA_SYSTEM); 7864 mutex_exit(sc->sc_lock); 7865 } 7866 7867 switch (cmd) { 7868 case FIOASYNC: 7869 if (*(int *)addr) { 7870 ma = kmem_alloc(sizeof(struct mixer_asyncs), KM_SLEEP); 7871 } else { 7872 ma = NULL; 7873 } 7874 mixer_remove(sc); /* remove old entry */ 7875 if (ma != NULL) { 7876 ma->next = sc->sc_async_mixer; 7877 ma->pid = curproc->p_pid; 7878 sc->sc_async_mixer = ma; 7879 } 7880 error = 0; 7881 break; 7882 7883 case AUDIO_GETDEV: 7884 TRACE(2, "AUDIO_GETDEV"); 7885 error = audio_enter_exclusive(sc); 7886 if (error) 7887 break; 7888 error = sc->hw_if->getdev(sc->hw_hdl, (audio_device_t *)addr); 7889 audio_exit_exclusive(sc); 7890 break; 7891 7892 case AUDIO_MIXER_DEVINFO: 7893 TRACE(2, "AUDIO_MIXER_DEVINFO"); 7894 mi = (mixer_devinfo_t *)addr; 7895 7896 mi->un.v.delta = 0; /* default */ 7897 mutex_enter(sc->sc_lock); 7898 error = audio_query_devinfo(sc, mi); 7899 mutex_exit(sc->sc_lock); 7900 break; 7901 7902 case AUDIO_MIXER_READ: 7903 TRACE(2, "AUDIO_MIXER_READ"); 7904 mc = (mixer_ctrl_t *)addr; 7905 7906 error = audio_enter_exclusive(sc); 7907 if (error) 7908 break; 7909 if (device_is_active(sc->hw_dev)) 7910 error = audio_get_port(sc, mc); 7911 else if (mc->dev < 0 || mc->dev >= sc->sc_nmixer_states) 7912 error = ENXIO; 7913 else { 7914 int dev = mc->dev; 7915 memcpy(mc, &sc->sc_mixer_state[dev], 7916 sizeof(mixer_ctrl_t)); 7917 error = 0; 7918 } 7919 audio_exit_exclusive(sc); 7920 break; 7921 7922 case AUDIO_MIXER_WRITE: 7923 TRACE(2, "AUDIO_MIXER_WRITE"); 7924 error = audio_enter_exclusive(sc); 7925 if (error) 7926 break; 7927 error = audio_set_port(sc, (mixer_ctrl_t *)addr); 7928 if (error) { 7929 audio_exit_exclusive(sc); 7930 break; 7931 } 7932 7933 if (sc->hw_if->commit_settings) { 7934 error = sc->hw_if->commit_settings(sc->hw_hdl); 7935 if (error) { 7936 audio_exit_exclusive(sc); 7937 break; 7938 } 7939 } 7940 mixer_signal(sc); 7941 audio_exit_exclusive(sc); 7942 break; 7943 7944 default: 7945 if (sc->hw_if->dev_ioctl) { 7946 error = audio_enter_exclusive(sc); 7947 if (error) 7948 break; 7949 error = sc->hw_if->dev_ioctl(sc->hw_hdl, 7950 cmd, addr, flag, l); 7951 audio_exit_exclusive(sc); 7952 } else 7953 error = EINVAL; 7954 break; 7955 } 7956 TRACE(2, "(%lu,'%c',%lu) result %d", 7957 IOCPARM_LEN(cmd), (char)IOCGROUP(cmd), cmd & 0xff, error); 7958 return error; 7959 } 7960 7961 /* 7962 * Must be called with sc_lock held. 7963 */ 7964 int 7965 au_portof(struct audio_softc *sc, char *name, int class) 7966 { 7967 mixer_devinfo_t mi; 7968 7969 KASSERT(mutex_owned(sc->sc_lock)); 7970 7971 for (mi.index = 0; audio_query_devinfo(sc, &mi) == 0; mi.index++) { 7972 if (mi.mixer_class == class && strcmp(mi.label.name, name) == 0) 7973 return mi.index; 7974 } 7975 return -1; 7976 } 7977 7978 /* 7979 * Must be called with sc_lock held. 7980 */ 7981 void 7982 au_setup_ports(struct audio_softc *sc, struct au_mixer_ports *ports, 7983 mixer_devinfo_t *mi, const struct portname *tbl) 7984 { 7985 int i, j; 7986 7987 KASSERT(mutex_owned(sc->sc_lock)); 7988 7989 ports->index = mi->index; 7990 if (mi->type == AUDIO_MIXER_ENUM) { 7991 ports->isenum = true; 7992 for(i = 0; tbl[i].name; i++) 7993 for(j = 0; j < mi->un.e.num_mem; j++) 7994 if (strcmp(mi->un.e.member[j].label.name, 7995 tbl[i].name) == 0) { 7996 ports->allports |= tbl[i].mask; 7997 ports->aumask[ports->nports] = tbl[i].mask; 7998 ports->misel[ports->nports] = 7999 mi->un.e.member[j].ord; 8000 ports->miport[ports->nports] = 8001 au_portof(sc, mi->un.e.member[j].label.name, 8002 mi->mixer_class); 8003 if (ports->mixerout != -1 && 8004 ports->miport[ports->nports] != -1) 8005 ports->isdual = true; 8006 ++ports->nports; 8007 } 8008 } else if (mi->type == AUDIO_MIXER_SET) { 8009 for(i = 0; tbl[i].name; i++) 8010 for(j = 0; j < mi->un.s.num_mem; j++) 8011 if (strcmp(mi->un.s.member[j].label.name, 8012 tbl[i].name) == 0) { 8013 ports->allports |= tbl[i].mask; 8014 ports->aumask[ports->nports] = tbl[i].mask; 8015 ports->misel[ports->nports] = 8016 mi->un.s.member[j].mask; 8017 ports->miport[ports->nports] = 8018 au_portof(sc, mi->un.s.member[j].label.name, 8019 mi->mixer_class); 8020 ++ports->nports; 8021 } 8022 } 8023 } 8024 8025 /* 8026 * Must be called with sc_lock && sc_exlock held. 8027 */ 8028 int 8029 au_set_lr_value(struct audio_softc *sc, mixer_ctrl_t *ct, int l, int r) 8030 { 8031 8032 KASSERT(mutex_owned(sc->sc_lock)); 8033 KASSERT(sc->sc_exlock); 8034 8035 ct->type = AUDIO_MIXER_VALUE; 8036 ct->un.value.num_channels = 2; 8037 ct->un.value.level[AUDIO_MIXER_LEVEL_LEFT] = l; 8038 ct->un.value.level[AUDIO_MIXER_LEVEL_RIGHT] = r; 8039 if (audio_set_port(sc, ct) == 0) 8040 return 0; 8041 ct->un.value.num_channels = 1; 8042 ct->un.value.level[AUDIO_MIXER_LEVEL_MONO] = (l+r)/2; 8043 return audio_set_port(sc, ct); 8044 } 8045 8046 /* 8047 * Must be called with sc_lock && sc_exlock held. 8048 */ 8049 int 8050 au_get_lr_value(struct audio_softc *sc, mixer_ctrl_t *ct, int *l, int *r) 8051 { 8052 int error; 8053 8054 KASSERT(mutex_owned(sc->sc_lock)); 8055 KASSERT(sc->sc_exlock); 8056 8057 ct->un.value.num_channels = 2; 8058 if (audio_get_port(sc, ct) == 0) { 8059 *l = ct->un.value.level[AUDIO_MIXER_LEVEL_LEFT]; 8060 *r = ct->un.value.level[AUDIO_MIXER_LEVEL_RIGHT]; 8061 } else { 8062 ct->un.value.num_channels = 1; 8063 error = audio_get_port(sc, ct); 8064 if (error) 8065 return error; 8066 *r = *l = ct->un.value.level[AUDIO_MIXER_LEVEL_MONO]; 8067 } 8068 return 0; 8069 } 8070 8071 /* 8072 * Must be called with sc_lock && sc_exlock held. 8073 */ 8074 int 8075 au_set_gain(struct audio_softc *sc, struct au_mixer_ports *ports, 8076 int gain, int balance) 8077 { 8078 mixer_ctrl_t ct; 8079 int i, error; 8080 int l, r; 8081 u_int mask; 8082 int nset; 8083 8084 KASSERT(mutex_owned(sc->sc_lock)); 8085 KASSERT(sc->sc_exlock); 8086 8087 if (balance == AUDIO_MID_BALANCE) { 8088 l = r = gain; 8089 } else if (balance < AUDIO_MID_BALANCE) { 8090 l = gain; 8091 r = (balance * gain) / AUDIO_MID_BALANCE; 8092 } else { 8093 r = gain; 8094 l = ((AUDIO_RIGHT_BALANCE - balance) * gain) 8095 / AUDIO_MID_BALANCE; 8096 } 8097 TRACE(2, "gain=%d balance=%d, l=%d r=%d", gain, balance, l, r); 8098 8099 if (ports->index == -1) { 8100 usemaster: 8101 if (ports->master == -1) 8102 return 0; /* just ignore it silently */ 8103 ct.dev = ports->master; 8104 error = au_set_lr_value(sc, &ct, l, r); 8105 } else { 8106 ct.dev = ports->index; 8107 if (ports->isenum) { 8108 ct.type = AUDIO_MIXER_ENUM; 8109 error = audio_get_port(sc, &ct); 8110 if (error) 8111 return error; 8112 if (ports->isdual) { 8113 if (ports->cur_port == -1) 8114 ct.dev = ports->master; 8115 else 8116 ct.dev = ports->miport[ports->cur_port]; 8117 error = au_set_lr_value(sc, &ct, l, r); 8118 } else { 8119 for(i = 0; i < ports->nports; i++) 8120 if (ports->misel[i] == ct.un.ord) { 8121 ct.dev = ports->miport[i]; 8122 if (ct.dev == -1 || 8123 au_set_lr_value(sc, &ct, l, r)) 8124 goto usemaster; 8125 else 8126 break; 8127 } 8128 } 8129 } else { 8130 ct.type = AUDIO_MIXER_SET; 8131 error = audio_get_port(sc, &ct); 8132 if (error) 8133 return error; 8134 mask = ct.un.mask; 8135 nset = 0; 8136 for(i = 0; i < ports->nports; i++) { 8137 if (ports->misel[i] & mask) { 8138 ct.dev = ports->miport[i]; 8139 if (ct.dev != -1 && 8140 au_set_lr_value(sc, &ct, l, r) == 0) 8141 nset++; 8142 } 8143 } 8144 if (nset == 0) 8145 goto usemaster; 8146 } 8147 } 8148 if (!error) 8149 mixer_signal(sc); 8150 return error; 8151 } 8152 8153 /* 8154 * Must be called with sc_lock && sc_exlock held. 8155 */ 8156 void 8157 au_get_gain(struct audio_softc *sc, struct au_mixer_ports *ports, 8158 u_int *pgain, u_char *pbalance) 8159 { 8160 mixer_ctrl_t ct; 8161 int i, l, r, n; 8162 int lgain, rgain; 8163 8164 KASSERT(mutex_owned(sc->sc_lock)); 8165 KASSERT(sc->sc_exlock); 8166 8167 lgain = AUDIO_MAX_GAIN / 2; 8168 rgain = AUDIO_MAX_GAIN / 2; 8169 if (ports->index == -1) { 8170 usemaster: 8171 if (ports->master == -1) 8172 goto bad; 8173 ct.dev = ports->master; 8174 ct.type = AUDIO_MIXER_VALUE; 8175 if (au_get_lr_value(sc, &ct, &lgain, &rgain)) 8176 goto bad; 8177 } else { 8178 ct.dev = ports->index; 8179 if (ports->isenum) { 8180 ct.type = AUDIO_MIXER_ENUM; 8181 if (audio_get_port(sc, &ct)) 8182 goto bad; 8183 ct.type = AUDIO_MIXER_VALUE; 8184 if (ports->isdual) { 8185 if (ports->cur_port == -1) 8186 ct.dev = ports->master; 8187 else 8188 ct.dev = ports->miport[ports->cur_port]; 8189 au_get_lr_value(sc, &ct, &lgain, &rgain); 8190 } else { 8191 for(i = 0; i < ports->nports; i++) 8192 if (ports->misel[i] == ct.un.ord) { 8193 ct.dev = ports->miport[i]; 8194 if (ct.dev == -1 || 8195 au_get_lr_value(sc, &ct, 8196 &lgain, &rgain)) 8197 goto usemaster; 8198 else 8199 break; 8200 } 8201 } 8202 } else { 8203 ct.type = AUDIO_MIXER_SET; 8204 if (audio_get_port(sc, &ct)) 8205 goto bad; 8206 ct.type = AUDIO_MIXER_VALUE; 8207 lgain = rgain = n = 0; 8208 for(i = 0; i < ports->nports; i++) { 8209 if (ports->misel[i] & ct.un.mask) { 8210 ct.dev = ports->miport[i]; 8211 if (ct.dev == -1 || 8212 au_get_lr_value(sc, &ct, &l, &r)) 8213 goto usemaster; 8214 else { 8215 lgain += l; 8216 rgain += r; 8217 n++; 8218 } 8219 } 8220 } 8221 if (n != 0) { 8222 lgain /= n; 8223 rgain /= n; 8224 } 8225 } 8226 } 8227 bad: 8228 if (lgain == rgain) { /* handles lgain==rgain==0 */ 8229 *pgain = lgain; 8230 *pbalance = AUDIO_MID_BALANCE; 8231 } else if (lgain < rgain) { 8232 *pgain = rgain; 8233 /* balance should be > AUDIO_MID_BALANCE */ 8234 *pbalance = AUDIO_RIGHT_BALANCE - 8235 (AUDIO_MID_BALANCE * lgain) / rgain; 8236 } else /* lgain > rgain */ { 8237 *pgain = lgain; 8238 /* balance should be < AUDIO_MID_BALANCE */ 8239 *pbalance = (AUDIO_MID_BALANCE * rgain) / lgain; 8240 } 8241 } 8242 8243 /* 8244 * Must be called with sc_lock && sc_exlock held. 8245 */ 8246 int 8247 au_set_port(struct audio_softc *sc, struct au_mixer_ports *ports, u_int port) 8248 { 8249 mixer_ctrl_t ct; 8250 int i, error, use_mixerout; 8251 8252 KASSERT(mutex_owned(sc->sc_lock)); 8253 KASSERT(sc->sc_exlock); 8254 8255 use_mixerout = 1; 8256 if (port == 0) { 8257 if (ports->allports == 0) 8258 return 0; /* Allow this special case. */ 8259 else if (ports->isdual) { 8260 if (ports->cur_port == -1) { 8261 return 0; 8262 } else { 8263 port = ports->aumask[ports->cur_port]; 8264 ports->cur_port = -1; 8265 use_mixerout = 0; 8266 } 8267 } 8268 } 8269 if (ports->index == -1) 8270 return EINVAL; 8271 ct.dev = ports->index; 8272 if (ports->isenum) { 8273 if (port & (port-1)) 8274 return EINVAL; /* Only one port allowed */ 8275 ct.type = AUDIO_MIXER_ENUM; 8276 error = EINVAL; 8277 for(i = 0; i < ports->nports; i++) 8278 if (ports->aumask[i] == port) { 8279 if (ports->isdual && use_mixerout) { 8280 ct.un.ord = ports->mixerout; 8281 ports->cur_port = i; 8282 } else { 8283 ct.un.ord = ports->misel[i]; 8284 } 8285 error = audio_set_port(sc, &ct); 8286 break; 8287 } 8288 } else { 8289 ct.type = AUDIO_MIXER_SET; 8290 ct.un.mask = 0; 8291 for(i = 0; i < ports->nports; i++) 8292 if (ports->aumask[i] & port) 8293 ct.un.mask |= ports->misel[i]; 8294 if (port != 0 && ct.un.mask == 0) 8295 error = EINVAL; 8296 else 8297 error = audio_set_port(sc, &ct); 8298 } 8299 if (!error) 8300 mixer_signal(sc); 8301 return error; 8302 } 8303 8304 /* 8305 * Must be called with sc_lock && sc_exlock held. 8306 */ 8307 int 8308 au_get_port(struct audio_softc *sc, struct au_mixer_ports *ports) 8309 { 8310 mixer_ctrl_t ct; 8311 int i, aumask; 8312 8313 KASSERT(mutex_owned(sc->sc_lock)); 8314 KASSERT(sc->sc_exlock); 8315 8316 if (ports->index == -1) 8317 return 0; 8318 ct.dev = ports->index; 8319 ct.type = ports->isenum ? AUDIO_MIXER_ENUM : AUDIO_MIXER_SET; 8320 if (audio_get_port(sc, &ct)) 8321 return 0; 8322 aumask = 0; 8323 if (ports->isenum) { 8324 if (ports->isdual && ports->cur_port != -1) { 8325 if (ports->mixerout == ct.un.ord) 8326 aumask = ports->aumask[ports->cur_port]; 8327 else 8328 ports->cur_port = -1; 8329 } 8330 if (aumask == 0) 8331 for(i = 0; i < ports->nports; i++) 8332 if (ports->misel[i] == ct.un.ord) 8333 aumask = ports->aumask[i]; 8334 } else { 8335 for(i = 0; i < ports->nports; i++) 8336 if (ct.un.mask & ports->misel[i]) 8337 aumask |= ports->aumask[i]; 8338 } 8339 return aumask; 8340 } 8341 8342 /* 8343 * It returns 0 if success, otherwise errno. 8344 * Must be called only if sc->sc_monitor_port != -1. 8345 * Must be called with sc_lock && sc_exlock held. 8346 */ 8347 static int 8348 au_set_monitor_gain(struct audio_softc *sc, int monitor_gain) 8349 { 8350 mixer_ctrl_t ct; 8351 8352 KASSERT(mutex_owned(sc->sc_lock)); 8353 KASSERT(sc->sc_exlock); 8354 8355 ct.dev = sc->sc_monitor_port; 8356 ct.type = AUDIO_MIXER_VALUE; 8357 ct.un.value.num_channels = 1; 8358 ct.un.value.level[AUDIO_MIXER_LEVEL_MONO] = monitor_gain; 8359 return audio_set_port(sc, &ct); 8360 } 8361 8362 /* 8363 * It returns monitor gain if success, otherwise -1. 8364 * Must be called only if sc->sc_monitor_port != -1. 8365 * Must be called with sc_lock && sc_exlock held. 8366 */ 8367 static int 8368 au_get_monitor_gain(struct audio_softc *sc) 8369 { 8370 mixer_ctrl_t ct; 8371 8372 KASSERT(mutex_owned(sc->sc_lock)); 8373 KASSERT(sc->sc_exlock); 8374 8375 ct.dev = sc->sc_monitor_port; 8376 ct.type = AUDIO_MIXER_VALUE; 8377 ct.un.value.num_channels = 1; 8378 if (audio_get_port(sc, &ct)) 8379 return -1; 8380 return ct.un.value.level[AUDIO_MIXER_LEVEL_MONO]; 8381 } 8382 8383 /* 8384 * Must be called with sc_lock && sc_exlock held. 8385 */ 8386 static int 8387 audio_set_port(struct audio_softc *sc, mixer_ctrl_t *mc) 8388 { 8389 8390 KASSERT(mutex_owned(sc->sc_lock)); 8391 KASSERT(sc->sc_exlock); 8392 8393 return sc->hw_if->set_port(sc->hw_hdl, mc); 8394 } 8395 8396 /* 8397 * Must be called with sc_lock && sc_exlock held. 8398 */ 8399 static int 8400 audio_get_port(struct audio_softc *sc, mixer_ctrl_t *mc) 8401 { 8402 8403 KASSERT(mutex_owned(sc->sc_lock)); 8404 KASSERT(sc->sc_exlock); 8405 8406 return sc->hw_if->get_port(sc->hw_hdl, mc); 8407 } 8408 8409 /* 8410 * Must be called with sc_lock && sc_exlock held. 8411 */ 8412 static void 8413 audio_mixer_capture(struct audio_softc *sc) 8414 { 8415 mixer_devinfo_t mi; 8416 mixer_ctrl_t *mc; 8417 8418 KASSERT(mutex_owned(sc->sc_lock)); 8419 KASSERT(sc->sc_exlock); 8420 8421 for (mi.index = 0;; mi.index++) { 8422 if (audio_query_devinfo(sc, &mi) != 0) 8423 break; 8424 KASSERT(mi.index < sc->sc_nmixer_states); 8425 if (mi.type == AUDIO_MIXER_CLASS) 8426 continue; 8427 mc = &sc->sc_mixer_state[mi.index]; 8428 mc->dev = mi.index; 8429 mc->type = mi.type; 8430 mc->un.value.num_channels = mi.un.v.num_channels; 8431 (void)audio_get_port(sc, mc); 8432 } 8433 8434 return; 8435 } 8436 8437 /* 8438 * Must be called with sc_lock && sc_exlock held. 8439 */ 8440 static void 8441 audio_mixer_restore(struct audio_softc *sc) 8442 { 8443 mixer_devinfo_t mi; 8444 mixer_ctrl_t *mc; 8445 8446 KASSERT(mutex_owned(sc->sc_lock)); 8447 KASSERT(sc->sc_exlock); 8448 8449 for (mi.index = 0; ; mi.index++) { 8450 if (audio_query_devinfo(sc, &mi) != 0) 8451 break; 8452 if (mi.type == AUDIO_MIXER_CLASS) 8453 continue; 8454 mc = &sc->sc_mixer_state[mi.index]; 8455 (void)audio_set_port(sc, mc); 8456 } 8457 if (sc->hw_if->commit_settings) 8458 sc->hw_if->commit_settings(sc->hw_hdl); 8459 8460 return; 8461 } 8462 8463 static void 8464 audio_volume_down(device_t dv) 8465 { 8466 struct audio_softc *sc = device_private(dv); 8467 mixer_devinfo_t mi; 8468 int newgain; 8469 u_int gain; 8470 u_char balance; 8471 8472 if (audio_enter_exclusive(sc) != 0) 8473 return; 8474 if (sc->sc_outports.index == -1 && sc->sc_outports.master != -1) { 8475 mi.index = sc->sc_outports.master; 8476 mi.un.v.delta = 0; 8477 if (audio_query_devinfo(sc, &mi) == 0) { 8478 au_get_gain(sc, &sc->sc_outports, &gain, &balance); 8479 newgain = gain - mi.un.v.delta; 8480 if (newgain < AUDIO_MIN_GAIN) 8481 newgain = AUDIO_MIN_GAIN; 8482 au_set_gain(sc, &sc->sc_outports, newgain, balance); 8483 } 8484 } 8485 audio_exit_exclusive(sc); 8486 } 8487 8488 static void 8489 audio_volume_up(device_t dv) 8490 { 8491 struct audio_softc *sc = device_private(dv); 8492 mixer_devinfo_t mi; 8493 u_int gain, newgain; 8494 u_char balance; 8495 8496 if (audio_enter_exclusive(sc) != 0) 8497 return; 8498 if (sc->sc_outports.index == -1 && sc->sc_outports.master != -1) { 8499 mi.index = sc->sc_outports.master; 8500 mi.un.v.delta = 0; 8501 if (audio_query_devinfo(sc, &mi) == 0) { 8502 au_get_gain(sc, &sc->sc_outports, &gain, &balance); 8503 newgain = gain + mi.un.v.delta; 8504 if (newgain > AUDIO_MAX_GAIN) 8505 newgain = AUDIO_MAX_GAIN; 8506 au_set_gain(sc, &sc->sc_outports, newgain, balance); 8507 } 8508 } 8509 audio_exit_exclusive(sc); 8510 } 8511 8512 static void 8513 audio_volume_toggle(device_t dv) 8514 { 8515 struct audio_softc *sc = device_private(dv); 8516 u_int gain, newgain; 8517 u_char balance; 8518 8519 if (audio_enter_exclusive(sc) != 0) 8520 return; 8521 au_get_gain(sc, &sc->sc_outports, &gain, &balance); 8522 if (gain != 0) { 8523 sc->sc_lastgain = gain; 8524 newgain = 0; 8525 } else 8526 newgain = sc->sc_lastgain; 8527 au_set_gain(sc, &sc->sc_outports, newgain, balance); 8528 audio_exit_exclusive(sc); 8529 } 8530 8531 static int 8532 audio_query_devinfo(struct audio_softc *sc, mixer_devinfo_t *di) 8533 { 8534 8535 KASSERT(mutex_owned(sc->sc_lock)); 8536 8537 return sc->hw_if->query_devinfo(sc->hw_hdl, di); 8538 } 8539 8540 #endif /* NAUDIO > 0 */ 8541 8542 #if NAUDIO == 0 && (NMIDI > 0 || NMIDIBUS > 0) 8543 #include <sys/param.h> 8544 #include <sys/systm.h> 8545 #include <sys/device.h> 8546 #include <sys/audioio.h> 8547 #include <dev/audio/audio_if.h> 8548 #endif 8549 8550 #if NAUDIO > 0 || (NMIDI > 0 || NMIDIBUS > 0) 8551 int 8552 audioprint(void *aux, const char *pnp) 8553 { 8554 struct audio_attach_args *arg; 8555 const char *type; 8556 8557 if (pnp != NULL) { 8558 arg = aux; 8559 switch (arg->type) { 8560 case AUDIODEV_TYPE_AUDIO: 8561 type = "audio"; 8562 break; 8563 case AUDIODEV_TYPE_MIDI: 8564 type = "midi"; 8565 break; 8566 case AUDIODEV_TYPE_OPL: 8567 type = "opl"; 8568 break; 8569 case AUDIODEV_TYPE_MPU: 8570 type = "mpu"; 8571 break; 8572 default: 8573 panic("audioprint: unknown type %d", arg->type); 8574 } 8575 aprint_normal("%s at %s", type, pnp); 8576 } 8577 return UNCONF; 8578 } 8579 8580 #endif /* NAUDIO > 0 || (NMIDI > 0 || NMIDIBUS > 0) */ 8581 8582 #ifdef _MODULE 8583 8584 devmajor_t audio_bmajor = -1, audio_cmajor = -1; 8585 8586 #include "ioconf.c" 8587 8588 #endif 8589 8590 MODULE(MODULE_CLASS_DRIVER, audio, NULL); 8591 8592 static int 8593 audio_modcmd(modcmd_t cmd, void *arg) 8594 { 8595 int error = 0; 8596 8597 #ifdef _MODULE 8598 switch (cmd) { 8599 case MODULE_CMD_INIT: 8600 error = devsw_attach(audio_cd.cd_name, NULL, &audio_bmajor, 8601 &audio_cdevsw, &audio_cmajor); 8602 if (error) 8603 break; 8604 8605 error = config_init_component(cfdriver_ioconf_audio, 8606 cfattach_ioconf_audio, cfdata_ioconf_audio); 8607 if (error) { 8608 devsw_detach(NULL, &audio_cdevsw); 8609 } 8610 break; 8611 case MODULE_CMD_FINI: 8612 devsw_detach(NULL, &audio_cdevsw); 8613 error = config_fini_component(cfdriver_ioconf_audio, 8614 cfattach_ioconf_audio, cfdata_ioconf_audio); 8615 if (error) 8616 devsw_attach(audio_cd.cd_name, NULL, &audio_bmajor, 8617 &audio_cdevsw, &audio_cmajor); 8618 break; 8619 default: 8620 error = ENOTTY; 8621 break; 8622 } 8623 #endif 8624 8625 return error; 8626 } 8627
Indexes created Thu Sep 25 16:09:42 GMT 2025