audio.c revision 1.4
1 /* $NetBSD: audio.c,v 1.4 2019/05/13 04:09:35 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 1.4 2019/05/13 04:09:35 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 * On half duplex hardware, the flags passed to 2050 * hw_if->open() is either FREAD or FWRITE. 2051 * see also arch/evbarm/mini2440/audio_mini2440.c 2052 */ 2053 if (fullduplex) { 2054 hwflags = FREAD | FWRITE; 2055 } else { 2056 /* Construct hwflags from af->mode. */ 2057 hwflags = 0; 2058 if ((af->mode & AUMODE_PLAY) != 0) 2059 hwflags |= FWRITE; 2060 if ((af->mode & AUMODE_RECORD) != 0) 2061 hwflags |= FREAD; 2062 } 2063 2064 mutex_enter(sc->sc_intr_lock); 2065 error = sc->hw_if->open(sc->hw_hdl, hwflags); 2066 mutex_exit(sc->sc_intr_lock); 2067 if (error) 2068 goto bad2; 2069 } 2070 2071 /* 2072 * Set speaker mode when a half duplex. 2073 * XXX I'm not sure this is correct. 2074 */ 2075 if (1/*XXX*/) { 2076 if (sc->hw_if->speaker_ctl) { 2077 int on; 2078 if (af->ptrack) { 2079 on = 1; 2080 } else { 2081 on = 0; 2082 } 2083 mutex_enter(sc->sc_intr_lock); 2084 error = sc->hw_if->speaker_ctl(sc->hw_hdl, on); 2085 mutex_exit(sc->sc_intr_lock); 2086 if (error) 2087 goto bad3; 2088 } 2089 } 2090 } else if (sc->sc_multiuser == false) { 2091 uid_t euid = kauth_cred_geteuid(kauth_cred_get()); 2092 if (euid != 0 && euid != kauth_cred_geteuid(sc->sc_cred)) { 2093 error = EPERM; 2094 goto bad2; 2095 } 2096 } 2097 2098 /* Call init_output if this is the first playback open. */ 2099 if (af->ptrack && sc->sc_popens == 0) { 2100 if (sc->hw_if->init_output) { 2101 hwbuf = &sc->sc_pmixer->hwbuf; 2102 mutex_enter(sc->sc_intr_lock); 2103 error = sc->hw_if->init_output(sc->hw_hdl, 2104 hwbuf->mem, 2105 hwbuf->capacity * 2106 hwbuf->fmt.channels * hwbuf->fmt.stride / NBBY); 2107 mutex_exit(sc->sc_intr_lock); 2108 if (error) 2109 goto bad3; 2110 } 2111 } 2112 /* Call init_input if this is the first recording open. */ 2113 if (af->rtrack && sc->sc_ropens == 0) { 2114 if (sc->hw_if->init_input) { 2115 hwbuf = &sc->sc_rmixer->hwbuf; 2116 mutex_enter(sc->sc_intr_lock); 2117 error = sc->hw_if->init_input(sc->hw_hdl, 2118 hwbuf->mem, 2119 hwbuf->capacity * 2120 hwbuf->fmt.channels * hwbuf->fmt.stride / NBBY); 2121 mutex_exit(sc->sc_intr_lock); 2122 if (error) 2123 goto bad3; 2124 } 2125 } 2126 2127 if (bell == NULL) { 2128 error = fd_allocfile(&fp, &fd); 2129 if (error) 2130 goto bad3; 2131 } 2132 2133 /* 2134 * Count up finally. 2135 * Don't fail from here. 2136 */ 2137 if (af->ptrack) 2138 sc->sc_popens++; 2139 if (af->rtrack) 2140 sc->sc_ropens++; 2141 mutex_enter(sc->sc_intr_lock); 2142 SLIST_INSERT_HEAD(&sc->sc_files, af, entry); 2143 mutex_exit(sc->sc_intr_lock); 2144 2145 if (bell) { 2146 bell->file = af; 2147 } else { 2148 error = fd_clone(fp, fd, flags, &audio_fileops, af); 2149 KASSERT(error == EMOVEFD); 2150 } 2151 2152 TRACEF(3, af, "done"); 2153 return error; 2154 2155 /* 2156 * Since track here is not yet linked to sc_files, 2157 * you can call track_destroy() without sc_intr_lock. 2158 */ 2159 bad3: 2160 if (sc->sc_popens + sc->sc_ropens == 0) { 2161 if (sc->hw_if->close) { 2162 mutex_enter(sc->sc_intr_lock); 2163 sc->hw_if->close(sc->hw_hdl); 2164 mutex_exit(sc->sc_intr_lock); 2165 } 2166 } 2167 bad2: 2168 if (af->rtrack) { 2169 audio_track_destroy(af->rtrack); 2170 af->rtrack = NULL; 2171 } 2172 if (af->ptrack) { 2173 audio_track_destroy(af->ptrack); 2174 af->ptrack = NULL; 2175 } 2176 bad1: 2177 kmem_free(af, sizeof(*af)); 2178 return error; 2179 } 2180 2181 int 2182 audio_close(struct audio_softc *sc, audio_file_t *file) 2183 { 2184 audio_track_t *oldtrack; 2185 int error; 2186 2187 KASSERT(!mutex_owned(sc->sc_lock)); 2188 KASSERT(file->lock); 2189 2190 TRACEF(1, file, "%spid=%d.%d po=%d ro=%d", 2191 (audiodebug >= 3) ? "start " : "", 2192 (int)curproc->p_pid, (int)curlwp->l_lid, 2193 sc->sc_popens, sc->sc_ropens); 2194 KASSERTMSG(sc->sc_popens + sc->sc_ropens > 0, 2195 "sc->sc_popens=%d, sc->sc_ropens=%d", 2196 sc->sc_popens, sc->sc_ropens); 2197 2198 /* 2199 * Drain first. 2200 * It must be done before acquiring exclusive lock. 2201 */ 2202 if (file->ptrack) { 2203 mutex_enter(sc->sc_lock); 2204 audio_track_drain(sc, file->ptrack); 2205 mutex_exit(sc->sc_lock); 2206 } 2207 2208 /* Then, acquire exclusive lock to protect counters. */ 2209 /* XXX what should I do when an error occurs? */ 2210 error = audio_enter_exclusive(sc); 2211 if (error) { 2212 audio_file_release(sc, file); 2213 return error; 2214 } 2215 2216 if (file->ptrack) { 2217 /* Call hw halt_output if this is the last playback track. */ 2218 if (sc->sc_popens == 1 && sc->sc_pbusy) { 2219 error = audio_pmixer_halt(sc); 2220 if (error) { 2221 device_printf(sc->sc_dev, 2222 "halt_output failed with %d\n", error); 2223 } 2224 } 2225 2226 /* Destroy the track. */ 2227 oldtrack = file->ptrack; 2228 mutex_enter(sc->sc_intr_lock); 2229 file->ptrack = NULL; 2230 mutex_exit(sc->sc_intr_lock); 2231 TRACET(3, oldtrack, "dropframes=%" PRIu64, 2232 oldtrack->dropframes); 2233 audio_track_destroy(oldtrack); 2234 2235 KASSERT(sc->sc_popens > 0); 2236 sc->sc_popens--; 2237 } 2238 if (file->rtrack) { 2239 /* Call hw halt_input if this is the last recording track. */ 2240 if (sc->sc_ropens == 1 && sc->sc_rbusy) { 2241 error = audio_rmixer_halt(sc); 2242 if (error) { 2243 device_printf(sc->sc_dev, 2244 "halt_input failed with %d\n", error); 2245 } 2246 } 2247 2248 /* Destroy the track. */ 2249 oldtrack = file->rtrack; 2250 mutex_enter(sc->sc_intr_lock); 2251 file->rtrack = NULL; 2252 mutex_exit(sc->sc_intr_lock); 2253 TRACET(3, oldtrack, "dropframes=%" PRIu64, 2254 oldtrack->dropframes); 2255 audio_track_destroy(oldtrack); 2256 2257 KASSERT(sc->sc_ropens > 0); 2258 sc->sc_ropens--; 2259 } 2260 2261 /* Call hw close if this is the last track. */ 2262 if (sc->sc_popens + sc->sc_ropens == 0) { 2263 if (sc->hw_if->close) { 2264 TRACE(2, "hw_if close"); 2265 mutex_enter(sc->sc_intr_lock); 2266 sc->hw_if->close(sc->hw_hdl); 2267 mutex_exit(sc->sc_intr_lock); 2268 } 2269 2270 kauth_cred_free(sc->sc_cred); 2271 } 2272 2273 mutex_enter(sc->sc_intr_lock); 2274 SLIST_REMOVE(&sc->sc_files, file, audio_file, entry); 2275 mutex_exit(sc->sc_intr_lock); 2276 2277 TRACE(3, "done"); 2278 audio_exit_exclusive(sc); 2279 return 0; 2280 } 2281 2282 int 2283 audio_read(struct audio_softc *sc, struct uio *uio, int ioflag, 2284 audio_file_t *file) 2285 { 2286 audio_track_t *track; 2287 audio_ring_t *usrbuf; 2288 audio_ring_t *input; 2289 int error; 2290 2291 track = file->rtrack; 2292 KASSERT(track); 2293 TRACET(2, track, "resid=%zd", uio->uio_resid); 2294 2295 KASSERT(!mutex_owned(sc->sc_lock)); 2296 KASSERT(file->lock); 2297 2298 /* I think it's better than EINVAL. */ 2299 if (track->mmapped) 2300 return EPERM; 2301 2302 #ifdef AUDIO_PM_IDLE 2303 mutex_enter(sc->sc_lock); 2304 if (device_is_active(&sc->sc_dev) || sc->sc_idle) 2305 device_active(&sc->sc_dev, DVA_SYSTEM); 2306 mutex_exit(sc->sc_lock); 2307 #endif 2308 2309 /* 2310 * On half-duplex hardware, O_RDWR is treated as O_WRONLY. 2311 * However read() system call itself can be called because it's 2312 * opened with O_RDWR. So in this case, deny this read(). 2313 */ 2314 if ((file->mode & AUMODE_RECORD) == 0) { 2315 return EBADF; 2316 } 2317 2318 TRACET(3, track, "resid=%zd", uio->uio_resid); 2319 2320 usrbuf = &track->usrbuf; 2321 input = track->input; 2322 2323 /* 2324 * The first read starts rmixer. 2325 */ 2326 error = audio_enter_exclusive(sc); 2327 if (error) 2328 return error; 2329 if (sc->sc_rbusy == false) 2330 audio_rmixer_start(sc); 2331 audio_exit_exclusive(sc); 2332 2333 error = 0; 2334 while (uio->uio_resid > 0 && error == 0) { 2335 int bytes; 2336 2337 TRACET(3, track, 2338 "while resid=%zd input=%d/%d/%d usrbuf=%d/%d/H%d", 2339 uio->uio_resid, 2340 input->head, input->used, input->capacity, 2341 usrbuf->head, usrbuf->used, track->usrbuf_usedhigh); 2342 2343 /* Wait when buffers are empty. */ 2344 mutex_enter(sc->sc_lock); 2345 for (;;) { 2346 bool empty; 2347 audio_track_lock_enter(track); 2348 empty = (input->used == 0 && usrbuf->used == 0); 2349 audio_track_lock_exit(track); 2350 if (!empty) 2351 break; 2352 2353 if ((ioflag & IO_NDELAY)) { 2354 mutex_exit(sc->sc_lock); 2355 return EWOULDBLOCK; 2356 } 2357 2358 TRACET(3, track, "sleep"); 2359 error = audio_track_waitio(sc, track); 2360 if (error) { 2361 mutex_exit(sc->sc_lock); 2362 return error; 2363 } 2364 } 2365 mutex_exit(sc->sc_lock); 2366 2367 audio_track_lock_enter(track); 2368 audio_track_record(track); 2369 audio_track_lock_exit(track); 2370 2371 /* uiomove from usrbuf as much as possible. */ 2372 bytes = uimin(usrbuf->used, uio->uio_resid); 2373 while (bytes > 0) { 2374 int head = usrbuf->head; 2375 int len = uimin(bytes, usrbuf->capacity - head); 2376 error = uiomove((uint8_t *)usrbuf->mem + head, len, 2377 uio); 2378 if (error) { 2379 device_printf(sc->sc_dev, 2380 "uiomove(len=%d) failed with %d\n", 2381 len, error); 2382 goto abort; 2383 } 2384 auring_take(usrbuf, len); 2385 track->useriobytes += len; 2386 TRACET(3, track, "uiomove(len=%d) usrbuf=%d/%d/C%d", 2387 len, 2388 usrbuf->head, usrbuf->used, usrbuf->capacity); 2389 bytes -= len; 2390 } 2391 } 2392 2393 abort: 2394 return error; 2395 } 2396 2397 2398 /* 2399 * Clear file's playback and/or record track buffer immediately. 2400 */ 2401 static void 2402 audio_file_clear(struct audio_softc *sc, audio_file_t *file) 2403 { 2404 2405 if (file->ptrack) 2406 audio_track_clear(sc, file->ptrack); 2407 if (file->rtrack) 2408 audio_track_clear(sc, file->rtrack); 2409 } 2410 2411 int 2412 audio_write(struct audio_softc *sc, struct uio *uio, int ioflag, 2413 audio_file_t *file) 2414 { 2415 audio_track_t *track; 2416 audio_ring_t *usrbuf; 2417 audio_ring_t *outbuf; 2418 int error; 2419 2420 track = file->ptrack; 2421 KASSERT(track); 2422 TRACET(2, track, "%sresid=%zd pid=%d.%d ioflag=0x%x", 2423 audiodebug >= 3 ? "begin " : "", 2424 uio->uio_resid, (int)curproc->p_pid, (int)curlwp->l_lid, ioflag); 2425 2426 KASSERT(!mutex_owned(sc->sc_lock)); 2427 KASSERT(file->lock); 2428 2429 /* I think it's better than EINVAL. */ 2430 if (track->mmapped) 2431 return EPERM; 2432 2433 if (uio->uio_resid == 0) { 2434 track->eofcounter++; 2435 return 0; 2436 } 2437 2438 #ifdef AUDIO_PM_IDLE 2439 mutex_enter(sc->sc_lock); 2440 if (device_is_active(&sc->sc_dev) || sc->sc_idle) 2441 device_active(&sc->sc_dev, DVA_SYSTEM); 2442 mutex_exit(sc->sc_lock); 2443 #endif 2444 2445 usrbuf = &track->usrbuf; 2446 outbuf = &track->outbuf; 2447 2448 /* 2449 * The first write starts pmixer. 2450 */ 2451 error = audio_enter_exclusive(sc); 2452 if (error) 2453 return error; 2454 if (sc->sc_pbusy == false) 2455 audio_pmixer_start(sc, false); 2456 audio_exit_exclusive(sc); 2457 2458 track->pstate = AUDIO_STATE_RUNNING; 2459 error = 0; 2460 while (uio->uio_resid > 0 && error == 0) { 2461 int bytes; 2462 2463 TRACET(3, track, "while resid=%zd usrbuf=%d/%d/H%d", 2464 uio->uio_resid, 2465 usrbuf->head, usrbuf->used, track->usrbuf_usedhigh); 2466 2467 /* Wait when buffers are full. */ 2468 mutex_enter(sc->sc_lock); 2469 for (;;) { 2470 bool full; 2471 audio_track_lock_enter(track); 2472 full = (usrbuf->used >= track->usrbuf_usedhigh && 2473 outbuf->used >= outbuf->capacity); 2474 audio_track_lock_exit(track); 2475 if (!full) 2476 break; 2477 2478 if ((ioflag & IO_NDELAY)) { 2479 error = EWOULDBLOCK; 2480 mutex_exit(sc->sc_lock); 2481 goto abort; 2482 } 2483 2484 TRACET(3, track, "sleep usrbuf=%d/H%d", 2485 usrbuf->used, track->usrbuf_usedhigh); 2486 error = audio_track_waitio(sc, track); 2487 if (error) { 2488 mutex_exit(sc->sc_lock); 2489 goto abort; 2490 } 2491 } 2492 mutex_exit(sc->sc_lock); 2493 2494 /* uiomove to usrbuf as much as possible. */ 2495 bytes = uimin(track->usrbuf_usedhigh - usrbuf->used, 2496 uio->uio_resid); 2497 while (bytes > 0) { 2498 int tail = auring_tail(usrbuf); 2499 int len = uimin(bytes, usrbuf->capacity - tail); 2500 error = uiomove((uint8_t *)usrbuf->mem + tail, len, 2501 uio); 2502 if (error) { 2503 device_printf(sc->sc_dev, 2504 "uiomove(len=%d) failed with %d\n", 2505 len, error); 2506 goto abort; 2507 } 2508 auring_push(usrbuf, len); 2509 track->useriobytes += len; 2510 TRACET(3, track, "uiomove(len=%d) usrbuf=%d/%d/C%d", 2511 len, 2512 usrbuf->head, usrbuf->used, usrbuf->capacity); 2513 bytes -= len; 2514 } 2515 2516 /* Convert them as much as possible. */ 2517 audio_track_lock_enter(track); 2518 while (usrbuf->used >= track->usrbuf_blksize && 2519 outbuf->used < outbuf->capacity) { 2520 audio_track_play(track); 2521 } 2522 audio_track_lock_exit(track); 2523 } 2524 2525 abort: 2526 TRACET(3, track, "done error=%d", error); 2527 return error; 2528 } 2529 2530 int 2531 audio_ioctl(dev_t dev, struct audio_softc *sc, u_long cmd, void *addr, int flag, 2532 struct lwp *l, audio_file_t *file) 2533 { 2534 struct audio_offset *ao; 2535 struct audio_info ai; 2536 audio_track_t *track; 2537 audio_encoding_t *ae; 2538 audio_format_query_t *query; 2539 u_int stamp; 2540 u_int offs; 2541 int fd; 2542 int index; 2543 int error; 2544 2545 KASSERT(!mutex_owned(sc->sc_lock)); 2546 KASSERT(file->lock); 2547 2548 #if defined(AUDIO_DEBUG) 2549 const char *ioctlnames[] = { 2550 " AUDIO_GETINFO", /* 21 */ 2551 " AUDIO_SETINFO", /* 22 */ 2552 " AUDIO_DRAIN", /* 23 */ 2553 " AUDIO_FLUSH", /* 24 */ 2554 " AUDIO_WSEEK", /* 25 */ 2555 " AUDIO_RERROR", /* 26 */ 2556 " AUDIO_GETDEV", /* 27 */ 2557 " AUDIO_GETENC", /* 28 */ 2558 " AUDIO_GETFD", /* 29 */ 2559 " AUDIO_SETFD", /* 30 */ 2560 " AUDIO_PERROR", /* 31 */ 2561 " AUDIO_GETIOFFS", /* 32 */ 2562 " AUDIO_GETOOFFS", /* 33 */ 2563 " AUDIO_GETPROPS", /* 34 */ 2564 " AUDIO_GETBUFINFO", /* 35 */ 2565 " AUDIO_SETCHAN", /* 36 */ 2566 " AUDIO_GETCHAN", /* 37 */ 2567 " AUDIO_QUERYFORMAT", /* 38 */ 2568 " AUDIO_GETFORMAT", /* 39 */ 2569 " AUDIO_SETFORMAT", /* 40 */ 2570 }; 2571 int nameidx = (cmd & 0xff); 2572 const char *ioctlname = ""; 2573 if (21 <= nameidx && nameidx <= 21 + __arraycount(ioctlnames)) 2574 ioctlname = ioctlnames[nameidx - 21]; 2575 TRACEF(2, file, "(%lu,'%c',%lu)%s pid=%d.%d", 2576 IOCPARM_LEN(cmd), (char)IOCGROUP(cmd), cmd&0xff, ioctlname, 2577 (int)curproc->p_pid, (int)l->l_lid); 2578 #endif 2579 2580 error = 0; 2581 switch (cmd) { 2582 case FIONBIO: 2583 /* All handled in the upper FS layer. */ 2584 break; 2585 2586 case FIONREAD: 2587 /* Get the number of bytes that can be read. */ 2588 if (file->rtrack) { 2589 *(int *)addr = audio_track_readablebytes(file->rtrack); 2590 } else { 2591 *(int *)addr = 0; 2592 } 2593 break; 2594 2595 case FIOASYNC: 2596 /* Set/Clear ASYNC I/O. */ 2597 if (*(int *)addr) { 2598 file->async_audio = curproc->p_pid; 2599 TRACEF(2, file, "FIOASYNC pid %d", file->async_audio); 2600 } else { 2601 file->async_audio = 0; 2602 TRACEF(2, file, "FIOASYNC off"); 2603 } 2604 break; 2605 2606 case AUDIO_FLUSH: 2607 /* XXX TODO: clear errors and restart? */ 2608 audio_file_clear(sc, file); 2609 break; 2610 2611 case AUDIO_RERROR: 2612 /* 2613 * Number of read bytes dropped. We don't know where 2614 * or when they were dropped (including conversion stage). 2615 * Therefore, the number of accurate bytes or samples is 2616 * also unknown. 2617 */ 2618 track = file->rtrack; 2619 if (track) { 2620 *(int *)addr = frametobyte(&track->usrbuf.fmt, 2621 track->dropframes); 2622 } 2623 break; 2624 2625 case AUDIO_PERROR: 2626 /* 2627 * Number of write bytes dropped. We don't know where 2628 * or when they were dropped (including conversion stage). 2629 * Therefore, the number of accurate bytes or samples is 2630 * also unknown. 2631 */ 2632 track = file->ptrack; 2633 if (track) { 2634 *(int *)addr = frametobyte(&track->usrbuf.fmt, 2635 track->dropframes); 2636 } 2637 break; 2638 2639 case AUDIO_GETIOFFS: 2640 /* XXX TODO */ 2641 ao = (struct audio_offset *)addr; 2642 ao->samples = 0; 2643 ao->deltablks = 0; 2644 ao->offset = 0; 2645 break; 2646 2647 case AUDIO_GETOOFFS: 2648 ao = (struct audio_offset *)addr; 2649 track = file->ptrack; 2650 if (track == NULL) { 2651 ao->samples = 0; 2652 ao->deltablks = 0; 2653 ao->offset = 0; 2654 break; 2655 } 2656 mutex_enter(sc->sc_lock); 2657 mutex_enter(sc->sc_intr_lock); 2658 /* figure out where next DMA will start */ 2659 stamp = track->usrbuf_stamp; 2660 offs = track->usrbuf.head; 2661 mutex_exit(sc->sc_intr_lock); 2662 mutex_exit(sc->sc_lock); 2663 2664 ao->samples = stamp; 2665 ao->deltablks = (stamp / track->usrbuf_blksize) - 2666 (track->usrbuf_stamp_last / track->usrbuf_blksize); 2667 track->usrbuf_stamp_last = stamp; 2668 offs = rounddown(offs, track->usrbuf_blksize) 2669 + track->usrbuf_blksize; 2670 if (offs >= track->usrbuf.capacity) 2671 offs -= track->usrbuf.capacity; 2672 ao->offset = offs; 2673 2674 TRACET(3, track, "GETOOFFS: samples=%u deltablks=%u offset=%u", 2675 ao->samples, ao->deltablks, ao->offset); 2676 break; 2677 2678 case AUDIO_WSEEK: 2679 /* XXX return value does not include outbuf one. */ 2680 if (file->ptrack) 2681 *(u_long *)addr = file->ptrack->usrbuf.used; 2682 break; 2683 2684 case AUDIO_SETINFO: 2685 error = audio_enter_exclusive(sc); 2686 if (error) 2687 break; 2688 error = audio_file_setinfo(sc, file, (struct audio_info *)addr); 2689 if (error) { 2690 audio_exit_exclusive(sc); 2691 break; 2692 } 2693 /* XXX TODO: update last_ai if /dev/sound ? */ 2694 if (ISDEVSOUND(dev)) 2695 error = audiogetinfo(sc, &sc->sc_ai, 0, file); 2696 audio_exit_exclusive(sc); 2697 break; 2698 2699 case AUDIO_GETINFO: 2700 error = audio_enter_exclusive(sc); 2701 if (error) 2702 break; 2703 error = audiogetinfo(sc, (struct audio_info *)addr, 1, file); 2704 audio_exit_exclusive(sc); 2705 break; 2706 2707 case AUDIO_GETBUFINFO: 2708 mutex_enter(sc->sc_lock); 2709 error = audiogetinfo(sc, (struct audio_info *)addr, 0, file); 2710 mutex_exit(sc->sc_lock); 2711 break; 2712 2713 case AUDIO_DRAIN: 2714 if (file->ptrack) { 2715 mutex_enter(sc->sc_lock); 2716 error = audio_track_drain(sc, file->ptrack); 2717 mutex_exit(sc->sc_lock); 2718 } 2719 break; 2720 2721 case AUDIO_GETDEV: 2722 mutex_enter(sc->sc_lock); 2723 error = sc->hw_if->getdev(sc->hw_hdl, (audio_device_t *)addr); 2724 mutex_exit(sc->sc_lock); 2725 break; 2726 2727 case AUDIO_GETENC: 2728 ae = (audio_encoding_t *)addr; 2729 index = ae->index; 2730 if (index < 0 || index >= __arraycount(audio_encodings)) { 2731 error = EINVAL; 2732 break; 2733 } 2734 *ae = audio_encodings[index]; 2735 ae->index = index; 2736 /* 2737 * EMULATED always. 2738 * EMULATED flag at that time used to mean that it could 2739 * not be passed directly to the hardware as-is. But 2740 * currently, all formats including hardware native is not 2741 * passed directly to the hardware. So I set EMULATED 2742 * flag for all formats. 2743 */ 2744 ae->flags = AUDIO_ENCODINGFLAG_EMULATED; 2745 break; 2746 2747 case AUDIO_GETFD: 2748 /* 2749 * Returns the current setting of full duplex mode. 2750 * If HW has full duplex mode and there are two mixers, 2751 * it is full duplex. Otherwise half duplex. 2752 */ 2753 mutex_enter(sc->sc_lock); 2754 fd = (audio_get_props(sc) & AUDIO_PROP_FULLDUPLEX) 2755 && (sc->sc_pmixer && sc->sc_rmixer); 2756 mutex_exit(sc->sc_lock); 2757 *(int *)addr = fd; 2758 break; 2759 2760 case AUDIO_GETPROPS: 2761 mutex_enter(sc->sc_lock); 2762 *(int *)addr = audio_get_props(sc); 2763 mutex_exit(sc->sc_lock); 2764 break; 2765 2766 case AUDIO_QUERYFORMAT: 2767 query = (audio_format_query_t *)addr; 2768 if (sc->hw_if->query_format) { 2769 mutex_enter(sc->sc_lock); 2770 error = sc->hw_if->query_format(sc->hw_hdl, query); 2771 mutex_exit(sc->sc_lock); 2772 /* Hide internal infomations */ 2773 query->fmt.driver_data = NULL; 2774 } else { 2775 error = ENODEV; 2776 } 2777 break; 2778 2779 case AUDIO_GETFORMAT: 2780 audio_mixers_get_format(sc, (struct audio_info *)addr); 2781 break; 2782 2783 case AUDIO_SETFORMAT: 2784 mutex_enter(sc->sc_lock); 2785 audio_mixers_get_format(sc, &ai); 2786 error = audio_mixers_set_format(sc, (struct audio_info *)addr); 2787 if (error) { 2788 /* Rollback */ 2789 audio_mixers_set_format(sc, &ai); 2790 } 2791 mutex_exit(sc->sc_lock); 2792 break; 2793 2794 case AUDIO_SETFD: 2795 case AUDIO_SETCHAN: 2796 case AUDIO_GETCHAN: 2797 /* Obsoleted */ 2798 break; 2799 2800 default: 2801 if (sc->hw_if->dev_ioctl) { 2802 error = audio_enter_exclusive(sc); 2803 if (error) 2804 break; 2805 error = sc->hw_if->dev_ioctl(sc->hw_hdl, 2806 cmd, addr, flag, l); 2807 audio_exit_exclusive(sc); 2808 } else { 2809 TRACEF(2, file, "unknown ioctl"); 2810 error = EINVAL; 2811 } 2812 break; 2813 } 2814 TRACEF(2, file, "(%lu,'%c',%lu)%s result %d", 2815 IOCPARM_LEN(cmd), (char)IOCGROUP(cmd), cmd&0xff, ioctlname, 2816 error); 2817 return error; 2818 } 2819 2820 /* 2821 * Returns the number of bytes that can be read on recording buffer. 2822 */ 2823 static __inline int 2824 audio_track_readablebytes(const audio_track_t *track) 2825 { 2826 int bytes; 2827 2828 KASSERT(track); 2829 KASSERT(track->mode == AUMODE_RECORD); 2830 2831 /* 2832 * Although usrbuf is primarily readable data, recorded data 2833 * also stays in track->input until reading. So it is necessary 2834 * to add it. track->input is in frame, usrbuf is in byte. 2835 */ 2836 bytes = track->usrbuf.used + 2837 track->input->used * frametobyte(&track->usrbuf.fmt, 1); 2838 return bytes; 2839 } 2840 2841 int 2842 audio_poll(struct audio_softc *sc, int events, struct lwp *l, 2843 audio_file_t *file) 2844 { 2845 audio_track_t *track; 2846 int revents; 2847 bool in_is_valid; 2848 bool out_is_valid; 2849 2850 KASSERT(!mutex_owned(sc->sc_lock)); 2851 KASSERT(file->lock); 2852 2853 #if defined(AUDIO_DEBUG) 2854 #define POLLEV_BITMAP "\177\020" \ 2855 "b\10WRBAND\0" \ 2856 "b\7RDBAND\0" "b\6RDNORM\0" "b\5NVAL\0" "b\4HUP\0" \ 2857 "b\3ERR\0" "b\2OUT\0" "b\1PRI\0" "b\0IN\0" 2858 char evbuf[64]; 2859 snprintb(evbuf, sizeof(evbuf), POLLEV_BITMAP, events); 2860 TRACEF(2, file, "pid=%d.%d events=%s", 2861 (int)curproc->p_pid, (int)l->l_lid, evbuf); 2862 #endif 2863 2864 revents = 0; 2865 in_is_valid = false; 2866 out_is_valid = false; 2867 if (events & (POLLIN | POLLRDNORM)) { 2868 track = file->rtrack; 2869 if (track) { 2870 int used; 2871 in_is_valid = true; 2872 used = audio_track_readablebytes(track); 2873 if (used > 0) 2874 revents |= events & (POLLIN | POLLRDNORM); 2875 } 2876 } 2877 if (events & (POLLOUT | POLLWRNORM)) { 2878 track = file->ptrack; 2879 if (track) { 2880 out_is_valid = true; 2881 if (track->usrbuf.used <= track->usrbuf_usedlow) 2882 revents |= events & (POLLOUT | POLLWRNORM); 2883 } 2884 } 2885 2886 if (revents == 0) { 2887 mutex_enter(sc->sc_lock); 2888 if (in_is_valid) { 2889 TRACEF(3, file, "selrecord rsel"); 2890 selrecord(l, &sc->sc_rsel); 2891 } 2892 if (out_is_valid) { 2893 TRACEF(3, file, "selrecord wsel"); 2894 selrecord(l, &sc->sc_wsel); 2895 } 2896 mutex_exit(sc->sc_lock); 2897 } 2898 2899 #if defined(AUDIO_DEBUG) 2900 snprintb(evbuf, sizeof(evbuf), POLLEV_BITMAP, revents); 2901 TRACEF(2, file, "revents=%s", evbuf); 2902 #endif 2903 return revents; 2904 } 2905 2906 static const struct filterops audioread_filtops = { 2907 .f_isfd = 1, 2908 .f_attach = NULL, 2909 .f_detach = filt_audioread_detach, 2910 .f_event = filt_audioread_event, 2911 }; 2912 2913 static void 2914 filt_audioread_detach(struct knote *kn) 2915 { 2916 struct audio_softc *sc; 2917 audio_file_t *file; 2918 2919 file = kn->kn_hook; 2920 sc = file->sc; 2921 TRACEF(3, file, ""); 2922 2923 mutex_enter(sc->sc_lock); 2924 SLIST_REMOVE(&sc->sc_rsel.sel_klist, kn, knote, kn_selnext); 2925 mutex_exit(sc->sc_lock); 2926 } 2927 2928 static int 2929 filt_audioread_event(struct knote *kn, long hint) 2930 { 2931 audio_file_t *file; 2932 audio_track_t *track; 2933 2934 file = kn->kn_hook; 2935 track = file->rtrack; 2936 2937 /* 2938 * kn_data must contain the number of bytes can be read. 2939 * The return value indicates whether the event occurs or not. 2940 */ 2941 2942 if (track == NULL) { 2943 /* can not read with this descriptor. */ 2944 kn->kn_data = 0; 2945 return 0; 2946 } 2947 2948 kn->kn_data = audio_track_readablebytes(track); 2949 TRACEF(3, file, "data=%" PRId64, kn->kn_data); 2950 return kn->kn_data > 0; 2951 } 2952 2953 static const struct filterops audiowrite_filtops = { 2954 .f_isfd = 1, 2955 .f_attach = NULL, 2956 .f_detach = filt_audiowrite_detach, 2957 .f_event = filt_audiowrite_event, 2958 }; 2959 2960 static void 2961 filt_audiowrite_detach(struct knote *kn) 2962 { 2963 struct audio_softc *sc; 2964 audio_file_t *file; 2965 2966 file = kn->kn_hook; 2967 sc = file->sc; 2968 TRACEF(3, file, ""); 2969 2970 mutex_enter(sc->sc_lock); 2971 SLIST_REMOVE(&sc->sc_wsel.sel_klist, kn, knote, kn_selnext); 2972 mutex_exit(sc->sc_lock); 2973 } 2974 2975 static int 2976 filt_audiowrite_event(struct knote *kn, long hint) 2977 { 2978 audio_file_t *file; 2979 audio_track_t *track; 2980 2981 file = kn->kn_hook; 2982 track = file->ptrack; 2983 2984 /* 2985 * kn_data must contain the number of bytes can be write. 2986 * The return value indicates whether the event occurs or not. 2987 */ 2988 2989 if (track == NULL) { 2990 /* can not write with this descriptor. */ 2991 kn->kn_data = 0; 2992 return 0; 2993 } 2994 2995 kn->kn_data = track->usrbuf_usedhigh - track->usrbuf.used; 2996 TRACEF(3, file, "data=%" PRId64, kn->kn_data); 2997 return (track->usrbuf.used < track->usrbuf_usedlow); 2998 } 2999 3000 int 3001 audio_kqfilter(struct audio_softc *sc, audio_file_t *file, struct knote *kn) 3002 { 3003 struct klist *klist; 3004 3005 KASSERT(!mutex_owned(sc->sc_lock)); 3006 KASSERT(file->lock); 3007 3008 TRACEF(3, file, "kn=%p kn_filter=%x", kn, (int)kn->kn_filter); 3009 3010 switch (kn->kn_filter) { 3011 case EVFILT_READ: 3012 klist = &sc->sc_rsel.sel_klist; 3013 kn->kn_fop = &audioread_filtops; 3014 break; 3015 3016 case EVFILT_WRITE: 3017 klist = &sc->sc_wsel.sel_klist; 3018 kn->kn_fop = &audiowrite_filtops; 3019 break; 3020 3021 default: 3022 return EINVAL; 3023 } 3024 3025 kn->kn_hook = file; 3026 3027 mutex_enter(sc->sc_lock); 3028 SLIST_INSERT_HEAD(klist, kn, kn_selnext); 3029 mutex_exit(sc->sc_lock); 3030 3031 return 0; 3032 } 3033 3034 int 3035 audio_mmap(struct audio_softc *sc, off_t *offp, size_t len, int prot, 3036 int *flagsp, int *advicep, struct uvm_object **uobjp, int *maxprotp, 3037 audio_file_t *file) 3038 { 3039 audio_track_t *track; 3040 vsize_t vsize; 3041 int error; 3042 3043 KASSERT(!mutex_owned(sc->sc_lock)); 3044 KASSERT(file->lock); 3045 3046 TRACEF(2, file, "off=%lld, prot=%d", (long long)(*offp), prot); 3047 3048 if (*offp < 0) 3049 return EINVAL; 3050 3051 #if 0 3052 /* XXX 3053 * The idea here was to use the protection to determine if 3054 * we are mapping the read or write buffer, but it fails. 3055 * The VM system is broken in (at least) two ways. 3056 * 1) If you map memory VM_PROT_WRITE you SIGSEGV 3057 * when writing to it, so VM_PROT_READ|VM_PROT_WRITE 3058 * has to be used for mmapping the play buffer. 3059 * 2) Even if calling mmap() with VM_PROT_READ|VM_PROT_WRITE 3060 * audio_mmap will get called at some point with VM_PROT_READ 3061 * only. 3062 * So, alas, we always map the play buffer for now. 3063 */ 3064 if (prot == (VM_PROT_READ|VM_PROT_WRITE) || 3065 prot == VM_PROT_WRITE) 3066 track = file->ptrack; 3067 else if (prot == VM_PROT_READ) 3068 track = file->rtrack; 3069 else 3070 return EINVAL; 3071 #else 3072 track = file->ptrack; 3073 #endif 3074 if (track == NULL) 3075 return EACCES; 3076 3077 vsize = roundup2(MAX(track->usrbuf.capacity, PAGE_SIZE), PAGE_SIZE); 3078 if (len > vsize) 3079 return EOVERFLOW; 3080 if (*offp > (uint)(vsize - len)) 3081 return EOVERFLOW; 3082 3083 /* XXX TODO: what happens when mmap twice. */ 3084 if (!track->mmapped) { 3085 track->mmapped = true; 3086 3087 if (!track->is_pause) { 3088 error = audio_enter_exclusive(sc); 3089 if (error) 3090 return error; 3091 if (sc->sc_pbusy == false) 3092 audio_pmixer_start(sc, true); 3093 audio_exit_exclusive(sc); 3094 } 3095 /* XXX mmapping record buffer is not supported */ 3096 } 3097 3098 /* get ringbuffer */ 3099 *uobjp = track->uobj; 3100 3101 /* Acquire a reference for the mmap. munmap will release. */ 3102 uao_reference(*uobjp); 3103 *maxprotp = prot; 3104 *advicep = UVM_ADV_RANDOM; 3105 *flagsp = MAP_SHARED; 3106 return 0; 3107 } 3108 3109 /* 3110 * /dev/audioctl has to be able to open at any time without interference 3111 * with any /dev/audio or /dev/sound. 3112 */ 3113 static int 3114 audioctl_open(dev_t dev, struct audio_softc *sc, int flags, int ifmt, 3115 struct lwp *l) 3116 { 3117 struct file *fp; 3118 audio_file_t *af; 3119 int fd; 3120 int error; 3121 3122 KASSERT(mutex_owned(sc->sc_lock)); 3123 KASSERT(sc->sc_exlock); 3124 3125 TRACE(1, ""); 3126 3127 error = fd_allocfile(&fp, &fd); 3128 if (error) 3129 return error; 3130 3131 af = kmem_zalloc(sizeof(audio_file_t), KM_SLEEP); 3132 af->sc = sc; 3133 af->dev = dev; 3134 3135 /* Not necessary to insert sc_files. */ 3136 3137 error = fd_clone(fp, fd, flags, &audio_fileops, af); 3138 KASSERT(error == EMOVEFD); 3139 3140 return error; 3141 } 3142 3143 /* 3144 * Reallocate 'memblock' with specified 'bytes' if 'bytes' > 0. 3145 * Or free 'memblock' and return NULL if 'byte' is zero. 3146 */ 3147 static void * 3148 audio_realloc(void *memblock, size_t bytes) 3149 { 3150 3151 if (memblock != NULL) { 3152 if (bytes != 0) { 3153 return kern_realloc(memblock, bytes, M_NOWAIT); 3154 } else { 3155 kern_free(memblock); 3156 return NULL; 3157 } 3158 } else { 3159 if (bytes != 0) { 3160 return kern_malloc(bytes, M_NOWAIT); 3161 } else { 3162 return NULL; 3163 } 3164 } 3165 } 3166 3167 /* 3168 * Free 'mem' if available, and initialize the pointer. 3169 * For this reason, this is implemented as macro. 3170 */ 3171 #define audio_free(mem) do { \ 3172 if (mem != NULL) { \ 3173 kern_free(mem); \ 3174 mem = NULL; \ 3175 } \ 3176 } while (0) 3177 3178 /* 3179 * (Re)allocate usrbuf with 'newbufsize' bytes. 3180 * Use this function for usrbuf because only usrbuf can be mmapped. 3181 * If successful, it updates track->usrbuf.mem, track->usrbuf.capacity and 3182 * returns 0. Otherwise, it clears track->usrbuf.mem, track->usrbuf.capacity 3183 * and returns errno. 3184 * It must be called before updating usrbuf.capacity. 3185 */ 3186 static int 3187 audio_realloc_usrbuf(audio_track_t *track, int newbufsize) 3188 { 3189 struct audio_softc *sc; 3190 vaddr_t vstart; 3191 vsize_t oldvsize; 3192 vsize_t newvsize; 3193 int error; 3194 3195 KASSERT(newbufsize > 0); 3196 sc = track->mixer->sc; 3197 3198 /* Get a nonzero multiple of PAGE_SIZE */ 3199 newvsize = roundup2(MAX(newbufsize, PAGE_SIZE), PAGE_SIZE); 3200 3201 if (track->usrbuf.mem != NULL) { 3202 oldvsize = roundup2(MAX(track->usrbuf.capacity, PAGE_SIZE), 3203 PAGE_SIZE); 3204 if (oldvsize == newvsize) { 3205 track->usrbuf.capacity = newbufsize; 3206 return 0; 3207 } 3208 vstart = (vaddr_t)track->usrbuf.mem; 3209 uvm_unmap(kernel_map, vstart, vstart + oldvsize); 3210 /* uvm_unmap also detach uobj */ 3211 track->uobj = NULL; /* paranoia */ 3212 track->usrbuf.mem = NULL; 3213 } 3214 3215 /* Create a uvm anonymous object */ 3216 track->uobj = uao_create(newvsize, 0); 3217 3218 /* Map it into the kernel virtual address space */ 3219 vstart = 0; 3220 error = uvm_map(kernel_map, &vstart, newvsize, track->uobj, 0, 0, 3221 UVM_MAPFLAG(UVM_PROT_RW, UVM_PROT_RW, UVM_INH_NONE, 3222 UVM_ADV_RANDOM, 0)); 3223 if (error) { 3224 device_printf(sc->sc_dev, "uvm_map failed with %d\n", error); 3225 uao_detach(track->uobj); /* release reference */ 3226 goto abort; 3227 } 3228 3229 error = uvm_map_pageable(kernel_map, vstart, vstart + newvsize, 3230 false, 0); 3231 if (error) { 3232 device_printf(sc->sc_dev, "uvm_map_pageable failed with %d\n", 3233 error); 3234 uvm_unmap(kernel_map, vstart, vstart + newvsize); 3235 /* uvm_unmap also detach uobj */ 3236 goto abort; 3237 } 3238 3239 track->usrbuf.mem = (void *)vstart; 3240 track->usrbuf.capacity = newbufsize; 3241 memset(track->usrbuf.mem, 0, newvsize); 3242 return 0; 3243 3244 /* failure */ 3245 abort: 3246 track->uobj = NULL; /* paranoia */ 3247 track->usrbuf.mem = NULL; 3248 track->usrbuf.capacity = 0; 3249 return error; 3250 } 3251 3252 /* 3253 * Free usrbuf (if available). 3254 */ 3255 static void 3256 audio_free_usrbuf(audio_track_t *track) 3257 { 3258 vaddr_t vstart; 3259 vsize_t vsize; 3260 3261 vstart = (vaddr_t)track->usrbuf.mem; 3262 vsize = roundup2(MAX(track->usrbuf.capacity, PAGE_SIZE), PAGE_SIZE); 3263 if (track->usrbuf.mem != NULL) { 3264 /* 3265 * Unmap the kernel mapping. uvm_unmap releases the 3266 * reference to the uvm object, and this should be the 3267 * last virtual mapping of the uvm object, so no need 3268 * to explicitly release (`detach') the object. 3269 */ 3270 uvm_unmap(kernel_map, vstart, vstart + vsize); 3271 3272 track->uobj = NULL; 3273 track->usrbuf.mem = NULL; 3274 track->usrbuf.capacity = 0; 3275 } 3276 } 3277 3278 /* 3279 * This filter changes the volume for each channel. 3280 * arg->context points track->ch_volume[]. 3281 */ 3282 static void 3283 audio_track_chvol(audio_filter_arg_t *arg) 3284 { 3285 int16_t *ch_volume; 3286 const aint_t *s; 3287 aint_t *d; 3288 u_int i; 3289 u_int ch; 3290 u_int channels; 3291 3292 DIAGNOSTIC_filter_arg(arg); 3293 KASSERT(arg->srcfmt->channels == arg->dstfmt->channels); 3294 KASSERT(arg->context != NULL); 3295 KASSERT(arg->srcfmt->channels <= AUDIO_MAX_CHANNELS); 3296 3297 s = arg->src; 3298 d = arg->dst; 3299 ch_volume = arg->context; 3300 3301 channels = arg->srcfmt->channels; 3302 for (i = 0; i < arg->count; i++) { 3303 for (ch = 0; ch < channels; ch++) { 3304 aint2_t val; 3305 val = *s++; 3306 #if defined(AUDIO_USE_C_IMPLEMENTATION_DEFINED_BEHAVIOR) && defined(__GNUC__) 3307 val = val * ch_volume[ch] >> 8; 3308 #else 3309 val = val * ch_volume[ch] / 256; 3310 #endif 3311 *d++ = (aint_t)val; 3312 } 3313 } 3314 } 3315 3316 /* 3317 * This filter performs conversion from stereo (or more channels) to mono. 3318 */ 3319 static void 3320 audio_track_chmix_mixLR(audio_filter_arg_t *arg) 3321 { 3322 const aint_t *s; 3323 aint_t *d; 3324 u_int i; 3325 3326 DIAGNOSTIC_filter_arg(arg); 3327 3328 s = arg->src; 3329 d = arg->dst; 3330 3331 for (i = 0; i < arg->count; i++) { 3332 #if defined(AUDIO_USE_C_IMPLEMENTATION_DEFINED_BEHAVIOR) && defined(__GNUC__) 3333 *d++ = (s[0] >> 1) + (s[1] >> 1); 3334 #else 3335 *d++ = (s[0] / 2) + (s[1] / 2); 3336 #endif 3337 s += arg->srcfmt->channels; 3338 } 3339 } 3340 3341 /* 3342 * This filter performs conversion from mono to stereo (or more channels). 3343 */ 3344 static void 3345 audio_track_chmix_dupLR(audio_filter_arg_t *arg) 3346 { 3347 const aint_t *s; 3348 aint_t *d; 3349 u_int i; 3350 u_int ch; 3351 u_int dstchannels; 3352 3353 DIAGNOSTIC_filter_arg(arg); 3354 3355 s = arg->src; 3356 d = arg->dst; 3357 dstchannels = arg->dstfmt->channels; 3358 3359 for (i = 0; i < arg->count; i++) { 3360 d[0] = s[0]; 3361 d[1] = s[0]; 3362 s++; 3363 d += dstchannels; 3364 } 3365 if (dstchannels > 2) { 3366 d = arg->dst; 3367 for (i = 0; i < arg->count; i++) { 3368 for (ch = 2; ch < dstchannels; ch++) { 3369 d[ch] = 0; 3370 } 3371 d += dstchannels; 3372 } 3373 } 3374 } 3375 3376 /* 3377 * This filter shrinks M channels into N channels. 3378 * Extra channels are discarded. 3379 */ 3380 static void 3381 audio_track_chmix_shrink(audio_filter_arg_t *arg) 3382 { 3383 const aint_t *s; 3384 aint_t *d; 3385 u_int i; 3386 u_int ch; 3387 3388 DIAGNOSTIC_filter_arg(arg); 3389 3390 s = arg->src; 3391 d = arg->dst; 3392 3393 for (i = 0; i < arg->count; i++) { 3394 for (ch = 0; ch < arg->dstfmt->channels; ch++) { 3395 *d++ = s[ch]; 3396 } 3397 s += arg->srcfmt->channels; 3398 } 3399 } 3400 3401 /* 3402 * This filter expands M channels into N channels. 3403 * Silence is inserted for missing channels. 3404 */ 3405 static void 3406 audio_track_chmix_expand(audio_filter_arg_t *arg) 3407 { 3408 const aint_t *s; 3409 aint_t *d; 3410 u_int i; 3411 u_int ch; 3412 u_int srcchannels; 3413 u_int dstchannels; 3414 3415 DIAGNOSTIC_filter_arg(arg); 3416 3417 s = arg->src; 3418 d = arg->dst; 3419 3420 srcchannels = arg->srcfmt->channels; 3421 dstchannels = arg->dstfmt->channels; 3422 for (i = 0; i < arg->count; i++) { 3423 for (ch = 0; ch < srcchannels; ch++) { 3424 *d++ = *s++; 3425 } 3426 for (; ch < dstchannels; ch++) { 3427 *d++ = 0; 3428 } 3429 } 3430 } 3431 3432 /* 3433 * This filter performs frequency conversion (up sampling). 3434 * It uses linear interpolation. 3435 */ 3436 static void 3437 audio_track_freq_up(audio_filter_arg_t *arg) 3438 { 3439 audio_track_t *track; 3440 audio_ring_t *src; 3441 audio_ring_t *dst; 3442 const aint_t *s; 3443 aint_t *d; 3444 aint_t prev[AUDIO_MAX_CHANNELS]; 3445 aint_t curr[AUDIO_MAX_CHANNELS]; 3446 aint_t grad[AUDIO_MAX_CHANNELS]; 3447 u_int i; 3448 u_int t; 3449 u_int step; 3450 u_int channels; 3451 u_int ch; 3452 int srcused; 3453 3454 track = arg->context; 3455 KASSERT(track); 3456 src = &track->freq.srcbuf; 3457 dst = track->freq.dst; 3458 DIAGNOSTIC_ring(dst); 3459 DIAGNOSTIC_ring(src); 3460 KASSERT(src->used > 0); 3461 KASSERT(src->fmt.channels == dst->fmt.channels); 3462 KASSERT(src->head % track->mixer->frames_per_block == 0); 3463 3464 s = arg->src; 3465 d = arg->dst; 3466 3467 /* 3468 * In order to faciliate interpolation for each block, slide (delay) 3469 * input by one sample. As a result, strictly speaking, the output 3470 * phase is delayed by 1/dstfreq. However, I believe there is no 3471 * observable impact. 3472 * 3473 * Example) 3474 * srcfreq:dstfreq = 1:3 3475 * 3476 * A - - 3477 * | 3478 * | 3479 * | B - - 3480 * +-----+-----> input timeframe 3481 * 0 1 3482 * 3483 * 0 1 3484 * +-----+-----> input timeframe 3485 * | A 3486 * | x x 3487 * | x x 3488 * x (B) 3489 * +-+-+-+-+-+-> output timeframe 3490 * 0 1 2 3 4 5 3491 */ 3492 3493 /* Last samples in previous block */ 3494 channels = src->fmt.channels; 3495 for (ch = 0; ch < channels; ch++) { 3496 prev[ch] = track->freq_prev[ch]; 3497 curr[ch] = track->freq_curr[ch]; 3498 grad[ch] = curr[ch] - prev[ch]; 3499 } 3500 3501 step = track->freq_step; 3502 t = track->freq_current; 3503 //#define FREQ_DEBUG 3504 #if defined(FREQ_DEBUG) 3505 #define PRINTF(fmt...) printf(fmt) 3506 #else 3507 #define PRINTF(fmt...) do { } while (0) 3508 #endif 3509 srcused = src->used; 3510 PRINTF("upstart step=%d leap=%d", step, track->freq_leap); 3511 PRINTF(" srcused=%d arg->count=%u", src->used, arg->count); 3512 PRINTF(" prev=%d curr=%d grad=%d", prev[0], curr[0], grad[0]); 3513 PRINTF(" t=%d\n", t); 3514 3515 for (i = 0; i < arg->count; i++) { 3516 PRINTF("i=%d t=%5d", i, t); 3517 if (t >= 65536) { 3518 for (ch = 0; ch < channels; ch++) { 3519 prev[ch] = curr[ch]; 3520 curr[ch] = *s++; 3521 grad[ch] = curr[ch] - prev[ch]; 3522 } 3523 PRINTF(" prev=%d s[%d]=%d", 3524 prev[0], src->used - srcused, curr[0]); 3525 3526 /* Update */ 3527 t -= 65536; 3528 srcused--; 3529 if (srcused < 0) { 3530 PRINTF(" break\n"); 3531 break; 3532 } 3533 } 3534 3535 for (ch = 0; ch < channels; ch++) { 3536 *d++ = prev[ch] + (aint2_t)grad[ch] * t / 65536; 3537 #if defined(FREQ_DEBUG) 3538 if (ch == 0) 3539 printf(" t=%5d *d=%d", t, d[-1]); 3540 #endif 3541 } 3542 t += step; 3543 3544 PRINTF("\n"); 3545 } 3546 PRINTF("end prev=%d curr=%d\n", prev[0], curr[0]); 3547 3548 auring_take(src, src->used); 3549 auring_push(dst, i); 3550 3551 /* Adjust */ 3552 t += track->freq_leap; 3553 3554 track->freq_current = t; 3555 for (ch = 0; ch < channels; ch++) { 3556 track->freq_prev[ch] = prev[ch]; 3557 track->freq_curr[ch] = curr[ch]; 3558 } 3559 } 3560 3561 /* 3562 * This filter performs frequency conversion (down sampling). 3563 * It uses simple thinning. 3564 */ 3565 static void 3566 audio_track_freq_down(audio_filter_arg_t *arg) 3567 { 3568 audio_track_t *track; 3569 audio_ring_t *src; 3570 audio_ring_t *dst; 3571 const aint_t *s0; 3572 aint_t *d; 3573 u_int i; 3574 u_int t; 3575 u_int step; 3576 u_int ch; 3577 u_int channels; 3578 3579 track = arg->context; 3580 KASSERT(track); 3581 src = &track->freq.srcbuf; 3582 dst = track->freq.dst; 3583 3584 DIAGNOSTIC_ring(dst); 3585 DIAGNOSTIC_ring(src); 3586 KASSERT(src->used > 0); 3587 KASSERT(src->fmt.channels == dst->fmt.channels); 3588 KASSERTMSG(src->head % track->mixer->frames_per_block == 0, 3589 "src->head=%d fpb=%d", 3590 src->head, track->mixer->frames_per_block); 3591 3592 s0 = arg->src; 3593 d = arg->dst; 3594 t = track->freq_current; 3595 step = track->freq_step; 3596 channels = dst->fmt.channels; 3597 PRINTF("downstart step=%d leap=%d", step, track->freq_leap); 3598 PRINTF(" srcused=%d arg->count=%u", src->used, arg->count); 3599 PRINTF(" t=%d\n", t); 3600 3601 for (i = 0; i < arg->count && t / 65536 < src->used; i++) { 3602 const aint_t *s; 3603 PRINTF("i=%4d t=%10d", i, t); 3604 s = s0 + (t / 65536) * channels; 3605 PRINTF(" s=%5ld", (s - s0) / channels); 3606 for (ch = 0; ch < channels; ch++) { 3607 if (ch == 0) PRINTF(" *s=%d", s[ch]); 3608 *d++ = s[ch]; 3609 } 3610 PRINTF("\n"); 3611 t += step; 3612 } 3613 t += track->freq_leap; 3614 PRINTF("end t=%d\n", t); 3615 auring_take(src, src->used); 3616 auring_push(dst, i); 3617 track->freq_current = t % 65536; 3618 } 3619 3620 /* 3621 * Creates track and returns it. 3622 */ 3623 audio_track_t * 3624 audio_track_create(struct audio_softc *sc, audio_trackmixer_t *mixer) 3625 { 3626 audio_track_t *track; 3627 static int newid = 0; 3628 3629 track = kmem_zalloc(sizeof(*track), KM_SLEEP); 3630 3631 track->id = newid++; 3632 track->mixer = mixer; 3633 track->mode = mixer->mode; 3634 3635 /* Do TRACE after id is assigned. */ 3636 TRACET(3, track, "for %s", 3637 mixer->mode == AUMODE_PLAY ? "playback" : "recording"); 3638 3639 #if defined(AUDIO_SUPPORT_TRACK_VOLUME) 3640 track->volume = 256; 3641 #endif 3642 for (int i = 0; i < AUDIO_MAX_CHANNELS; i++) { 3643 track->ch_volume[i] = 256; 3644 } 3645 3646 return track; 3647 } 3648 3649 /* 3650 * Release all resources of the track and track itself. 3651 * track must not be NULL. Don't specify the track within the file 3652 * structure linked from sc->sc_files. 3653 */ 3654 static void 3655 audio_track_destroy(audio_track_t *track) 3656 { 3657 3658 KASSERT(track); 3659 3660 audio_free_usrbuf(track); 3661 audio_free(track->codec.srcbuf.mem); 3662 audio_free(track->chvol.srcbuf.mem); 3663 audio_free(track->chmix.srcbuf.mem); 3664 audio_free(track->freq.srcbuf.mem); 3665 audio_free(track->outbuf.mem); 3666 3667 kmem_free(track, sizeof(*track)); 3668 } 3669 3670 /* 3671 * It returns encoding conversion filter according to src and dst format. 3672 * If it is not a convertible pair, it returns NULL. Either src or dst 3673 * must be internal format. 3674 */ 3675 static audio_filter_t 3676 audio_track_get_codec(audio_track_t *track, const audio_format2_t *src, 3677 const audio_format2_t *dst) 3678 { 3679 3680 if (audio_format2_is_internal(src)) { 3681 if (dst->encoding == AUDIO_ENCODING_ULAW) { 3682 return audio_internal_to_mulaw; 3683 } else if (dst->encoding == AUDIO_ENCODING_ALAW) { 3684 return audio_internal_to_alaw; 3685 } else if (audio_format2_is_linear(dst)) { 3686 switch (dst->stride) { 3687 case 8: 3688 return audio_internal_to_linear8; 3689 case 16: 3690 return audio_internal_to_linear16; 3691 #if defined(AUDIO_SUPPORT_LINEAR24) 3692 case 24: 3693 return audio_internal_to_linear24; 3694 #endif 3695 case 32: 3696 return audio_internal_to_linear32; 3697 default: 3698 TRACET(1, track, "unsupported %s stride %d", 3699 "dst", dst->stride); 3700 goto abort; 3701 } 3702 } 3703 } else if (audio_format2_is_internal(dst)) { 3704 if (src->encoding == AUDIO_ENCODING_ULAW) { 3705 return audio_mulaw_to_internal; 3706 } else if (src->encoding == AUDIO_ENCODING_ALAW) { 3707 return audio_alaw_to_internal; 3708 } else if (audio_format2_is_linear(src)) { 3709 switch (src->stride) { 3710 case 8: 3711 return audio_linear8_to_internal; 3712 case 16: 3713 return audio_linear16_to_internal; 3714 #if defined(AUDIO_SUPPORT_LINEAR24) 3715 case 24: 3716 return audio_linear24_to_internal; 3717 #endif 3718 case 32: 3719 return audio_linear32_to_internal; 3720 default: 3721 TRACET(1, track, "unsupported %s stride %d", 3722 "src", src->stride); 3723 goto abort; 3724 } 3725 } 3726 } 3727 3728 TRACET(1, track, "unsupported encoding"); 3729 abort: 3730 #if defined(AUDIO_DEBUG) 3731 if (audiodebug >= 2) { 3732 char buf[100]; 3733 audio_format2_tostr(buf, sizeof(buf), src); 3734 TRACET(2, track, "src %s", buf); 3735 audio_format2_tostr(buf, sizeof(buf), dst); 3736 TRACET(2, track, "dst %s", buf); 3737 } 3738 #endif 3739 return NULL; 3740 } 3741 3742 /* 3743 * Initialize the codec stage of this track as necessary. 3744 * If successful, it initializes the codec stage as necessary, stores updated 3745 * last_dst in *last_dstp in any case, and returns 0. 3746 * Otherwise, it returns errno without modifying *last_dstp. 3747 */ 3748 static int 3749 audio_track_init_codec(audio_track_t *track, audio_ring_t **last_dstp) 3750 { 3751 struct audio_softc *sc; 3752 audio_ring_t *last_dst; 3753 audio_ring_t *srcbuf; 3754 audio_format2_t *srcfmt; 3755 audio_format2_t *dstfmt; 3756 audio_filter_arg_t *arg; 3757 u_int len; 3758 int error; 3759 3760 KASSERT(track); 3761 3762 sc = track->mixer->sc; 3763 last_dst = *last_dstp; 3764 dstfmt = &last_dst->fmt; 3765 srcfmt = &track->inputfmt; 3766 srcbuf = &track->codec.srcbuf; 3767 error = 0; 3768 3769 if (srcfmt->encoding != dstfmt->encoding 3770 || srcfmt->precision != dstfmt->precision 3771 || srcfmt->stride != dstfmt->stride) { 3772 track->codec.dst = last_dst; 3773 3774 srcbuf->fmt = *dstfmt; 3775 srcbuf->fmt.encoding = srcfmt->encoding; 3776 srcbuf->fmt.precision = srcfmt->precision; 3777 srcbuf->fmt.stride = srcfmt->stride; 3778 3779 track->codec.filter = audio_track_get_codec(track, 3780 &srcbuf->fmt, dstfmt); 3781 if (track->codec.filter == NULL) { 3782 error = EINVAL; 3783 goto abort; 3784 } 3785 3786 srcbuf->head = 0; 3787 srcbuf->used = 0; 3788 srcbuf->capacity = frame_per_block(track->mixer, &srcbuf->fmt); 3789 len = auring_bytelen(srcbuf); 3790 srcbuf->mem = audio_realloc(srcbuf->mem, len); 3791 if (srcbuf->mem == NULL) { 3792 device_printf(sc->sc_dev, "%s: malloc(%d) failed\n", 3793 __func__, len); 3794 error = ENOMEM; 3795 goto abort; 3796 } 3797 3798 arg = &track->codec.arg; 3799 arg->srcfmt = &srcbuf->fmt; 3800 arg->dstfmt = dstfmt; 3801 arg->context = NULL; 3802 3803 *last_dstp = srcbuf; 3804 return 0; 3805 } 3806 3807 abort: 3808 track->codec.filter = NULL; 3809 audio_free(srcbuf->mem); 3810 return error; 3811 } 3812 3813 /* 3814 * Initialize the chvol stage of this track as necessary. 3815 * If successful, it initializes the chvol stage as necessary, stores updated 3816 * last_dst in *last_dstp in any case, and returns 0. 3817 * Otherwise, it returns errno without modifying *last_dstp. 3818 */ 3819 static int 3820 audio_track_init_chvol(audio_track_t *track, audio_ring_t **last_dstp) 3821 { 3822 struct audio_softc *sc; 3823 audio_ring_t *last_dst; 3824 audio_ring_t *srcbuf; 3825 audio_format2_t *srcfmt; 3826 audio_format2_t *dstfmt; 3827 audio_filter_arg_t *arg; 3828 u_int len; 3829 int error; 3830 3831 KASSERT(track); 3832 3833 sc = track->mixer->sc; 3834 last_dst = *last_dstp; 3835 dstfmt = &last_dst->fmt; 3836 srcfmt = &track->inputfmt; 3837 srcbuf = &track->chvol.srcbuf; 3838 error = 0; 3839 3840 /* Check whether channel volume conversion is necessary. */ 3841 bool use_chvol = false; 3842 for (int ch = 0; ch < srcfmt->channels; ch++) { 3843 if (track->ch_volume[ch] != 256) { 3844 use_chvol = true; 3845 break; 3846 } 3847 } 3848 3849 if (use_chvol == true) { 3850 track->chvol.dst = last_dst; 3851 track->chvol.filter = audio_track_chvol; 3852 3853 srcbuf->fmt = *dstfmt; 3854 /* no format conversion occurs */ 3855 3856 srcbuf->head = 0; 3857 srcbuf->used = 0; 3858 srcbuf->capacity = frame_per_block(track->mixer, &srcbuf->fmt); 3859 len = auring_bytelen(srcbuf); 3860 srcbuf->mem = audio_realloc(srcbuf->mem, len); 3861 if (srcbuf->mem == NULL) { 3862 device_printf(sc->sc_dev, "%s: malloc(%d) failed\n", 3863 __func__, len); 3864 error = ENOMEM; 3865 goto abort; 3866 } 3867 3868 arg = &track->chvol.arg; 3869 arg->srcfmt = &srcbuf->fmt; 3870 arg->dstfmt = dstfmt; 3871 arg->context = track->ch_volume; 3872 3873 *last_dstp = srcbuf; 3874 return 0; 3875 } 3876 3877 abort: 3878 track->chvol.filter = NULL; 3879 audio_free(srcbuf->mem); 3880 return error; 3881 } 3882 3883 /* 3884 * Initialize the chmix stage of this track as necessary. 3885 * If successful, it initializes the chmix stage as necessary, stores updated 3886 * last_dst in *last_dstp in any case, and returns 0. 3887 * Otherwise, it returns errno without modifying *last_dstp. 3888 */ 3889 static int 3890 audio_track_init_chmix(audio_track_t *track, audio_ring_t **last_dstp) 3891 { 3892 struct audio_softc *sc; 3893 audio_ring_t *last_dst; 3894 audio_ring_t *srcbuf; 3895 audio_format2_t *srcfmt; 3896 audio_format2_t *dstfmt; 3897 audio_filter_arg_t *arg; 3898 u_int srcch; 3899 u_int dstch; 3900 u_int len; 3901 int error; 3902 3903 KASSERT(track); 3904 3905 sc = track->mixer->sc; 3906 last_dst = *last_dstp; 3907 dstfmt = &last_dst->fmt; 3908 srcfmt = &track->inputfmt; 3909 srcbuf = &track->chmix.srcbuf; 3910 error = 0; 3911 3912 srcch = srcfmt->channels; 3913 dstch = dstfmt->channels; 3914 if (srcch != dstch) { 3915 track->chmix.dst = last_dst; 3916 3917 if (srcch >= 2 && dstch == 1) { 3918 track->chmix.filter = audio_track_chmix_mixLR; 3919 } else if (srcch == 1 && dstch >= 2) { 3920 track->chmix.filter = audio_track_chmix_dupLR; 3921 } else if (srcch > dstch) { 3922 track->chmix.filter = audio_track_chmix_shrink; 3923 } else { 3924 track->chmix.filter = audio_track_chmix_expand; 3925 } 3926 3927 srcbuf->fmt = *dstfmt; 3928 srcbuf->fmt.channels = srcch; 3929 3930 srcbuf->head = 0; 3931 srcbuf->used = 0; 3932 /* XXX The buffer size should be able to calculate. */ 3933 srcbuf->capacity = frame_per_block(track->mixer, &srcbuf->fmt); 3934 len = auring_bytelen(srcbuf); 3935 srcbuf->mem = audio_realloc(srcbuf->mem, len); 3936 if (srcbuf->mem == NULL) { 3937 device_printf(sc->sc_dev, "%s: malloc(%d) failed\n", 3938 __func__, len); 3939 error = ENOMEM; 3940 goto abort; 3941 } 3942 3943 arg = &track->chmix.arg; 3944 arg->srcfmt = &srcbuf->fmt; 3945 arg->dstfmt = dstfmt; 3946 arg->context = NULL; 3947 3948 *last_dstp = srcbuf; 3949 return 0; 3950 } 3951 3952 abort: 3953 track->chmix.filter = NULL; 3954 audio_free(srcbuf->mem); 3955 return error; 3956 } 3957 3958 /* 3959 * Initialize the freq stage of this track as necessary. 3960 * If successful, it initializes the freq stage as necessary, stores updated 3961 * last_dst in *last_dstp in any case, and returns 0. 3962 * Otherwise, it returns errno without modifying *last_dstp. 3963 */ 3964 static int 3965 audio_track_init_freq(audio_track_t *track, audio_ring_t **last_dstp) 3966 { 3967 struct audio_softc *sc; 3968 audio_ring_t *last_dst; 3969 audio_ring_t *srcbuf; 3970 audio_format2_t *srcfmt; 3971 audio_format2_t *dstfmt; 3972 audio_filter_arg_t *arg; 3973 uint32_t srcfreq; 3974 uint32_t dstfreq; 3975 u_int dst_capacity; 3976 u_int mod; 3977 u_int len; 3978 int error; 3979 3980 KASSERT(track); 3981 3982 sc = track->mixer->sc; 3983 last_dst = *last_dstp; 3984 dstfmt = &last_dst->fmt; 3985 srcfmt = &track->inputfmt; 3986 srcbuf = &track->freq.srcbuf; 3987 error = 0; 3988 3989 srcfreq = srcfmt->sample_rate; 3990 dstfreq = dstfmt->sample_rate; 3991 if (srcfreq != dstfreq) { 3992 track->freq.dst = last_dst; 3993 3994 memset(track->freq_prev, 0, sizeof(track->freq_prev)); 3995 memset(track->freq_curr, 0, sizeof(track->freq_curr)); 3996 3997 /* freq_step is the ratio of src/dst when let dst 65536. */ 3998 track->freq_step = (uint64_t)srcfreq * 65536 / dstfreq; 3999 4000 dst_capacity = frame_per_block(track->mixer, dstfmt); 4001 mod = (uint64_t)srcfreq * 65536 % dstfreq; 4002 track->freq_leap = (mod * dst_capacity + dstfreq / 2) / dstfreq; 4003 4004 if (track->freq_step < 65536) { 4005 track->freq.filter = audio_track_freq_up; 4006 /* In order to carry at the first time. */ 4007 track->freq_current = 65536; 4008 } else { 4009 track->freq.filter = audio_track_freq_down; 4010 track->freq_current = 0; 4011 } 4012 4013 srcbuf->fmt = *dstfmt; 4014 srcbuf->fmt.sample_rate = srcfreq; 4015 4016 srcbuf->head = 0; 4017 srcbuf->used = 0; 4018 srcbuf->capacity = frame_per_block(track->mixer, &srcbuf->fmt); 4019 len = auring_bytelen(srcbuf); 4020 srcbuf->mem = audio_realloc(srcbuf->mem, len); 4021 if (srcbuf->mem == NULL) { 4022 device_printf(sc->sc_dev, "%s: malloc(%d) failed\n", 4023 __func__, len); 4024 error = ENOMEM; 4025 goto abort; 4026 } 4027 4028 arg = &track->freq.arg; 4029 arg->srcfmt = &srcbuf->fmt; 4030 arg->dstfmt = dstfmt;/*&last_dst->fmt;*/ 4031 arg->context = track; 4032 4033 *last_dstp = srcbuf; 4034 return 0; 4035 } 4036 4037 abort: 4038 track->freq.filter = NULL; 4039 audio_free(srcbuf->mem); 4040 return error; 4041 } 4042 4043 /* 4044 * When playing back: (e.g. if codec and freq stage are valid) 4045 * 4046 * write 4047 * | uiomove 4048 * v 4049 * usrbuf [...............] byte ring buffer (mmap-able) 4050 * | memcpy 4051 * v 4052 * codec.srcbuf[....] 1 block (ring) buffer <-- stage input 4053 * .dst ----+ 4054 * | convert 4055 * v 4056 * freq.srcbuf [....] 1 block (ring) buffer 4057 * .dst ----+ 4058 * | convert 4059 * v 4060 * outbuf [...............] NBLKOUT blocks ring buffer 4061 * 4062 * 4063 * When recording: 4064 * 4065 * freq.srcbuf [...............] NBLKOUT blocks ring buffer <-- stage input 4066 * .dst ----+ 4067 * | convert 4068 * v 4069 * codec.srcbuf[.....] 1 block (ring) buffer 4070 * .dst ----+ 4071 * | convert 4072 * v 4073 * outbuf [.....] 1 block (ring) buffer 4074 * | memcpy 4075 * v 4076 * usrbuf [...............] byte ring buffer (mmap-able *) 4077 * | uiomove 4078 * v 4079 * read 4080 * 4081 * *: usrbuf for recording is also mmap-able due to symmetry with 4082 * playback buffer, but for now mmap will never happen for recording. 4083 */ 4084 4085 /* 4086 * Set the userland format of this track. 4087 * usrfmt argument should be parameter verified with audio_check_params(). 4088 * It will release and reallocate all internal conversion buffers. 4089 * It returns 0 if successful. Otherwise it returns errno with clearing all 4090 * internal buffers. 4091 * It must be called without sc_intr_lock since uvm_* routines require non 4092 * intr_lock state. 4093 * It must be called with track lock held since it may release and reallocate 4094 * outbuf. 4095 */ 4096 static int 4097 audio_track_set_format(audio_track_t *track, audio_format2_t *usrfmt) 4098 { 4099 struct audio_softc *sc; 4100 u_int newbufsize; 4101 u_int oldblksize; 4102 u_int len; 4103 int error; 4104 4105 KASSERT(track); 4106 sc = track->mixer->sc; 4107 4108 /* usrbuf is the closest buffer to the userland. */ 4109 track->usrbuf.fmt = *usrfmt; 4110 4111 /* 4112 * For references, one block size (in 40msec) is: 4113 * 320 bytes = 204 blocks/64KB for mulaw/8kHz/1ch 4114 * 7680 bytes = 8 blocks/64KB for s16/48kHz/2ch 4115 * 30720 bytes = 90 KB/3blocks for s16/48kHz/8ch 4116 * 61440 bytes = 180 KB/3blocks for s16/96kHz/8ch 4117 * 245760 bytes = 720 KB/3blocks for s32/192kHz/8ch 4118 * 4119 * For example, 4120 * 1) If usrbuf_blksize = 7056 (s16/44.1k/2ch) and PAGE_SIZE = 8192, 4121 * newbufsize = rounddown(65536 / 7056) = 63504 4122 * newvsize = roundup2(63504, PAGE_SIZE) = 65536 4123 * Therefore it maps 8 * 8K pages and usrbuf->capacity = 63504. 4124 * 4125 * 2) If usrbuf_blksize = 7680 (s16/48k/2ch) and PAGE_SIZE = 4096, 4126 * newbufsize = rounddown(65536 / 7680) = 61440 4127 * newvsize = roundup2(61440, PAGE_SIZE) = 61440 (= 15 pages) 4128 * Therefore it maps 15 * 4K pages and usrbuf->capacity = 61440. 4129 */ 4130 oldblksize = track->usrbuf_blksize; 4131 track->usrbuf_blksize = frametobyte(&track->usrbuf.fmt, 4132 frame_per_block(track->mixer, &track->usrbuf.fmt)); 4133 track->usrbuf.head = 0; 4134 track->usrbuf.used = 0; 4135 newbufsize = MAX(track->usrbuf_blksize * AUMINNOBLK, 65536); 4136 newbufsize = rounddown(newbufsize, track->usrbuf_blksize); 4137 error = audio_realloc_usrbuf(track, newbufsize); 4138 if (error) { 4139 device_printf(sc->sc_dev, "malloc usrbuf(%d) failed\n", 4140 newbufsize); 4141 goto error; 4142 } 4143 4144 /* Recalc water mark. */ 4145 if (track->usrbuf_blksize != oldblksize) { 4146 if (audio_track_is_playback(track)) { 4147 /* Set high at 100%, low at 75%. */ 4148 track->usrbuf_usedhigh = track->usrbuf.capacity; 4149 track->usrbuf_usedlow = track->usrbuf.capacity * 3 / 4; 4150 } else { 4151 /* Set high at 100% minus 1block(?), low at 0% */ 4152 track->usrbuf_usedhigh = track->usrbuf.capacity - 4153 track->usrbuf_blksize; 4154 track->usrbuf_usedlow = 0; 4155 } 4156 } 4157 4158 /* Stage buffer */ 4159 audio_ring_t *last_dst = &track->outbuf; 4160 if (audio_track_is_playback(track)) { 4161 /* On playback, initialize from the mixer side in order. */ 4162 track->inputfmt = *usrfmt; 4163 track->outbuf.fmt = track->mixer->track_fmt; 4164 4165 if ((error = audio_track_init_freq(track, &last_dst)) != 0) 4166 goto error; 4167 if ((error = audio_track_init_chmix(track, &last_dst)) != 0) 4168 goto error; 4169 if ((error = audio_track_init_chvol(track, &last_dst)) != 0) 4170 goto error; 4171 if ((error = audio_track_init_codec(track, &last_dst)) != 0) 4172 goto error; 4173 } else { 4174 /* On recording, initialize from userland side in order. */ 4175 track->inputfmt = track->mixer->track_fmt; 4176 track->outbuf.fmt = *usrfmt; 4177 4178 if ((error = audio_track_init_codec(track, &last_dst)) != 0) 4179 goto error; 4180 if ((error = audio_track_init_chvol(track, &last_dst)) != 0) 4181 goto error; 4182 if ((error = audio_track_init_chmix(track, &last_dst)) != 0) 4183 goto error; 4184 if ((error = audio_track_init_freq(track, &last_dst)) != 0) 4185 goto error; 4186 } 4187 #if 0 4188 /* debug */ 4189 if (track->freq.filter) { 4190 audio_print_format2("freq src", &track->freq.srcbuf.fmt); 4191 audio_print_format2("freq dst", &track->freq.dst->fmt); 4192 } 4193 if (track->chmix.filter) { 4194 audio_print_format2("chmix src", &track->chmix.srcbuf.fmt); 4195 audio_print_format2("chmix dst", &track->chmix.dst->fmt); 4196 } 4197 if (track->chvol.filter) { 4198 audio_print_format2("chvol src", &track->chvol.srcbuf.fmt); 4199 audio_print_format2("chvol dst", &track->chvol.dst->fmt); 4200 } 4201 if (track->codec.filter) { 4202 audio_print_format2("codec src", &track->codec.srcbuf.fmt); 4203 audio_print_format2("codec dst", &track->codec.dst->fmt); 4204 } 4205 #endif 4206 4207 /* Stage input buffer */ 4208 track->input = last_dst; 4209 4210 /* 4211 * On the recording track, make the first stage a ring buffer. 4212 * XXX is there a better way? 4213 */ 4214 if (audio_track_is_record(track)) { 4215 track->input->capacity = NBLKOUT * 4216 frame_per_block(track->mixer, &track->input->fmt); 4217 len = auring_bytelen(track->input); 4218 track->input->mem = audio_realloc(track->input->mem, len); 4219 if (track->input->mem == NULL) { 4220 device_printf(sc->sc_dev, "malloc input(%d) failed\n", 4221 len); 4222 error = ENOMEM; 4223 goto error; 4224 } 4225 } 4226 4227 /* 4228 * Output buffer. 4229 * On the playback track, its capacity is NBLKOUT blocks. 4230 * On the recording track, its capacity is 1 block. 4231 */ 4232 track->outbuf.head = 0; 4233 track->outbuf.used = 0; 4234 track->outbuf.capacity = frame_per_block(track->mixer, 4235 &track->outbuf.fmt); 4236 if (audio_track_is_playback(track)) 4237 track->outbuf.capacity *= NBLKOUT; 4238 len = auring_bytelen(&track->outbuf); 4239 track->outbuf.mem = audio_realloc(track->outbuf.mem, len); 4240 if (track->outbuf.mem == NULL) { 4241 device_printf(sc->sc_dev, "malloc outbuf(%d) failed\n", len); 4242 error = ENOMEM; 4243 goto error; 4244 } 4245 4246 #if defined(AUDIO_DEBUG) 4247 if (audiodebug >= 3) { 4248 struct audio_track_debugbuf m; 4249 4250 memset(&m, 0, sizeof(m)); 4251 snprintf(m.outbuf, sizeof(m.outbuf), " out=%d", 4252 track->outbuf.capacity * frametobyte(&track->outbuf.fmt,1)); 4253 if (track->freq.filter) 4254 snprintf(m.freq, sizeof(m.freq), " freq=%d", 4255 track->freq.srcbuf.capacity * 4256 frametobyte(&track->freq.srcbuf.fmt, 1)); 4257 if (track->chmix.filter) 4258 snprintf(m.chmix, sizeof(m.chmix), " chmix=%d", 4259 track->chmix.srcbuf.capacity * 4260 frametobyte(&track->chmix.srcbuf.fmt, 1)); 4261 if (track->chvol.filter) 4262 snprintf(m.chvol, sizeof(m.chvol), " chvol=%d", 4263 track->chvol.srcbuf.capacity * 4264 frametobyte(&track->chvol.srcbuf.fmt, 1)); 4265 if (track->codec.filter) 4266 snprintf(m.codec, sizeof(m.codec), " codec=%d", 4267 track->codec.srcbuf.capacity * 4268 frametobyte(&track->codec.srcbuf.fmt, 1)); 4269 snprintf(m.usrbuf, sizeof(m.usrbuf), 4270 " usr=%d", track->usrbuf.capacity); 4271 4272 if (audio_track_is_playback(track)) { 4273 TRACET(0, track, "bufsize%s%s%s%s%s%s", 4274 m.outbuf, m.freq, m.chmix, 4275 m.chvol, m.codec, m.usrbuf); 4276 } else { 4277 TRACET(0, track, "bufsize%s%s%s%s%s%s", 4278 m.freq, m.chmix, m.chvol, 4279 m.codec, m.outbuf, m.usrbuf); 4280 } 4281 } 4282 #endif 4283 return 0; 4284 4285 error: 4286 audio_free_usrbuf(track); 4287 audio_free(track->codec.srcbuf.mem); 4288 audio_free(track->chvol.srcbuf.mem); 4289 audio_free(track->chmix.srcbuf.mem); 4290 audio_free(track->freq.srcbuf.mem); 4291 audio_free(track->outbuf.mem); 4292 return error; 4293 } 4294 4295 /* 4296 * Fill silence frames (as the internal format) up to 1 block 4297 * if the ring is not empty and less than 1 block. 4298 * It returns the number of appended frames. 4299 */ 4300 static int 4301 audio_append_silence(audio_track_t *track, audio_ring_t *ring) 4302 { 4303 int fpb; 4304 int n; 4305 4306 KASSERT(track); 4307 KASSERT(audio_format2_is_internal(&ring->fmt)); 4308 4309 /* XXX is n correct? */ 4310 /* XXX memset uses frametobyte()? */ 4311 4312 if (ring->used == 0) 4313 return 0; 4314 4315 fpb = frame_per_block(track->mixer, &ring->fmt); 4316 if (ring->used >= fpb) 4317 return 0; 4318 4319 n = (ring->capacity - ring->used) % fpb; 4320 4321 KASSERT(auring_get_contig_free(ring) >= n); 4322 4323 memset(auring_tailptr_aint(ring), 0, 4324 n * ring->fmt.channels * sizeof(aint_t)); 4325 auring_push(ring, n); 4326 return n; 4327 } 4328 4329 /* 4330 * Execute the conversion stage. 4331 * It prepares arg from this stage and executes stage->filter. 4332 * It must be called only if stage->filter is not NULL. 4333 * 4334 * For stages other than frequency conversion, the function increments 4335 * src and dst counters here. For frequency conversion stage, on the 4336 * other hand, the function does not touch src and dst counters and 4337 * filter side has to increment them. 4338 */ 4339 static void 4340 audio_apply_stage(audio_track_t *track, audio_stage_t *stage, bool isfreq) 4341 { 4342 audio_filter_arg_t *arg; 4343 int srccount; 4344 int dstcount; 4345 int count; 4346 4347 KASSERT(track); 4348 KASSERT(stage->filter); 4349 4350 srccount = auring_get_contig_used(&stage->srcbuf); 4351 dstcount = auring_get_contig_free(stage->dst); 4352 4353 if (isfreq) { 4354 KASSERTMSG(srccount > 0, "freq but srccount == %d", srccount); 4355 count = uimin(dstcount, track->mixer->frames_per_block); 4356 } else { 4357 count = uimin(srccount, dstcount); 4358 } 4359 4360 if (count > 0) { 4361 arg = &stage->arg; 4362 arg->src = auring_headptr(&stage->srcbuf); 4363 arg->dst = auring_tailptr(stage->dst); 4364 arg->count = count; 4365 4366 stage->filter(arg); 4367 4368 if (!isfreq) { 4369 auring_take(&stage->srcbuf, count); 4370 auring_push(stage->dst, count); 4371 } 4372 } 4373 } 4374 4375 /* 4376 * Produce output buffer for playback from user input buffer. 4377 * It must be called only if usrbuf is not empty and outbuf is 4378 * available at least one free block. 4379 */ 4380 static void 4381 audio_track_play(audio_track_t *track) 4382 { 4383 audio_ring_t *usrbuf; 4384 audio_ring_t *input; 4385 int count; 4386 int framesize; 4387 int bytes; 4388 u_int dropcount; 4389 4390 KASSERT(track); 4391 KASSERT(track->lock); 4392 TRACET(4, track, "start pstate=%d", track->pstate); 4393 4394 /* At this point usrbuf must not be empty. */ 4395 KASSERT(track->usrbuf.used > 0); 4396 /* Also, outbuf must be available at least one block. */ 4397 count = auring_get_contig_free(&track->outbuf); 4398 KASSERTMSG(count >= frame_per_block(track->mixer, &track->outbuf.fmt), 4399 "count=%d fpb=%d", 4400 count, frame_per_block(track->mixer, &track->outbuf.fmt)); 4401 4402 /* XXX TODO: is this necessary for now? */ 4403 int track_count_0 = track->outbuf.used; 4404 4405 usrbuf = &track->usrbuf; 4406 input = track->input; 4407 dropcount = 0; 4408 4409 /* 4410 * framesize is always 1 byte or more since all formats supported as 4411 * usrfmt(=input) have 8bit or more stride. 4412 */ 4413 framesize = frametobyte(&input->fmt, 1); 4414 KASSERT(framesize >= 1); 4415 4416 /* The next stage of usrbuf (=input) must be available. */ 4417 KASSERT(auring_get_contig_free(input) > 0); 4418 4419 /* 4420 * Copy usrbuf up to 1block to input buffer. 4421 * count is the number of frames to copy from usrbuf. 4422 * bytes is the number of bytes to copy from usrbuf. However it is 4423 * not copied less than one frame. 4424 */ 4425 count = uimin(usrbuf->used, track->usrbuf_blksize) / framesize; 4426 bytes = count * framesize; 4427 4428 /* 4429 * If bytes is less than one block, 4430 * if not draining, buffer is not filled so return. 4431 * if draining, fall through. 4432 */ 4433 if (count < track->usrbuf_blksize / framesize) { 4434 dropcount = track->usrbuf_blksize / framesize - count; 4435 4436 if (track->pstate != AUDIO_STATE_DRAINING) { 4437 /* Wait until filled. */ 4438 TRACET(4, track, "not enough; return"); 4439 return; 4440 } 4441 } 4442 4443 track->usrbuf_stamp += bytes; 4444 4445 if (usrbuf->head + bytes < usrbuf->capacity) { 4446 memcpy((uint8_t *)input->mem + auring_tail(input) * framesize, 4447 (uint8_t *)usrbuf->mem + usrbuf->head, 4448 bytes); 4449 auring_push(input, count); 4450 auring_take(usrbuf, bytes); 4451 } else { 4452 int bytes1; 4453 int bytes2; 4454 4455 bytes1 = auring_get_contig_used(usrbuf); 4456 KASSERT(bytes1 % framesize == 0); 4457 memcpy((uint8_t *)input->mem + auring_tail(input) * framesize, 4458 (uint8_t *)usrbuf->mem + usrbuf->head, 4459 bytes1); 4460 auring_push(input, bytes1 / framesize); 4461 auring_take(usrbuf, bytes1); 4462 4463 bytes2 = bytes - bytes1; 4464 memcpy((uint8_t *)input->mem + auring_tail(input) * framesize, 4465 (uint8_t *)usrbuf->mem + usrbuf->head, 4466 bytes2); 4467 auring_push(input, bytes2 / framesize); 4468 auring_take(usrbuf, bytes2); 4469 } 4470 4471 /* Encoding conversion */ 4472 if (track->codec.filter) 4473 audio_apply_stage(track, &track->codec, false); 4474 4475 /* Channel volume */ 4476 if (track->chvol.filter) 4477 audio_apply_stage(track, &track->chvol, false); 4478 4479 /* Channel mix */ 4480 if (track->chmix.filter) 4481 audio_apply_stage(track, &track->chmix, false); 4482 4483 /* Frequency conversion */ 4484 /* 4485 * Since the frequency conversion needs correction for each block, 4486 * it rounds up to 1 block. 4487 */ 4488 if (track->freq.filter) { 4489 int n; 4490 n = audio_append_silence(track, &track->freq.srcbuf); 4491 if (n > 0) { 4492 TRACET(4, track, 4493 "freq.srcbuf add silence %d -> %d/%d/%d", 4494 n, 4495 track->freq.srcbuf.head, 4496 track->freq.srcbuf.used, 4497 track->freq.srcbuf.capacity); 4498 } 4499 if (track->freq.srcbuf.used > 0) { 4500 audio_apply_stage(track, &track->freq, true); 4501 } 4502 } 4503 4504 if (dropcount != 0) { 4505 /* 4506 * Clear all conversion buffer pointer if the conversion was 4507 * not exactly one block. These conversion stage buffers are 4508 * certainly circular buffers because of symmetry with the 4509 * previous and next stage buffer. However, since they are 4510 * treated as simple contiguous buffers in operation, so head 4511 * always should point 0. This may happen during drain-age. 4512 */ 4513 TRACET(4, track, "reset stage"); 4514 if (track->codec.filter) { 4515 KASSERT(track->codec.srcbuf.used == 0); 4516 track->codec.srcbuf.head = 0; 4517 } 4518 if (track->chvol.filter) { 4519 KASSERT(track->chvol.srcbuf.used == 0); 4520 track->chvol.srcbuf.head = 0; 4521 } 4522 if (track->chmix.filter) { 4523 KASSERT(track->chmix.srcbuf.used == 0); 4524 track->chmix.srcbuf.head = 0; 4525 } 4526 if (track->freq.filter) { 4527 KASSERT(track->freq.srcbuf.used == 0); 4528 track->freq.srcbuf.head = 0; 4529 } 4530 } 4531 4532 if (track->input == &track->outbuf) { 4533 track->outputcounter = track->inputcounter; 4534 } else { 4535 track->outputcounter += track->outbuf.used - track_count_0; 4536 } 4537 4538 #if defined(AUDIO_DEBUG) 4539 if (audiodebug >= 3) { 4540 struct audio_track_debugbuf m; 4541 audio_track_bufstat(track, &m); 4542 TRACET(0, track, "end%s%s%s%s%s%s", 4543 m.outbuf, m.freq, m.chvol, m.chmix, m.codec, m.usrbuf); 4544 } 4545 #endif 4546 } 4547 4548 /* 4549 * Produce user output buffer for recording from input buffer. 4550 */ 4551 static void 4552 audio_track_record(audio_track_t *track) 4553 { 4554 audio_ring_t *outbuf; 4555 audio_ring_t *usrbuf; 4556 int count; 4557 int bytes; 4558 int framesize; 4559 4560 KASSERT(track); 4561 KASSERT(track->lock); 4562 4563 /* Number of frames to process */ 4564 count = auring_get_contig_used(track->input); 4565 count = uimin(count, track->mixer->frames_per_block); 4566 if (count == 0) { 4567 TRACET(4, track, "count == 0"); 4568 return; 4569 } 4570 4571 /* Frequency conversion */ 4572 if (track->freq.filter) { 4573 if (track->freq.srcbuf.used > 0) { 4574 audio_apply_stage(track, &track->freq, true); 4575 /* XXX should input of freq be from beginning of buf? */ 4576 } 4577 } 4578 4579 /* Channel mix */ 4580 if (track->chmix.filter) 4581 audio_apply_stage(track, &track->chmix, false); 4582 4583 /* Channel volume */ 4584 if (track->chvol.filter) 4585 audio_apply_stage(track, &track->chvol, false); 4586 4587 /* Encoding conversion */ 4588 if (track->codec.filter) 4589 audio_apply_stage(track, &track->codec, false); 4590 4591 /* Copy outbuf to usrbuf */ 4592 outbuf = &track->outbuf; 4593 usrbuf = &track->usrbuf; 4594 /* 4595 * framesize is always 1 byte or more since all formats supported 4596 * as usrfmt(=output) have 8bit or more stride. 4597 */ 4598 framesize = frametobyte(&outbuf->fmt, 1); 4599 KASSERT(framesize >= 1); 4600 /* 4601 * count is the number of frames to copy to usrbuf. 4602 * bytes is the number of bytes to copy to usrbuf. 4603 */ 4604 count = outbuf->used; 4605 count = uimin(count, 4606 (track->usrbuf_usedhigh - usrbuf->used) / framesize); 4607 bytes = count * framesize; 4608 if (auring_tail(usrbuf) + bytes < usrbuf->capacity) { 4609 memcpy((uint8_t *)usrbuf->mem + auring_tail(usrbuf), 4610 (uint8_t *)outbuf->mem + outbuf->head * framesize, 4611 bytes); 4612 auring_push(usrbuf, bytes); 4613 auring_take(outbuf, count); 4614 } else { 4615 int bytes1; 4616 int bytes2; 4617 4618 bytes1 = auring_get_contig_used(usrbuf); 4619 KASSERT(bytes1 % framesize == 0); 4620 memcpy((uint8_t *)usrbuf->mem + auring_tail(usrbuf), 4621 (uint8_t *)outbuf->mem + outbuf->head * framesize, 4622 bytes1); 4623 auring_push(usrbuf, bytes1); 4624 auring_take(outbuf, bytes1 / framesize); 4625 4626 bytes2 = bytes - bytes1; 4627 memcpy((uint8_t *)usrbuf->mem + auring_tail(usrbuf), 4628 (uint8_t *)outbuf->mem + outbuf->head * framesize, 4629 bytes2); 4630 auring_push(usrbuf, bytes2); 4631 auring_take(outbuf, bytes2 / framesize); 4632 } 4633 4634 /* XXX TODO: any counters here? */ 4635 4636 #if defined(AUDIO_DEBUG) 4637 if (audiodebug >= 3) { 4638 struct audio_track_debugbuf m; 4639 audio_track_bufstat(track, &m); 4640 TRACET(0, track, "end%s%s%s%s%s%s", 4641 m.freq, m.chvol, m.chmix, m.codec, m.outbuf, m.usrbuf); 4642 } 4643 #endif 4644 } 4645 4646 /* 4647 * Calcurate blktime [msec] from mixer(.hwbuf.fmt). 4648 * Must be called with sc_lock held. 4649 */ 4650 static u_int 4651 audio_mixer_calc_blktime(struct audio_softc *sc, audio_trackmixer_t *mixer) 4652 { 4653 audio_format2_t *fmt; 4654 u_int blktime; 4655 u_int frames_per_block; 4656 4657 KASSERT(mutex_owned(sc->sc_lock)); 4658 4659 fmt = &mixer->hwbuf.fmt; 4660 blktime = sc->sc_blk_ms; 4661 4662 /* 4663 * If stride is not multiples of 8, special treatment is necessary. 4664 * For now, it is only x68k's vs(4), 4 bit/sample ADPCM. 4665 */ 4666 if (fmt->stride == 4) { 4667 frames_per_block = fmt->sample_rate * blktime / 1000; 4668 if ((frames_per_block & 1) != 0) 4669 blktime *= 2; 4670 } 4671 #ifdef DIAGNOSTIC 4672 else if (fmt->stride % NBBY != 0) { 4673 panic("unsupported HW stride %d", fmt->stride); 4674 } 4675 #endif 4676 4677 return blktime; 4678 } 4679 4680 /* 4681 * Initialize the mixer corresponding to the mode. 4682 * Set AUMODE_PLAY to the 'mode' for playback or AUMODE_RECORD for recording. 4683 * sc->sc_[pr]mixer (corresponding to the 'mode') must be zero-filled. 4684 * This function returns 0 on sucessful. Otherwise returns errno. 4685 * Must be called with sc_lock held. 4686 */ 4687 static int 4688 audio_mixer_init(struct audio_softc *sc, int mode, 4689 const audio_format2_t *hwfmt, const audio_filter_reg_t *reg) 4690 { 4691 char codecbuf[64]; 4692 audio_trackmixer_t *mixer; 4693 void (*softint_handler)(void *); 4694 int len; 4695 int blksize; 4696 int capacity; 4697 size_t bufsize; 4698 int hwblks; 4699 int blkms; 4700 int error; 4701 4702 KASSERT(hwfmt != NULL); 4703 KASSERT(reg != NULL); 4704 KASSERT(mutex_owned(sc->sc_lock)); 4705 4706 error = 0; 4707 if (mode == AUMODE_PLAY) 4708 mixer = sc->sc_pmixer; 4709 else 4710 mixer = sc->sc_rmixer; 4711 4712 mixer->sc = sc; 4713 mixer->mode = mode; 4714 4715 mixer->hwbuf.fmt = *hwfmt; 4716 mixer->volume = 256; 4717 mixer->blktime_d = 1000; 4718 mixer->blktime_n = audio_mixer_calc_blktime(sc, mixer); 4719 sc->sc_blk_ms = mixer->blktime_n; 4720 hwblks = NBLKHW; 4721 4722 mixer->frames_per_block = frame_per_block(mixer, &mixer->hwbuf.fmt); 4723 blksize = frametobyte(&mixer->hwbuf.fmt, mixer->frames_per_block); 4724 if (sc->hw_if->round_blocksize) { 4725 int rounded; 4726 audio_params_t p = format2_to_params(&mixer->hwbuf.fmt); 4727 rounded = sc->hw_if->round_blocksize(sc->hw_hdl, blksize, 4728 mode, &p); 4729 TRACE(2, "round_blocksize %d -> %d", blksize, rounded); 4730 if (rounded != blksize) { 4731 if ((rounded * NBBY) % (mixer->hwbuf.fmt.stride * 4732 mixer->hwbuf.fmt.channels) != 0) { 4733 device_printf(sc->sc_dev, 4734 "blksize not configured %d -> %d\n", 4735 blksize, rounded); 4736 return EINVAL; 4737 } 4738 /* Recalculation */ 4739 blksize = rounded; 4740 mixer->frames_per_block = blksize * NBBY / 4741 (mixer->hwbuf.fmt.stride * 4742 mixer->hwbuf.fmt.channels); 4743 } 4744 } 4745 mixer->blktime_n = mixer->frames_per_block; 4746 mixer->blktime_d = mixer->hwbuf.fmt.sample_rate; 4747 4748 capacity = mixer->frames_per_block * hwblks; 4749 bufsize = frametobyte(&mixer->hwbuf.fmt, capacity); 4750 if (sc->hw_if->round_buffersize) { 4751 size_t rounded; 4752 rounded = sc->hw_if->round_buffersize(sc->hw_hdl, mode, 4753 bufsize); 4754 TRACE(2, "round_buffersize %zd -> %zd", bufsize, rounded); 4755 if (rounded < bufsize) { 4756 /* buffersize needs NBLKHW blocks at least. */ 4757 device_printf(sc->sc_dev, 4758 "buffersize too small: buffersize=%zd blksize=%d\n", 4759 rounded, blksize); 4760 return EINVAL; 4761 } 4762 if (rounded % blksize != 0) { 4763 /* buffersize/blksize constraint mismatch? */ 4764 device_printf(sc->sc_dev, 4765 "buffersize must be multiple of blksize: " 4766 "buffersize=%zu blksize=%d\n", 4767 rounded, blksize); 4768 return EINVAL; 4769 } 4770 if (rounded != bufsize) { 4771 /* Recalcuration */ 4772 bufsize = rounded; 4773 hwblks = bufsize / blksize; 4774 capacity = mixer->frames_per_block * hwblks; 4775 } 4776 } 4777 TRACE(2, "buffersize for %s = %zu", 4778 (mode == AUMODE_PLAY) ? "playback" : "recording", 4779 bufsize); 4780 mixer->hwbuf.capacity = capacity; 4781 4782 /* 4783 * XXX need to release sc_lock for compatibility? 4784 */ 4785 if (sc->hw_if->allocm) { 4786 mixer->hwbuf.mem = sc->hw_if->allocm(sc->hw_hdl, mode, bufsize); 4787 if (mixer->hwbuf.mem == NULL) { 4788 device_printf(sc->sc_dev, "%s: allocm(%zu) failed\n", 4789 __func__, bufsize); 4790 return ENOMEM; 4791 } 4792 } else { 4793 mixer->hwbuf.mem = kern_malloc(bufsize, M_NOWAIT); 4794 if (mixer->hwbuf.mem == NULL) { 4795 device_printf(sc->sc_dev, 4796 "%s: malloc hwbuf(%zu) failed\n", 4797 __func__, bufsize); 4798 return ENOMEM; 4799 } 4800 } 4801 4802 /* From here, audio_mixer_destroy is necessary to exit. */ 4803 if (mode == AUMODE_PLAY) { 4804 cv_init(&mixer->outcv, "audiowr"); 4805 } else { 4806 cv_init(&mixer->outcv, "audiord"); 4807 } 4808 4809 if (mode == AUMODE_PLAY) { 4810 softint_handler = audio_softintr_wr; 4811 } else { 4812 softint_handler = audio_softintr_rd; 4813 } 4814 mixer->sih = softint_establish(SOFTINT_SERIAL | SOFTINT_MPSAFE, 4815 softint_handler, sc); 4816 if (mixer->sih == NULL) { 4817 device_printf(sc->sc_dev, "softint_establish failed\n"); 4818 goto abort; 4819 } 4820 4821 mixer->track_fmt.encoding = AUDIO_ENCODING_SLINEAR_NE; 4822 mixer->track_fmt.precision = AUDIO_INTERNAL_BITS; 4823 mixer->track_fmt.stride = AUDIO_INTERNAL_BITS; 4824 mixer->track_fmt.channels = mixer->hwbuf.fmt.channels; 4825 mixer->track_fmt.sample_rate = mixer->hwbuf.fmt.sample_rate; 4826 4827 if (mixer->hwbuf.fmt.encoding == AUDIO_ENCODING_SLINEAR_OE && 4828 mixer->hwbuf.fmt.precision == AUDIO_INTERNAL_BITS) { 4829 mixer->swap_endian = true; 4830 TRACE(1, "swap_endian"); 4831 } 4832 4833 if (mode == AUMODE_PLAY) { 4834 /* Mixing buffer */ 4835 mixer->mixfmt = mixer->track_fmt; 4836 mixer->mixfmt.precision *= 2; 4837 mixer->mixfmt.stride *= 2; 4838 /* XXX TODO: use some macros? */ 4839 len = mixer->frames_per_block * mixer->mixfmt.channels * 4840 mixer->mixfmt.stride / NBBY; 4841 mixer->mixsample = audio_realloc(mixer->mixsample, len); 4842 if (mixer->mixsample == NULL) { 4843 device_printf(sc->sc_dev, 4844 "%s: malloc mixsample(%d) failed\n", 4845 __func__, len); 4846 error = ENOMEM; 4847 goto abort; 4848 } 4849 } else { 4850 /* No mixing buffer for recording */ 4851 } 4852 4853 if (reg->codec) { 4854 mixer->codec = reg->codec; 4855 mixer->codecarg.context = reg->context; 4856 if (mode == AUMODE_PLAY) { 4857 mixer->codecarg.srcfmt = &mixer->track_fmt; 4858 mixer->codecarg.dstfmt = &mixer->hwbuf.fmt; 4859 } else { 4860 mixer->codecarg.srcfmt = &mixer->hwbuf.fmt; 4861 mixer->codecarg.dstfmt = &mixer->track_fmt; 4862 } 4863 mixer->codecbuf.fmt = mixer->track_fmt; 4864 mixer->codecbuf.capacity = mixer->frames_per_block; 4865 len = auring_bytelen(&mixer->codecbuf); 4866 mixer->codecbuf.mem = audio_realloc(mixer->codecbuf.mem, len); 4867 if (mixer->codecbuf.mem == NULL) { 4868 device_printf(sc->sc_dev, 4869 "%s: malloc codecbuf(%d) failed\n", 4870 __func__, len); 4871 error = ENOMEM; 4872 goto abort; 4873 } 4874 } 4875 4876 /* Succeeded so display it. */ 4877 codecbuf[0] = '\0'; 4878 if (mixer->codec || mixer->swap_endian) { 4879 snprintf(codecbuf, sizeof(codecbuf), " %s %s:%d", 4880 (mode == AUMODE_PLAY) ? "->" : "<-", 4881 audio_encoding_name(mixer->hwbuf.fmt.encoding), 4882 mixer->hwbuf.fmt.precision); 4883 } 4884 blkms = mixer->blktime_n * 1000 / mixer->blktime_d; 4885 aprint_normal_dev(sc->sc_dev, "%s:%d%s %dch %dHz, blk %dms for %s\n", 4886 audio_encoding_name(mixer->track_fmt.encoding), 4887 mixer->track_fmt.precision, 4888 codecbuf, 4889 mixer->track_fmt.channels, 4890 mixer->track_fmt.sample_rate, 4891 blkms, 4892 (mode == AUMODE_PLAY) ? "playback" : "recording"); 4893 4894 return 0; 4895 4896 abort: 4897 audio_mixer_destroy(sc, mixer); 4898 return error; 4899 } 4900 4901 /* 4902 * Releases all resources of 'mixer'. 4903 * Note that it does not release the memory area of 'mixer' itself. 4904 * Must be called with sc_lock held. 4905 */ 4906 static void 4907 audio_mixer_destroy(struct audio_softc *sc, audio_trackmixer_t *mixer) 4908 { 4909 int mode; 4910 4911 KASSERT(mutex_owned(sc->sc_lock)); 4912 4913 mode = mixer->mode; 4914 KASSERT(mode == AUMODE_PLAY || mode == AUMODE_RECORD); 4915 4916 if (mixer->hwbuf.mem != NULL) { 4917 if (sc->hw_if->freem) { 4918 sc->hw_if->freem(sc->hw_hdl, mixer->hwbuf.mem, mode); 4919 } else { 4920 kern_free(mixer->hwbuf.mem); 4921 } 4922 mixer->hwbuf.mem = NULL; 4923 } 4924 4925 audio_free(mixer->codecbuf.mem); 4926 audio_free(mixer->mixsample); 4927 4928 cv_destroy(&mixer->outcv); 4929 4930 if (mixer->sih) { 4931 softint_disestablish(mixer->sih); 4932 mixer->sih = NULL; 4933 } 4934 } 4935 4936 /* 4937 * Starts playback mixer. 4938 * Must be called only if sc_pbusy is false. 4939 * Must be called with sc_lock held. 4940 * Must not be called from the interrupt context. 4941 */ 4942 static void 4943 audio_pmixer_start(struct audio_softc *sc, bool force) 4944 { 4945 audio_trackmixer_t *mixer; 4946 int minimum; 4947 4948 KASSERT(mutex_owned(sc->sc_lock)); 4949 KASSERT(sc->sc_pbusy == false); 4950 4951 mutex_enter(sc->sc_intr_lock); 4952 4953 mixer = sc->sc_pmixer; 4954 TRACE(2, "%smixseq=%d hwseq=%d hwbuf=%d/%d/%d%s", 4955 (audiodebug >= 3) ? "begin " : "", 4956 (int)mixer->mixseq, (int)mixer->hwseq, 4957 mixer->hwbuf.head, mixer->hwbuf.used, mixer->hwbuf.capacity, 4958 force ? " force" : ""); 4959 4960 /* Need two blocks to start normally. */ 4961 minimum = (force) ? 1 : 2; 4962 while (mixer->hwbuf.used < mixer->frames_per_block * minimum) { 4963 audio_pmixer_process(sc); 4964 } 4965 4966 /* Start output */ 4967 audio_pmixer_output(sc); 4968 sc->sc_pbusy = true; 4969 4970 TRACE(3, "end mixseq=%d hwseq=%d hwbuf=%d/%d/%d", 4971 (int)mixer->mixseq, (int)mixer->hwseq, 4972 mixer->hwbuf.head, mixer->hwbuf.used, mixer->hwbuf.capacity); 4973 4974 mutex_exit(sc->sc_intr_lock); 4975 } 4976 4977 /* 4978 * When playing back with MD filter: 4979 * 4980 * track track ... 4981 * v v 4982 * + mix (with aint2_t) 4983 * | master volume (with aint2_t) 4984 * v 4985 * mixsample [::::] wide-int 1 block (ring) buffer 4986 * | 4987 * | convert aint2_t -> aint_t 4988 * v 4989 * codecbuf [....] 1 block (ring) buffer 4990 * | 4991 * | convert to hw format 4992 * v 4993 * hwbuf [............] NBLKHW blocks ring buffer 4994 * 4995 * When playing back without MD filter: 4996 * 4997 * mixsample [::::] wide-int 1 block (ring) buffer 4998 * | 4999 * | convert aint2_t -> aint_t 5000 * | (with byte swap if necessary) 5001 * v 5002 * hwbuf [............] NBLKHW blocks ring buffer 5003 * 5004 * mixsample: slinear_NE, wide internal precision, HW ch, HW freq. 5005 * codecbuf: slinear_NE, internal precision, HW ch, HW freq. 5006 * hwbuf: HW encoding, HW precision, HW ch, HW freq. 5007 */ 5008 5009 /* 5010 * Performs track mixing and converts it to hwbuf. 5011 * Note that this function doesn't transfer hwbuf to hardware. 5012 * Must be called with sc_intr_lock held. 5013 */ 5014 static void 5015 audio_pmixer_process(struct audio_softc *sc) 5016 { 5017 audio_trackmixer_t *mixer; 5018 audio_file_t *f; 5019 int frame_count; 5020 int sample_count; 5021 int mixed; 5022 int i; 5023 aint2_t *m; 5024 aint_t *h; 5025 5026 mixer = sc->sc_pmixer; 5027 5028 frame_count = mixer->frames_per_block; 5029 KASSERT(auring_get_contig_free(&mixer->hwbuf) >= frame_count); 5030 sample_count = frame_count * mixer->mixfmt.channels; 5031 5032 mixer->mixseq++; 5033 5034 /* Mix all tracks */ 5035 mixed = 0; 5036 SLIST_FOREACH(f, &sc->sc_files, entry) { 5037 audio_track_t *track = f->ptrack; 5038 5039 if (track == NULL) 5040 continue; 5041 5042 if (track->is_pause) { 5043 TRACET(4, track, "skip; paused"); 5044 continue; 5045 } 5046 5047 /* Skip if the track is used by process context. */ 5048 if (audio_track_lock_tryenter(track) == false) { 5049 TRACET(4, track, "skip; in use"); 5050 continue; 5051 } 5052 5053 /* Emulate mmap'ped track */ 5054 if (track->mmapped) { 5055 auring_push(&track->usrbuf, track->usrbuf_blksize); 5056 TRACET(4, track, "mmap; usr=%d/%d/C%d", 5057 track->usrbuf.head, 5058 track->usrbuf.used, 5059 track->usrbuf.capacity); 5060 } 5061 5062 if (track->outbuf.used < mixer->frames_per_block && 5063 track->usrbuf.used > 0) { 5064 TRACET(4, track, "process"); 5065 audio_track_play(track); 5066 } 5067 5068 if (track->outbuf.used > 0) { 5069 mixed = audio_pmixer_mix_track(mixer, track, mixed); 5070 } else { 5071 TRACET(4, track, "skip; empty"); 5072 } 5073 5074 audio_track_lock_exit(track); 5075 } 5076 5077 if (mixed == 0) { 5078 /* Silence */ 5079 memset(mixer->mixsample, 0, 5080 frametobyte(&mixer->mixfmt, frame_count)); 5081 } else { 5082 aint2_t ovf_plus; 5083 aint2_t ovf_minus; 5084 int vol; 5085 5086 /* Overflow detection */ 5087 ovf_plus = AINT_T_MAX; 5088 ovf_minus = AINT_T_MIN; 5089 m = mixer->mixsample; 5090 for (i = 0; i < sample_count; i++) { 5091 aint2_t val; 5092 5093 val = *m++; 5094 if (val > ovf_plus) 5095 ovf_plus = val; 5096 else if (val < ovf_minus) 5097 ovf_minus = val; 5098 } 5099 5100 /* Master Volume Auto Adjust */ 5101 vol = mixer->volume; 5102 if (ovf_plus > (aint2_t)AINT_T_MAX 5103 || ovf_minus < (aint2_t)AINT_T_MIN) { 5104 aint2_t ovf; 5105 int vol2; 5106 5107 /* XXX TODO: Check AINT2_T_MIN ? */ 5108 ovf = ovf_plus; 5109 if (ovf < -ovf_minus) 5110 ovf = -ovf_minus; 5111 5112 /* Turn down the volume if overflow occured. */ 5113 vol2 = (int)((aint2_t)AINT_T_MAX * 256 / ovf); 5114 if (vol2 < vol) 5115 vol = vol2; 5116 5117 if (vol < mixer->volume) { 5118 /* Turn down gradually to 128. */ 5119 if (mixer->volume > 128) { 5120 mixer->volume = 5121 (mixer->volume * 95) / 100; 5122 device_printf(sc->sc_dev, 5123 "auto volume adjust: volume %d\n", 5124 mixer->volume); 5125 } 5126 } 5127 } 5128 5129 /* Apply Master Volume. */ 5130 if (vol != 256) { 5131 m = mixer->mixsample; 5132 for (i = 0; i < sample_count; i++) { 5133 #if defined(AUDIO_USE_C_IMPLEMENTATION_DEFINED_BEHAVIOR) && defined(__GNUC__) 5134 *m = *m * vol >> 8; 5135 #else 5136 *m = *m * vol / 256; 5137 #endif 5138 m++; 5139 } 5140 } 5141 } 5142 5143 /* 5144 * The rest is the hardware part. 5145 */ 5146 5147 if (mixer->codec) { 5148 h = auring_tailptr_aint(&mixer->codecbuf); 5149 } else { 5150 h = auring_tailptr_aint(&mixer->hwbuf); 5151 } 5152 5153 m = mixer->mixsample; 5154 if (mixer->swap_endian) { 5155 for (i = 0; i < sample_count; i++) { 5156 *h++ = bswap16(*m++); 5157 } 5158 } else { 5159 for (i = 0; i < sample_count; i++) { 5160 *h++ = *m++; 5161 } 5162 } 5163 5164 /* Hardware driver's codec */ 5165 if (mixer->codec) { 5166 auring_push(&mixer->codecbuf, frame_count); 5167 mixer->codecarg.src = auring_headptr(&mixer->codecbuf); 5168 mixer->codecarg.dst = auring_tailptr(&mixer->hwbuf); 5169 mixer->codecarg.count = frame_count; 5170 mixer->codec(&mixer->codecarg); 5171 auring_take(&mixer->codecbuf, mixer->codecarg.count); 5172 } 5173 5174 auring_push(&mixer->hwbuf, frame_count); 5175 5176 TRACE(4, "done mixseq=%d hwbuf=%d/%d/%d%s", 5177 (int)mixer->mixseq, 5178 mixer->hwbuf.head, mixer->hwbuf.used, mixer->hwbuf.capacity, 5179 (mixed == 0) ? " silent" : ""); 5180 } 5181 5182 /* 5183 * Mix one track. 5184 * 'mixed' specifies the number of tracks mixed so far. 5185 * It returns the number of tracks mixed. In other words, it returns 5186 * mixed + 1 if this track is mixed. 5187 */ 5188 static int 5189 audio_pmixer_mix_track(audio_trackmixer_t *mixer, audio_track_t *track, 5190 int mixed) 5191 { 5192 int count; 5193 int sample_count; 5194 int remain; 5195 int i; 5196 const aint_t *s; 5197 aint2_t *d; 5198 5199 /* XXX TODO: Is this necessary for now? */ 5200 if (mixer->mixseq < track->seq) 5201 return mixed; 5202 5203 count = auring_get_contig_used(&track->outbuf); 5204 count = uimin(count, mixer->frames_per_block); 5205 5206 s = auring_headptr_aint(&track->outbuf); 5207 d = mixer->mixsample; 5208 5209 /* 5210 * Apply track volume with double-sized integer and perform 5211 * additive synthesis. 5212 * 5213 * XXX If you limit the track volume to 1.0 or less (<= 256), 5214 * it would be better to do this in the track conversion stage 5215 * rather than here. However, if you accept the volume to 5216 * be greater than 1.0 (> 256), it's better to do it here. 5217 * Because the operation here is done by double-sized integer. 5218 */ 5219 sample_count = count * mixer->mixfmt.channels; 5220 if (mixed == 0) { 5221 /* If this is the first track, assignment can be used. */ 5222 #if defined(AUDIO_SUPPORT_TRACK_VOLUME) 5223 if (track->volume != 256) { 5224 for (i = 0; i < sample_count; i++) { 5225 #if defined(AUDIO_USE_C_IMPLEMENTATION_DEFINED_BEHAVIOR) && defined(__GNUC__) 5226 *d++ = ((aint2_t)*s++) * track->volume >> 8; 5227 #else 5228 *d++ = ((aint2_t)*s++) * track->volume / 256; 5229 #endif 5230 } 5231 } else 5232 #endif 5233 { 5234 for (i = 0; i < sample_count; i++) { 5235 *d++ = ((aint2_t)*s++); 5236 } 5237 } 5238 } else { 5239 /* If this is the second or later, add it. */ 5240 #if defined(AUDIO_SUPPORT_TRACK_VOLUME) 5241 if (track->volume != 256) { 5242 for (i = 0; i < sample_count; i++) { 5243 #if defined(AUDIO_USE_C_IMPLEMENTATION_DEFINED_BEHAVIOR) && defined(__GNUC__) 5244 *d++ += ((aint2_t)*s++) * track->volume >> 8; 5245 #else 5246 *d++ += ((aint2_t)*s++) * track->volume / 256; 5247 #endif 5248 } 5249 } else 5250 #endif 5251 { 5252 for (i = 0; i < sample_count; i++) { 5253 *d++ += ((aint2_t)*s++); 5254 } 5255 } 5256 } 5257 5258 auring_take(&track->outbuf, count); 5259 /* 5260 * The counters have to align block even if outbuf is less than 5261 * one block. XXX Is this still necessary? 5262 */ 5263 remain = mixer->frames_per_block - count; 5264 if (__predict_false(remain != 0)) { 5265 auring_push(&track->outbuf, remain); 5266 auring_take(&track->outbuf, remain); 5267 } 5268 5269 /* 5270 * Update track sequence. 5271 * mixseq has previous value yet at this point. 5272 */ 5273 track->seq = mixer->mixseq + 1; 5274 5275 return mixed + 1; 5276 } 5277 5278 /* 5279 * Output one block from hwbuf to HW. 5280 * Must be called with sc_intr_lock held. 5281 */ 5282 static void 5283 audio_pmixer_output(struct audio_softc *sc) 5284 { 5285 audio_trackmixer_t *mixer; 5286 audio_params_t params; 5287 void *start; 5288 void *end; 5289 int blksize; 5290 int error; 5291 5292 mixer = sc->sc_pmixer; 5293 TRACE(4, "pbusy=%d hwbuf=%d/%d/%d", 5294 sc->sc_pbusy, 5295 mixer->hwbuf.head, mixer->hwbuf.used, mixer->hwbuf.capacity); 5296 KASSERT(mixer->hwbuf.used >= mixer->frames_per_block); 5297 5298 blksize = frametobyte(&mixer->hwbuf.fmt, mixer->frames_per_block); 5299 5300 if (sc->hw_if->trigger_output) { 5301 /* trigger (at once) */ 5302 if (!sc->sc_pbusy) { 5303 start = mixer->hwbuf.mem; 5304 end = (uint8_t *)start + auring_bytelen(&mixer->hwbuf); 5305 params = format2_to_params(&mixer->hwbuf.fmt); 5306 5307 error = sc->hw_if->trigger_output(sc->hw_hdl, 5308 start, end, blksize, audio_pintr, sc, ¶ms); 5309 if (error) { 5310 device_printf(sc->sc_dev, 5311 "trigger_output failed with %d", error); 5312 return; 5313 } 5314 } 5315 } else { 5316 /* start (everytime) */ 5317 start = auring_headptr(&mixer->hwbuf); 5318 5319 error = sc->hw_if->start_output(sc->hw_hdl, 5320 start, blksize, audio_pintr, sc); 5321 if (error) { 5322 device_printf(sc->sc_dev, 5323 "start_output failed with %d", error); 5324 return; 5325 } 5326 } 5327 } 5328 5329 /* 5330 * This is an interrupt handler for playback. 5331 * It is called with sc_intr_lock held. 5332 * 5333 * It is usually called from hardware interrupt. However, note that 5334 * for some drivers (e.g. uaudio) it is called from software interrupt. 5335 */ 5336 static void 5337 audio_pintr(void *arg) 5338 { 5339 struct audio_softc *sc; 5340 audio_trackmixer_t *mixer; 5341 5342 sc = arg; 5343 KASSERT(mutex_owned(sc->sc_intr_lock)); 5344 5345 if (sc->sc_dying) 5346 return; 5347 #if defined(DIAGNOSTIC) 5348 if (sc->sc_pbusy == false) { 5349 device_printf(sc->sc_dev, "stray interrupt\n"); 5350 return; 5351 } 5352 #endif 5353 5354 mixer = sc->sc_pmixer; 5355 mixer->hw_complete_counter += mixer->frames_per_block; 5356 mixer->hwseq++; 5357 5358 auring_take(&mixer->hwbuf, mixer->frames_per_block); 5359 5360 TRACE(4, 5361 "HW_INT ++hwseq=%" PRIu64 " cmplcnt=%" PRIu64 " hwbuf=%d/%d/%d", 5362 mixer->hwseq, mixer->hw_complete_counter, 5363 mixer->hwbuf.head, mixer->hwbuf.used, mixer->hwbuf.capacity); 5364 5365 #if !defined(_KERNEL) 5366 /* This is a debug code for userland test. */ 5367 return; 5368 #endif 5369 5370 #if defined(AUDIO_HW_SINGLE_BUFFER) 5371 /* 5372 * Create a new block here and output it immediately. 5373 * It makes a latency lower but needs machine power. 5374 */ 5375 audio_pmixer_process(sc); 5376 audio_pmixer_output(sc); 5377 #else 5378 /* 5379 * It is called when block N output is done. 5380 * Output immediately block N+1 created by the last interrupt. 5381 * And then create block N+2 for the next interrupt. 5382 * This method makes playback robust even on slower machines. 5383 * Instead the latency is increased by one block. 5384 */ 5385 5386 /* At first, output ready block. */ 5387 if (mixer->hwbuf.used >= mixer->frames_per_block) { 5388 audio_pmixer_output(sc); 5389 } 5390 5391 bool later = false; 5392 5393 if (mixer->hwbuf.used < mixer->frames_per_block) { 5394 later = true; 5395 } 5396 5397 /* Then, process next block. */ 5398 audio_pmixer_process(sc); 5399 5400 if (later) { 5401 audio_pmixer_output(sc); 5402 } 5403 #endif 5404 5405 /* 5406 * When this interrupt is the real hardware interrupt, disabling 5407 * preemption here is not necessary. But some drivers (e.g. uaudio) 5408 * emulate it by software interrupt, so kpreempt_disable is necessary. 5409 */ 5410 kpreempt_disable(); 5411 softint_schedule(mixer->sih); 5412 kpreempt_enable(); 5413 } 5414 5415 /* 5416 * Starts record mixer. 5417 * Must be called only if sc_rbusy is false. 5418 * Must be called with sc_lock held. 5419 * Must not be called from the interrupt context. 5420 */ 5421 static void 5422 audio_rmixer_start(struct audio_softc *sc) 5423 { 5424 5425 KASSERT(mutex_owned(sc->sc_lock)); 5426 KASSERT(sc->sc_rbusy == false); 5427 5428 mutex_enter(sc->sc_intr_lock); 5429 5430 TRACE(2, "%s", (audiodebug >= 3) ? "begin" : ""); 5431 audio_rmixer_input(sc); 5432 sc->sc_rbusy = true; 5433 TRACE(3, "end"); 5434 5435 mutex_exit(sc->sc_intr_lock); 5436 } 5437 5438 /* 5439 * When recording with MD filter: 5440 * 5441 * hwbuf [............] NBLKHW blocks ring buffer 5442 * | 5443 * | convert from hw format 5444 * v 5445 * codecbuf [....] 1 block (ring) buffer 5446 * | | 5447 * v v 5448 * track track ... 5449 * 5450 * When recording without MD filter: 5451 * 5452 * hwbuf [............] NBLKHW blocks ring buffer 5453 * | | 5454 * v v 5455 * track track ... 5456 * 5457 * hwbuf: HW encoding, HW precision, HW ch, HW freq. 5458 * codecbuf: slinear_NE, internal precision, HW ch, HW freq. 5459 */ 5460 5461 /* 5462 * Distribute a recorded block to all recording tracks. 5463 */ 5464 static void 5465 audio_rmixer_process(struct audio_softc *sc) 5466 { 5467 audio_trackmixer_t *mixer; 5468 audio_ring_t *mixersrc; 5469 audio_file_t *f; 5470 aint_t *p; 5471 int count; 5472 int bytes; 5473 int i; 5474 5475 mixer = sc->sc_rmixer; 5476 5477 /* 5478 * count is the number of frames to be retrieved this time. 5479 * count should be one block. 5480 */ 5481 count = auring_get_contig_used(&mixer->hwbuf); 5482 count = uimin(count, mixer->frames_per_block); 5483 if (count <= 0) { 5484 TRACE(4, "count %d: too short", count); 5485 return; 5486 } 5487 bytes = frametobyte(&mixer->track_fmt, count); 5488 5489 /* Hardware driver's codec */ 5490 if (mixer->codec) { 5491 mixer->codecarg.src = auring_headptr(&mixer->hwbuf); 5492 mixer->codecarg.dst = auring_tailptr(&mixer->codecbuf); 5493 mixer->codecarg.count = count; 5494 mixer->codec(&mixer->codecarg); 5495 auring_take(&mixer->hwbuf, mixer->codecarg.count); 5496 auring_push(&mixer->codecbuf, mixer->codecarg.count); 5497 mixersrc = &mixer->codecbuf; 5498 } else { 5499 mixersrc = &mixer->hwbuf; 5500 } 5501 5502 if (mixer->swap_endian) { 5503 /* inplace conversion */ 5504 p = auring_headptr_aint(mixersrc); 5505 for (i = 0; i < count * mixer->track_fmt.channels; i++, p++) { 5506 *p = bswap16(*p); 5507 } 5508 } 5509 5510 /* Distribute to all tracks. */ 5511 SLIST_FOREACH(f, &sc->sc_files, entry) { 5512 audio_track_t *track = f->rtrack; 5513 audio_ring_t *input; 5514 5515 if (track == NULL) 5516 continue; 5517 5518 if (track->is_pause) { 5519 TRACET(4, track, "skip; paused"); 5520 continue; 5521 } 5522 5523 if (audio_track_lock_tryenter(track) == false) { 5524 TRACET(4, track, "skip; in use"); 5525 continue; 5526 } 5527 5528 /* If the track buffer is full, discard the oldest one? */ 5529 input = track->input; 5530 if (input->capacity - input->used < mixer->frames_per_block) { 5531 int drops = mixer->frames_per_block - 5532 (input->capacity - input->used); 5533 track->dropframes += drops; 5534 TRACET(4, track, "drop %d frames: inp=%d/%d/%d", 5535 drops, 5536 input->head, input->used, input->capacity); 5537 auring_take(input, drops); 5538 } 5539 KASSERT(input->used % mixer->frames_per_block == 0); 5540 5541 memcpy(auring_tailptr_aint(input), 5542 auring_headptr_aint(mixersrc), 5543 bytes); 5544 auring_push(input, count); 5545 5546 /* XXX sequence counter? */ 5547 5548 audio_track_lock_exit(track); 5549 } 5550 5551 auring_take(mixersrc, count); 5552 } 5553 5554 /* 5555 * Input one block from HW to hwbuf. 5556 * Must be called with sc_intr_lock held. 5557 */ 5558 static void 5559 audio_rmixer_input(struct audio_softc *sc) 5560 { 5561 audio_trackmixer_t *mixer; 5562 audio_params_t params; 5563 void *start; 5564 void *end; 5565 int blksize; 5566 int error; 5567 5568 mixer = sc->sc_rmixer; 5569 blksize = frametobyte(&mixer->hwbuf.fmt, mixer->frames_per_block); 5570 5571 if (sc->hw_if->trigger_input) { 5572 /* trigger (at once) */ 5573 if (!sc->sc_rbusy) { 5574 start = mixer->hwbuf.mem; 5575 end = (uint8_t *)start + auring_bytelen(&mixer->hwbuf); 5576 params = format2_to_params(&mixer->hwbuf.fmt); 5577 5578 error = sc->hw_if->trigger_input(sc->hw_hdl, 5579 start, end, blksize, audio_rintr, sc, ¶ms); 5580 if (error) { 5581 device_printf(sc->sc_dev, 5582 "trigger_input failed with %d", error); 5583 return; 5584 } 5585 } 5586 } else { 5587 /* start (everytime) */ 5588 start = auring_tailptr(&mixer->hwbuf); 5589 5590 error = sc->hw_if->start_input(sc->hw_hdl, 5591 start, blksize, audio_rintr, sc); 5592 if (error) { 5593 device_printf(sc->sc_dev, 5594 "start_input failed with %d", error); 5595 return; 5596 } 5597 } 5598 } 5599 5600 /* 5601 * This is an interrupt handler for recording. 5602 * It is called with sc_intr_lock. 5603 * 5604 * It is usually called from hardware interrupt. However, note that 5605 * for some drivers (e.g. uaudio) it is called from software interrupt. 5606 */ 5607 static void 5608 audio_rintr(void *arg) 5609 { 5610 struct audio_softc *sc; 5611 audio_trackmixer_t *mixer; 5612 5613 sc = arg; 5614 KASSERT(mutex_owned(sc->sc_intr_lock)); 5615 5616 if (sc->sc_dying) 5617 return; 5618 #if defined(DIAGNOSTIC) 5619 if (sc->sc_rbusy == false) { 5620 device_printf(sc->sc_dev, "stray interrupt\n"); 5621 return; 5622 } 5623 #endif 5624 5625 mixer = sc->sc_rmixer; 5626 mixer->hw_complete_counter += mixer->frames_per_block; 5627 mixer->hwseq++; 5628 5629 auring_push(&mixer->hwbuf, mixer->frames_per_block); 5630 5631 TRACE(4, 5632 "HW_INT ++hwseq=%" PRIu64 " cmplcnt=%" PRIu64 " hwbuf=%d/%d/%d", 5633 mixer->hwseq, mixer->hw_complete_counter, 5634 mixer->hwbuf.head, mixer->hwbuf.used, mixer->hwbuf.capacity); 5635 5636 /* Distrubute recorded block */ 5637 audio_rmixer_process(sc); 5638 5639 /* Request next block */ 5640 audio_rmixer_input(sc); 5641 5642 /* 5643 * When this interrupt is the real hardware interrupt, disabling 5644 * preemption here is not necessary. But some drivers (e.g. uaudio) 5645 * emulate it by software interrupt, so kpreempt_disable is necessary. 5646 */ 5647 kpreempt_disable(); 5648 softint_schedule(mixer->sih); 5649 kpreempt_enable(); 5650 } 5651 5652 /* 5653 * Halts playback mixer. 5654 * This function also clears related parameters, so call this function 5655 * instead of calling halt_output directly. 5656 * Must be called only if sc_pbusy is true. 5657 * Must be called with sc_lock && sc_exlock held. 5658 */ 5659 static int 5660 audio_pmixer_halt(struct audio_softc *sc) 5661 { 5662 int error; 5663 5664 TRACE(2, ""); 5665 KASSERT(mutex_owned(sc->sc_lock)); 5666 KASSERT(sc->sc_exlock); 5667 5668 mutex_enter(sc->sc_intr_lock); 5669 error = sc->hw_if->halt_output(sc->hw_hdl); 5670 mutex_exit(sc->sc_intr_lock); 5671 5672 /* Halts anyway even if some error has occurred. */ 5673 sc->sc_pbusy = false; 5674 sc->sc_pmixer->hwbuf.head = 0; 5675 sc->sc_pmixer->hwbuf.used = 0; 5676 sc->sc_pmixer->mixseq = 0; 5677 sc->sc_pmixer->hwseq = 0; 5678 5679 return error; 5680 } 5681 5682 /* 5683 * Halts recording mixer. 5684 * This function also clears related parameters, so call this function 5685 * instead of calling halt_input directly. 5686 * Must be called only if sc_rbusy is true. 5687 * Must be called with sc_lock && sc_exlock held. 5688 */ 5689 static int 5690 audio_rmixer_halt(struct audio_softc *sc) 5691 { 5692 int error; 5693 5694 TRACE(2, ""); 5695 KASSERT(mutex_owned(sc->sc_lock)); 5696 KASSERT(sc->sc_exlock); 5697 5698 mutex_enter(sc->sc_intr_lock); 5699 error = sc->hw_if->halt_input(sc->hw_hdl); 5700 mutex_exit(sc->sc_intr_lock); 5701 5702 /* Halts anyway even if some error has occurred. */ 5703 sc->sc_rbusy = false; 5704 sc->sc_rmixer->hwbuf.head = 0; 5705 sc->sc_rmixer->hwbuf.used = 0; 5706 sc->sc_rmixer->mixseq = 0; 5707 sc->sc_rmixer->hwseq = 0; 5708 5709 return error; 5710 } 5711 5712 /* 5713 * Flush this track. 5714 * Halts all operations, clears all buffers, reset error counters. 5715 * XXX I'm not sure... 5716 */ 5717 static void 5718 audio_track_clear(struct audio_softc *sc, audio_track_t *track) 5719 { 5720 5721 KASSERT(track); 5722 TRACET(3, track, "clear"); 5723 5724 audio_track_lock_enter(track); 5725 5726 track->usrbuf.used = 0; 5727 /* Clear all internal parameters. */ 5728 if (track->codec.filter) { 5729 track->codec.srcbuf.used = 0; 5730 track->codec.srcbuf.head = 0; 5731 } 5732 if (track->chvol.filter) { 5733 track->chvol.srcbuf.used = 0; 5734 track->chvol.srcbuf.head = 0; 5735 } 5736 if (track->chmix.filter) { 5737 track->chmix.srcbuf.used = 0; 5738 track->chmix.srcbuf.head = 0; 5739 } 5740 if (track->freq.filter) { 5741 track->freq.srcbuf.used = 0; 5742 track->freq.srcbuf.head = 0; 5743 if (track->freq_step < 65536) 5744 track->freq_current = 65536; 5745 else 5746 track->freq_current = 0; 5747 memset(track->freq_prev, 0, sizeof(track->freq_prev)); 5748 memset(track->freq_curr, 0, sizeof(track->freq_curr)); 5749 } 5750 /* Clear buffer, then operation halts naturally. */ 5751 track->outbuf.used = 0; 5752 5753 /* Clear counters. */ 5754 track->dropframes = 0; 5755 5756 audio_track_lock_exit(track); 5757 } 5758 5759 /* 5760 * Drain the track. 5761 * track must be present and for playback. 5762 * If successful, it returns 0. Otherwise returns errno. 5763 * Must be called with sc_lock held. 5764 */ 5765 static int 5766 audio_track_drain(struct audio_softc *sc, audio_track_t *track) 5767 { 5768 audio_trackmixer_t *mixer; 5769 int done; 5770 int error; 5771 5772 KASSERT(track); 5773 TRACET(3, track, "start"); 5774 mixer = track->mixer; 5775 KASSERT(mutex_owned(sc->sc_lock)); 5776 5777 /* Ignore them if pause. */ 5778 if (track->is_pause) { 5779 TRACET(3, track, "pause -> clear"); 5780 track->pstate = AUDIO_STATE_CLEAR; 5781 } 5782 /* Terminate early here if there is no data in the track. */ 5783 if (track->pstate == AUDIO_STATE_CLEAR) { 5784 TRACET(3, track, "no need to drain"); 5785 return 0; 5786 } 5787 track->pstate = AUDIO_STATE_DRAINING; 5788 5789 for (;;) { 5790 /* I want to display it bofore condition evaluation. */ 5791 TRACET(3, track, "pid=%d.%d trkseq=%d hwseq=%d out=%d/%d/%d", 5792 (int)curproc->p_pid, (int)curlwp->l_lid, 5793 (int)track->seq, (int)mixer->hwseq, 5794 track->outbuf.head, track->outbuf.used, 5795 track->outbuf.capacity); 5796 5797 /* Condition to terminate */ 5798 audio_track_lock_enter(track); 5799 done = (track->usrbuf.used < frametobyte(&track->inputfmt, 1) && 5800 track->outbuf.used == 0 && 5801 track->seq <= mixer->hwseq); 5802 audio_track_lock_exit(track); 5803 if (done) 5804 break; 5805 5806 TRACET(3, track, "sleep"); 5807 error = audio_track_waitio(sc, track); 5808 if (error) 5809 return error; 5810 5811 /* XXX call audio_track_play here ? */ 5812 } 5813 5814 track->pstate = AUDIO_STATE_CLEAR; 5815 TRACET(3, track, "done trk_inp=%d trk_out=%d", 5816 (int)track->inputcounter, (int)track->outputcounter); 5817 return 0; 5818 } 5819 5820 /* 5821 * This is software interrupt handler for record. 5822 * It is called from recording hardware interrupt everytime. 5823 * It does: 5824 * - Deliver SIGIO for all async processes. 5825 * - Notify to audio_read() that data has arrived. 5826 * - selnotify() for select/poll-ing processes. 5827 */ 5828 /* 5829 * XXX If a process issues FIOASYNC between hardware interrupt and 5830 * software interrupt, (stray) SIGIO will be sent to the process 5831 * despite the fact that it has not receive recorded data yet. 5832 */ 5833 static void 5834 audio_softintr_rd(void *cookie) 5835 { 5836 struct audio_softc *sc = cookie; 5837 audio_file_t *f; 5838 proc_t *p; 5839 pid_t pid; 5840 5841 mutex_enter(sc->sc_lock); 5842 mutex_enter(sc->sc_intr_lock); 5843 5844 SLIST_FOREACH(f, &sc->sc_files, entry) { 5845 audio_track_t *track = f->rtrack; 5846 5847 if (track == NULL) 5848 continue; 5849 5850 TRACET(4, track, "broadcast; inp=%d/%d/%d", 5851 track->input->head, 5852 track->input->used, 5853 track->input->capacity); 5854 5855 pid = f->async_audio; 5856 if (pid != 0) { 5857 TRACEF(4, f, "sending SIGIO %d", pid); 5858 mutex_enter(proc_lock); 5859 if ((p = proc_find(pid)) != NULL) 5860 psignal(p, SIGIO); 5861 mutex_exit(proc_lock); 5862 } 5863 } 5864 mutex_exit(sc->sc_intr_lock); 5865 5866 /* Notify that data has arrived. */ 5867 selnotify(&sc->sc_rsel, 0, NOTE_SUBMIT); 5868 KNOTE(&sc->sc_rsel.sel_klist, 0); 5869 cv_broadcast(&sc->sc_rmixer->outcv); 5870 5871 mutex_exit(sc->sc_lock); 5872 } 5873 5874 /* 5875 * This is software interrupt handler for playback. 5876 * It is called from playback hardware interrupt everytime. 5877 * It does: 5878 * - Deliver SIGIO for all async and writable (used < lowat) processes. 5879 * - Notify to audio_write() that outbuf block available. 5880 * - selnotify() for select/poll-ing processes if there are any writable 5881 * (used < lowat) processes. Checking each descriptor will be done by 5882 * filt_audiowrite_event(). 5883 */ 5884 static void 5885 audio_softintr_wr(void *cookie) 5886 { 5887 struct audio_softc *sc = cookie; 5888 audio_file_t *f; 5889 bool found; 5890 proc_t *p; 5891 pid_t pid; 5892 5893 TRACE(4, "called"); 5894 found = false; 5895 5896 mutex_enter(sc->sc_lock); 5897 mutex_enter(sc->sc_intr_lock); 5898 5899 SLIST_FOREACH(f, &sc->sc_files, entry) { 5900 audio_track_t *track = f->ptrack; 5901 5902 if (track == NULL) 5903 continue; 5904 5905 TRACET(4, track, "broadcast; trseq=%d out=%d/%d/%d", 5906 (int)track->seq, 5907 track->outbuf.head, 5908 track->outbuf.used, 5909 track->outbuf.capacity); 5910 5911 /* 5912 * Send a signal if the process is async mode and 5913 * used is lower than lowat. 5914 */ 5915 if (track->usrbuf.used <= track->usrbuf_usedlow && 5916 !track->is_pause) { 5917 found = true; 5918 pid = f->async_audio; 5919 if (pid != 0) { 5920 TRACEF(4, f, "sending SIGIO %d", pid); 5921 mutex_enter(proc_lock); 5922 if ((p = proc_find(pid)) != NULL) 5923 psignal(p, SIGIO); 5924 mutex_exit(proc_lock); 5925 } 5926 } 5927 } 5928 mutex_exit(sc->sc_intr_lock); 5929 5930 /* 5931 * Notify for select/poll when someone become writable. 5932 * It needs sc_lock (and not sc_intr_lock). 5933 */ 5934 if (found) { 5935 TRACE(4, "selnotify"); 5936 selnotify(&sc->sc_wsel, 0, NOTE_SUBMIT); 5937 KNOTE(&sc->sc_wsel.sel_klist, 0); 5938 } 5939 5940 /* Notify to audio_write() that outbuf available. */ 5941 cv_broadcast(&sc->sc_pmixer->outcv); 5942 5943 mutex_exit(sc->sc_lock); 5944 } 5945 5946 /* 5947 * Check (and convert) the format *p came from userland. 5948 * If successful, it writes back the converted format to *p if necessary 5949 * and returns 0. Otherwise returns errno (*p may change even this case). 5950 */ 5951 static int 5952 audio_check_params(audio_format2_t *p) 5953 { 5954 5955 /* Convert obsoleted AUDIO_ENCODING_PCM* */ 5956 /* XXX Is this conversion right? */ 5957 if (p->encoding == AUDIO_ENCODING_PCM16) { 5958 if (p->precision == 8) 5959 p->encoding = AUDIO_ENCODING_ULINEAR; 5960 else 5961 p->encoding = AUDIO_ENCODING_SLINEAR; 5962 } else if (p->encoding == AUDIO_ENCODING_PCM8) { 5963 if (p->precision == 8) 5964 p->encoding = AUDIO_ENCODING_ULINEAR; 5965 else 5966 return EINVAL; 5967 } 5968 5969 /* 5970 * Convert obsoleted AUDIO_ENCODING_[SU]LINEAR without endianness 5971 * suffix. 5972 */ 5973 if (p->encoding == AUDIO_ENCODING_SLINEAR) 5974 p->encoding = AUDIO_ENCODING_SLINEAR_NE; 5975 if (p->encoding == AUDIO_ENCODING_ULINEAR) 5976 p->encoding = AUDIO_ENCODING_ULINEAR_NE; 5977 5978 switch (p->encoding) { 5979 case AUDIO_ENCODING_ULAW: 5980 case AUDIO_ENCODING_ALAW: 5981 if (p->precision != 8) 5982 return EINVAL; 5983 break; 5984 case AUDIO_ENCODING_ADPCM: 5985 if (p->precision != 4 && p->precision != 8) 5986 return EINVAL; 5987 break; 5988 case AUDIO_ENCODING_SLINEAR_LE: 5989 case AUDIO_ENCODING_SLINEAR_BE: 5990 case AUDIO_ENCODING_ULINEAR_LE: 5991 case AUDIO_ENCODING_ULINEAR_BE: 5992 if (p->precision != 8 && p->precision != 16 && 5993 p->precision != 24 && p->precision != 32) 5994 return EINVAL; 5995 5996 /* 8bit format does not have endianness. */ 5997 if (p->precision == 8) { 5998 if (p->encoding == AUDIO_ENCODING_SLINEAR_OE) 5999 p->encoding = AUDIO_ENCODING_SLINEAR_NE; 6000 if (p->encoding == AUDIO_ENCODING_ULINEAR_OE) 6001 p->encoding = AUDIO_ENCODING_ULINEAR_NE; 6002 } 6003 6004 if (p->precision > p->stride) 6005 return EINVAL; 6006 break; 6007 case AUDIO_ENCODING_MPEG_L1_STREAM: 6008 case AUDIO_ENCODING_MPEG_L1_PACKETS: 6009 case AUDIO_ENCODING_MPEG_L1_SYSTEM: 6010 case AUDIO_ENCODING_MPEG_L2_STREAM: 6011 case AUDIO_ENCODING_MPEG_L2_PACKETS: 6012 case AUDIO_ENCODING_MPEG_L2_SYSTEM: 6013 case AUDIO_ENCODING_AC3: 6014 break; 6015 default: 6016 return EINVAL; 6017 } 6018 6019 /* sanity check # of channels*/ 6020 if (p->channels < 1 || p->channels > AUDIO_MAX_CHANNELS) 6021 return EINVAL; 6022 6023 return 0; 6024 } 6025 6026 /* 6027 * Initialize playback and record mixers. 6028 * mode (AUMODE_{PLAY,RECORD}) indicates the mixer to be initalized. 6029 * phwfmt and rhwfmt indicate the hardware format. pfil and rfil indicate 6030 * the filter registration information. These four must not be NULL. 6031 * If successful returns 0. Otherwise returns errno. 6032 * Must be called with sc_lock held. 6033 * Must not be called if there are any tracks. 6034 * Caller should check that the initialization succeed by whether 6035 * sc_[pr]mixer is not NULL. 6036 */ 6037 static int 6038 audio_mixers_init(struct audio_softc *sc, int mode, 6039 const audio_format2_t *phwfmt, const audio_format2_t *rhwfmt, 6040 const audio_filter_reg_t *pfil, const audio_filter_reg_t *rfil) 6041 { 6042 int error; 6043 6044 KASSERT(phwfmt != NULL); 6045 KASSERT(rhwfmt != NULL); 6046 KASSERT(pfil != NULL); 6047 KASSERT(rfil != NULL); 6048 KASSERT(mutex_owned(sc->sc_lock)); 6049 6050 if ((mode & AUMODE_PLAY)) { 6051 if (sc->sc_pmixer) { 6052 audio_mixer_destroy(sc, sc->sc_pmixer); 6053 kmem_free(sc->sc_pmixer, sizeof(*sc->sc_pmixer)); 6054 } 6055 sc->sc_pmixer = kmem_zalloc(sizeof(*sc->sc_pmixer), KM_SLEEP); 6056 error = audio_mixer_init(sc, AUMODE_PLAY, phwfmt, pfil); 6057 if (error) { 6058 aprint_error_dev(sc->sc_dev, 6059 "configuring playback mode failed\n"); 6060 kmem_free(sc->sc_pmixer, sizeof(*sc->sc_pmixer)); 6061 sc->sc_pmixer = NULL; 6062 return error; 6063 } 6064 } 6065 if ((mode & AUMODE_RECORD)) { 6066 if (sc->sc_rmixer) { 6067 audio_mixer_destroy(sc, sc->sc_rmixer); 6068 kmem_free(sc->sc_rmixer, sizeof(*sc->sc_rmixer)); 6069 } 6070 sc->sc_rmixer = kmem_zalloc(sizeof(*sc->sc_rmixer), KM_SLEEP); 6071 error = audio_mixer_init(sc, AUMODE_RECORD, rhwfmt, rfil); 6072 if (error) { 6073 aprint_error_dev(sc->sc_dev, 6074 "configuring record mode failed\n"); 6075 kmem_free(sc->sc_rmixer, sizeof(*sc->sc_rmixer)); 6076 sc->sc_rmixer = NULL; 6077 return error; 6078 } 6079 } 6080 6081 return 0; 6082 } 6083 6084 /* 6085 * Select a frequency. 6086 * Prioritize 48kHz and 44.1kHz. Otherwise choose the highest one. 6087 * XXX Better algorithm? 6088 */ 6089 static int 6090 audio_select_freq(const struct audio_format *fmt) 6091 { 6092 int freq; 6093 int high; 6094 int low; 6095 int j; 6096 6097 if (fmt->frequency_type == 0) { 6098 low = fmt->frequency[0]; 6099 high = fmt->frequency[1]; 6100 freq = 48000; 6101 if (low <= freq && freq <= high) { 6102 return freq; 6103 } 6104 freq = 44100; 6105 if (low <= freq && freq <= high) { 6106 return freq; 6107 } 6108 return high; 6109 } else { 6110 for (j = 0; j < fmt->frequency_type; j++) { 6111 if (fmt->frequency[j] == 48000) { 6112 return fmt->frequency[j]; 6113 } 6114 } 6115 high = 0; 6116 for (j = 0; j < fmt->frequency_type; j++) { 6117 if (fmt->frequency[j] == 44100) { 6118 return fmt->frequency[j]; 6119 } 6120 if (fmt->frequency[j] > high) { 6121 high = fmt->frequency[j]; 6122 } 6123 } 6124 return high; 6125 } 6126 } 6127 6128 /* 6129 * Probe playback and/or recording format (depending on *modep). 6130 * *modep is an in-out parameter. It indicates the direction to configure 6131 * as an argument, and the direction configured is written back as out 6132 * parameter. 6133 * If successful, probed hardware format is stored into *phwfmt, *rhwfmt 6134 * depending on *modep, and return 0. Otherwise it returns errno. 6135 * Must be called with sc_lock held. 6136 */ 6137 static int 6138 audio_hw_probe(struct audio_softc *sc, int is_indep, int *modep, 6139 audio_format2_t *phwfmt, audio_format2_t *rhwfmt) 6140 { 6141 audio_format2_t fmt; 6142 int mode; 6143 int error = 0; 6144 6145 KASSERT(mutex_owned(sc->sc_lock)); 6146 6147 mode = *modep; 6148 KASSERTMSG((mode & (AUMODE_PLAY | AUMODE_RECORD)) != 0, 6149 "invalid mode = %x", mode); 6150 6151 if (is_indep) { 6152 /* On independent devices, probe separately. */ 6153 if ((mode & AUMODE_PLAY) != 0) { 6154 error = audio_hw_probe_fmt(sc, phwfmt, AUMODE_PLAY); 6155 if (error) 6156 mode &= ~AUMODE_PLAY; 6157 } 6158 if ((mode & AUMODE_RECORD) != 0) { 6159 error = audio_hw_probe_fmt(sc, rhwfmt, AUMODE_RECORD); 6160 if (error) 6161 mode &= ~AUMODE_RECORD; 6162 } 6163 } else { 6164 /* On non independent devices, probe simultaneously. */ 6165 error = audio_hw_probe_fmt(sc, &fmt, mode); 6166 if (error) { 6167 mode = 0; 6168 } else { 6169 *phwfmt = fmt; 6170 *rhwfmt = fmt; 6171 } 6172 } 6173 6174 *modep = mode; 6175 return error; 6176 } 6177 6178 /* 6179 * Choose the most preferred hardware format. 6180 * If successful, it will store the chosen format into *cand and return 0. 6181 * Otherwise, return errno. 6182 * Must be called with sc_lock held. 6183 */ 6184 static int 6185 audio_hw_probe_fmt(struct audio_softc *sc, audio_format2_t *cand, int mode) 6186 { 6187 audio_format_query_t query; 6188 int cand_score; 6189 int score; 6190 int i; 6191 int error; 6192 6193 KASSERT(mutex_owned(sc->sc_lock)); 6194 6195 /* 6196 * Score each formats and choose the highest one. 6197 * 6198 * +---- priority(0-3) 6199 * |+--- encoding/precision 6200 * ||+-- channels 6201 * score = 0x000000PEC 6202 */ 6203 6204 cand_score = 0; 6205 for (i = 0; ; i++) { 6206 memset(&query, 0, sizeof(query)); 6207 query.index = i; 6208 6209 error = sc->hw_if->query_format(sc->hw_hdl, &query); 6210 if (error == EINVAL) 6211 break; 6212 if (error) 6213 return error; 6214 6215 #if defined(AUDIO_DEBUG) 6216 DPRINTF(1, "fmt[%d] %c%c pri=%d %s,%d/%dbit,%dch,", i, 6217 (query.fmt.mode & AUMODE_PLAY) ? 'P' : '-', 6218 (query.fmt.mode & AUMODE_RECORD) ? 'R' : '-', 6219 query.fmt.priority, 6220 audio_encoding_name(query.fmt.encoding), 6221 query.fmt.validbits, 6222 query.fmt.precision, 6223 query.fmt.channels); 6224 if (query.fmt.frequency_type == 0) { 6225 DPRINTF(1, "{%d-%d", 6226 query.fmt.frequency[0], query.fmt.frequency[1]); 6227 } else { 6228 int j; 6229 for (j = 0; j < query.fmt.frequency_type; j++) { 6230 DPRINTF(1, "%c%d", 6231 (j == 0) ? '{' : ',', 6232 query.fmt.frequency[j]); 6233 } 6234 } 6235 DPRINTF(1, "}\n"); 6236 #endif 6237 6238 if ((query.fmt.mode & mode) == 0) { 6239 DPRINTF(1, "fmt[%d] skip; mode not match %d\n", i, 6240 mode); 6241 continue; 6242 } 6243 6244 if (query.fmt.priority < 0) { 6245 DPRINTF(1, "fmt[%d] skip; unsupported encoding\n", i); 6246 continue; 6247 } 6248 6249 /* Score */ 6250 score = (query.fmt.priority & 3) * 0x100; 6251 if (query.fmt.encoding == AUDIO_ENCODING_SLINEAR_NE && 6252 query.fmt.validbits == AUDIO_INTERNAL_BITS && 6253 query.fmt.precision == AUDIO_INTERNAL_BITS) { 6254 score += 0x20; 6255 } else if (query.fmt.encoding == AUDIO_ENCODING_SLINEAR_OE && 6256 query.fmt.validbits == AUDIO_INTERNAL_BITS && 6257 query.fmt.precision == AUDIO_INTERNAL_BITS) { 6258 score += 0x10; 6259 } 6260 score += query.fmt.channels; 6261 6262 if (score < cand_score) { 6263 DPRINTF(1, "fmt[%d] skip; score 0x%x < 0x%x\n", i, 6264 score, cand_score); 6265 continue; 6266 } 6267 6268 /* Update candidate */ 6269 cand_score = score; 6270 cand->encoding = query.fmt.encoding; 6271 cand->precision = query.fmt.validbits; 6272 cand->stride = query.fmt.precision; 6273 cand->channels = query.fmt.channels; 6274 cand->sample_rate = audio_select_freq(&query.fmt); 6275 DPRINTF(1, "fmt[%d] candidate (score=0x%x)" 6276 " pri=%d %s,%d/%d,%dch,%dHz\n", i, 6277 cand_score, query.fmt.priority, 6278 audio_encoding_name(query.fmt.encoding), 6279 cand->precision, cand->stride, 6280 cand->channels, cand->sample_rate); 6281 } 6282 6283 if (cand_score == 0) { 6284 DPRINTF(1, "%s no fmt\n", __func__); 6285 return ENXIO; 6286 } 6287 DPRINTF(1, "%s selected: %s,%d/%d,%dch,%dHz\n", __func__, 6288 audio_encoding_name(cand->encoding), 6289 cand->precision, cand->stride, cand->channels, cand->sample_rate); 6290 return 0; 6291 } 6292 6293 /* 6294 * Validate fmt with query_format. 6295 * If fmt is included in the result of query_format, returns 0. 6296 * Otherwise returns EINVAL. 6297 * Must be called with sc_lock held. 6298 */ 6299 static int 6300 audio_hw_validate_format(struct audio_softc *sc, int mode, 6301 const audio_format2_t *fmt) 6302 { 6303 audio_format_query_t query; 6304 struct audio_format *q; 6305 int index; 6306 int error; 6307 int j; 6308 6309 KASSERT(mutex_owned(sc->sc_lock)); 6310 6311 /* 6312 * If query_format is not supported by hardware driver, 6313 * a rough check instead will be performed. 6314 * XXX This will gone in the future. 6315 */ 6316 if (sc->hw_if->query_format == NULL) { 6317 if (fmt->encoding != AUDIO_ENCODING_SLINEAR_NE) 6318 return EINVAL; 6319 if (fmt->precision != AUDIO_INTERNAL_BITS) 6320 return EINVAL; 6321 if (fmt->stride != AUDIO_INTERNAL_BITS) 6322 return EINVAL; 6323 return 0; 6324 } 6325 6326 for (index = 0; ; index++) { 6327 query.index = index; 6328 error = sc->hw_if->query_format(sc->hw_hdl, &query); 6329 if (error == EINVAL) 6330 break; 6331 if (error) 6332 return error; 6333 6334 q = &query.fmt; 6335 /* 6336 * Note that fmt is audio_format2_t (precision/stride) but 6337 * q is audio_format_t (validbits/precision). 6338 */ 6339 if ((q->mode & mode) == 0) { 6340 continue; 6341 } 6342 if (fmt->encoding != q->encoding) { 6343 continue; 6344 } 6345 if (fmt->precision != q->validbits) { 6346 continue; 6347 } 6348 if (fmt->stride != q->precision) { 6349 continue; 6350 } 6351 if (fmt->channels != q->channels) { 6352 continue; 6353 } 6354 if (q->frequency_type == 0) { 6355 if (fmt->sample_rate < q->frequency[0] || 6356 fmt->sample_rate > q->frequency[1]) { 6357 continue; 6358 } 6359 } else { 6360 for (j = 0; j < q->frequency_type; j++) { 6361 if (fmt->sample_rate == q->frequency[j]) 6362 break; 6363 } 6364 if (j == query.fmt.frequency_type) { 6365 continue; 6366 } 6367 } 6368 6369 /* Matched. */ 6370 return 0; 6371 } 6372 6373 return EINVAL; 6374 } 6375 6376 /* 6377 * Set track mixer's format depending on ai->mode. 6378 * If AUMODE_PLAY is set in ai->mode, it set up the playback mixer 6379 * with ai.play.{channels, sample_rate}. 6380 * If AUMODE_RECORD is set in ai->mode, it set up the recording mixer 6381 * with ai.record.{channels, sample_rate}. 6382 * All other fields in ai are ignored. 6383 * If successful returns 0. Otherwise returns errno. 6384 * This function does not roll back even if it fails. 6385 * Must be called with sc_lock held. 6386 */ 6387 static int 6388 audio_mixers_set_format(struct audio_softc *sc, const struct audio_info *ai) 6389 { 6390 audio_format2_t phwfmt; 6391 audio_format2_t rhwfmt; 6392 audio_filter_reg_t pfil; 6393 audio_filter_reg_t rfil; 6394 int mode; 6395 int props; 6396 int error; 6397 6398 KASSERT(mutex_owned(sc->sc_lock)); 6399 6400 /* 6401 * Even when setting either one of playback and recording, 6402 * both must be halted. 6403 */ 6404 if (sc->sc_popens + sc->sc_ropens > 0) 6405 return EBUSY; 6406 6407 if (!SPECIFIED(ai->mode) || ai->mode == 0) 6408 return ENOTTY; 6409 6410 /* Only channels and sample_rate are changeable. */ 6411 mode = ai->mode; 6412 if ((mode & AUMODE_PLAY)) { 6413 phwfmt.encoding = ai->play.encoding; 6414 phwfmt.precision = ai->play.precision; 6415 phwfmt.stride = ai->play.precision; 6416 phwfmt.channels = ai->play.channels; 6417 phwfmt.sample_rate = ai->play.sample_rate; 6418 } 6419 if ((mode & AUMODE_RECORD)) { 6420 rhwfmt.encoding = ai->record.encoding; 6421 rhwfmt.precision = ai->record.precision; 6422 rhwfmt.stride = ai->record.precision; 6423 rhwfmt.channels = ai->record.channels; 6424 rhwfmt.sample_rate = ai->record.sample_rate; 6425 } 6426 6427 /* On non-independent devices, use the same format for both. */ 6428 props = audio_get_props(sc); 6429 if ((props & AUDIO_PROP_INDEPENDENT) == 0) { 6430 if (mode == AUMODE_RECORD) { 6431 phwfmt = rhwfmt; 6432 } else { 6433 rhwfmt = phwfmt; 6434 } 6435 mode = AUMODE_PLAY | AUMODE_RECORD; 6436 } 6437 6438 /* Then, unset the direction not exist on the hardware. */ 6439 if ((props & AUDIO_PROP_PLAYBACK) == 0) 6440 mode &= ~AUMODE_PLAY; 6441 if ((props & AUDIO_PROP_CAPTURE) == 0) 6442 mode &= ~AUMODE_RECORD; 6443 6444 /* debug */ 6445 if ((mode & AUMODE_PLAY)) { 6446 TRACE(1, "play=%s/%d/%d/%dch/%dHz", 6447 audio_encoding_name(phwfmt.encoding), 6448 phwfmt.precision, 6449 phwfmt.stride, 6450 phwfmt.channels, 6451 phwfmt.sample_rate); 6452 } 6453 if ((mode & AUMODE_RECORD)) { 6454 TRACE(1, "rec =%s/%d/%d/%dch/%dHz", 6455 audio_encoding_name(rhwfmt.encoding), 6456 rhwfmt.precision, 6457 rhwfmt.stride, 6458 rhwfmt.channels, 6459 rhwfmt.sample_rate); 6460 } 6461 6462 /* Check the format */ 6463 if ((mode & AUMODE_PLAY)) { 6464 if (audio_hw_validate_format(sc, AUMODE_PLAY, &phwfmt)) { 6465 TRACE(1, "invalid format"); 6466 return EINVAL; 6467 } 6468 } 6469 if ((mode & AUMODE_RECORD)) { 6470 if (audio_hw_validate_format(sc, AUMODE_RECORD, &rhwfmt)) { 6471 TRACE(1, "invalid format"); 6472 return EINVAL; 6473 } 6474 } 6475 6476 /* Configure the mixers. */ 6477 memset(&pfil, 0, sizeof(pfil)); 6478 memset(&rfil, 0, sizeof(rfil)); 6479 error = audio_hw_set_format(sc, mode, &phwfmt, &rhwfmt, &pfil, &rfil); 6480 if (error) 6481 return error; 6482 6483 error = audio_mixers_init(sc, mode, &phwfmt, &rhwfmt, &pfil, &rfil); 6484 if (error) 6485 return error; 6486 6487 return 0; 6488 } 6489 6490 /* 6491 * Store current mixers format into *ai. 6492 */ 6493 static void 6494 audio_mixers_get_format(struct audio_softc *sc, struct audio_info *ai) 6495 { 6496 /* 6497 * There is no stride information in audio_info but it doesn't matter. 6498 * trackmixer always treats stride and precision as the same. 6499 */ 6500 AUDIO_INITINFO(ai); 6501 ai->mode = 0; 6502 if (sc->sc_pmixer) { 6503 audio_format2_t *fmt = &sc->sc_pmixer->track_fmt; 6504 ai->play.encoding = fmt->encoding; 6505 ai->play.precision = fmt->precision; 6506 ai->play.channels = fmt->channels; 6507 ai->play.sample_rate = fmt->sample_rate; 6508 ai->mode |= AUMODE_PLAY; 6509 } 6510 if (sc->sc_rmixer) { 6511 audio_format2_t *fmt = &sc->sc_rmixer->track_fmt; 6512 ai->record.encoding = fmt->encoding; 6513 ai->record.precision = fmt->precision; 6514 ai->record.channels = fmt->channels; 6515 ai->record.sample_rate = fmt->sample_rate; 6516 ai->mode |= AUMODE_RECORD; 6517 } 6518 } 6519 6520 /* 6521 * audio_info details: 6522 * 6523 * ai.{play,record}.sample_rate (R/W) 6524 * ai.{play,record}.encoding (R/W) 6525 * ai.{play,record}.precision (R/W) 6526 * ai.{play,record}.channels (R/W) 6527 * These specify the playback or recording format. 6528 * Ignore members within an inactive track. 6529 * 6530 * ai.mode (R/W) 6531 * It specifies the playback or recording mode, AUMODE_*. 6532 * Currently, a mode change operation by ai.mode after opening is 6533 * prohibited. In addition, AUMODE_PLAY_ALL no longer makes sense. 6534 * However, it's possible to get or to set for backward compatibility. 6535 * 6536 * ai.{hiwat,lowat} (R/W) 6537 * These specify the high water mark and low water mark for playback 6538 * track. The unit is block. 6539 * 6540 * ai.{play,record}.gain (R/W) 6541 * It specifies the HW mixer volume in 0-255. 6542 * It is historical reason that the gain is connected to HW mixer. 6543 * 6544 * ai.{play,record}.balance (R/W) 6545 * It specifies the left-right balance of HW mixer in 0-64. 6546 * 32 means the center. 6547 * It is historical reason that the balance is connected to HW mixer. 6548 * 6549 * ai.{play,record}.port (R/W) 6550 * It specifies the input/output port of HW mixer. 6551 * 6552 * ai.monitor_gain (R/W) 6553 * It specifies the recording monitor gain(?) of HW mixer. 6554 * 6555 * ai.{play,record}.pause (R/W) 6556 * Non-zero means the track is paused. 6557 * 6558 * ai.play.seek (R/-) 6559 * It indicates the number of bytes written but not processed. 6560 * ai.record.seek (R/-) 6561 * It indicates the number of bytes to be able to read. 6562 * 6563 * ai.{play,record}.avail_ports (R/-) 6564 * Mixer info. 6565 * 6566 * ai.{play,record}.buffer_size (R/-) 6567 * It indicates the buffer size in bytes. Internally it means usrbuf. 6568 * 6569 * ai.{play,record}.samples (R/-) 6570 * It indicates the total number of bytes played or recorded. 6571 * 6572 * ai.{play,record}.eof (R/-) 6573 * It indicates the number of times reached EOF(?). 6574 * 6575 * ai.{play,record}.error (R/-) 6576 * Non-zero indicates overflow/underflow has occured. 6577 * 6578 * ai.{play,record}.waiting (R/-) 6579 * Non-zero indicates that other process waits to open. 6580 * It will never happen anymore. 6581 * 6582 * ai.{play,record}.open (R/-) 6583 * Non-zero indicates the direction is opened by this process(?). 6584 * XXX Is this better to indicate that "the device is opened by 6585 * at least one process"? 6586 * 6587 * ai.{play,record}.active (R/-) 6588 * Non-zero indicates that I/O is currently active. 6589 * 6590 * ai.blocksize (R/-) 6591 * It indicates the block size in bytes. 6592 * XXX The blocksize of playback and recording may be different. 6593 */ 6594 6595 /* 6596 * Pause consideration: 6597 * 6598 * The introduction of these two behavior makes pause/unpause operation 6599 * simple. 6600 * 1. The first read/write access of the first track makes mixer start. 6601 * 2. A pause of the last track doesn't make mixer stop. 6602 */ 6603 6604 /* 6605 * Set both track's parameters within a file depending on ai. 6606 * Update sc_sound_[pr]* if set. 6607 * Must be called with sc_lock and sc_exlock held. 6608 */ 6609 static int 6610 audio_file_setinfo(struct audio_softc *sc, audio_file_t *file, 6611 const struct audio_info *ai) 6612 { 6613 const struct audio_prinfo *pi; 6614 const struct audio_prinfo *ri; 6615 audio_track_t *ptrack; 6616 audio_track_t *rtrack; 6617 audio_format2_t pfmt; 6618 audio_format2_t rfmt; 6619 int pchanges; 6620 int rchanges; 6621 int mode; 6622 struct audio_info saved_ai; 6623 audio_format2_t saved_pfmt; 6624 audio_format2_t saved_rfmt; 6625 int error; 6626 6627 KASSERT(mutex_owned(sc->sc_lock)); 6628 KASSERT(sc->sc_exlock); 6629 6630 pi = &ai->play; 6631 ri = &ai->record; 6632 pchanges = 0; 6633 rchanges = 0; 6634 6635 ptrack = file->ptrack; 6636 rtrack = file->rtrack; 6637 6638 #if defined(AUDIO_DEBUG) 6639 if (audiodebug >= 2) { 6640 char buf[256]; 6641 char p[64]; 6642 int buflen; 6643 int plen; 6644 #define SPRINTF(var, fmt...) do { \ 6645 var##len += snprintf(var + var##len, sizeof(var) - var##len, fmt); \ 6646 } while (0) 6647 6648 buflen = 0; 6649 plen = 0; 6650 if (SPECIFIED(pi->encoding)) 6651 SPRINTF(p, "/%s", audio_encoding_name(pi->encoding)); 6652 if (SPECIFIED(pi->precision)) 6653 SPRINTF(p, "/%dbit", pi->precision); 6654 if (SPECIFIED(pi->channels)) 6655 SPRINTF(p, "/%dch", pi->channels); 6656 if (SPECIFIED(pi->sample_rate)) 6657 SPRINTF(p, "/%dHz", pi->sample_rate); 6658 if (plen > 0) 6659 SPRINTF(buf, ",play.param=%s", p + 1); 6660 6661 plen = 0; 6662 if (SPECIFIED(ri->encoding)) 6663 SPRINTF(p, "/%s", audio_encoding_name(ri->encoding)); 6664 if (SPECIFIED(ri->precision)) 6665 SPRINTF(p, "/%dbit", ri->precision); 6666 if (SPECIFIED(ri->channels)) 6667 SPRINTF(p, "/%dch", ri->channels); 6668 if (SPECIFIED(ri->sample_rate)) 6669 SPRINTF(p, "/%dHz", ri->sample_rate); 6670 if (plen > 0) 6671 SPRINTF(buf, ",record.param=%s", p + 1); 6672 6673 if (SPECIFIED(ai->mode)) 6674 SPRINTF(buf, ",mode=%d", ai->mode); 6675 if (SPECIFIED(ai->hiwat)) 6676 SPRINTF(buf, ",hiwat=%d", ai->hiwat); 6677 if (SPECIFIED(ai->lowat)) 6678 SPRINTF(buf, ",lowat=%d", ai->lowat); 6679 if (SPECIFIED(ai->play.gain)) 6680 SPRINTF(buf, ",play.gain=%d", ai->play.gain); 6681 if (SPECIFIED(ai->record.gain)) 6682 SPRINTF(buf, ",record.gain=%d", ai->record.gain); 6683 if (SPECIFIED_CH(ai->play.balance)) 6684 SPRINTF(buf, ",play.balance=%d", ai->play.balance); 6685 if (SPECIFIED_CH(ai->record.balance)) 6686 SPRINTF(buf, ",record.balance=%d", ai->record.balance); 6687 if (SPECIFIED(ai->play.port)) 6688 SPRINTF(buf, ",play.port=%d", ai->play.port); 6689 if (SPECIFIED(ai->record.port)) 6690 SPRINTF(buf, ",record.port=%d", ai->record.port); 6691 if (SPECIFIED(ai->monitor_gain)) 6692 SPRINTF(buf, ",monitor_gain=%d", ai->monitor_gain); 6693 if (SPECIFIED_CH(ai->play.pause)) 6694 SPRINTF(buf, ",play.pause=%d", ai->play.pause); 6695 if (SPECIFIED_CH(ai->record.pause)) 6696 SPRINTF(buf, ",record.pause=%d", ai->record.pause); 6697 6698 if (buflen > 0) 6699 TRACE(2, "specified %s", buf + 1); 6700 } 6701 #endif 6702 6703 AUDIO_INITINFO(&saved_ai); 6704 /* XXX shut up gcc */ 6705 memset(&saved_pfmt, 0, sizeof(saved_pfmt)); 6706 memset(&saved_rfmt, 0, sizeof(saved_rfmt)); 6707 6708 /* Set default value and save current parameters */ 6709 if (ptrack) { 6710 pfmt = ptrack->usrbuf.fmt; 6711 saved_pfmt = ptrack->usrbuf.fmt; 6712 saved_ai.play.pause = ptrack->is_pause; 6713 } 6714 if (rtrack) { 6715 rfmt = rtrack->usrbuf.fmt; 6716 saved_rfmt = rtrack->usrbuf.fmt; 6717 saved_ai.record.pause = rtrack->is_pause; 6718 } 6719 saved_ai.mode = file->mode; 6720 6721 /* Overwrite if specified */ 6722 mode = file->mode; 6723 if (SPECIFIED(ai->mode)) { 6724 /* 6725 * Setting ai->mode no longer does anything because it's 6726 * prohibited to change playback/recording mode after open 6727 * and AUMODE_PLAY_ALL is obsoleted. However, it still 6728 * keeps the state of AUMODE_PLAY_ALL itself for backward 6729 * compatibility. 6730 * In the internal, only file->mode has the state of 6731 * AUMODE_PLAY_ALL flag and track->mode in both track does 6732 * not have. 6733 */ 6734 if ((file->mode & AUMODE_PLAY)) { 6735 mode = (file->mode & (AUMODE_PLAY | AUMODE_RECORD)) 6736 | (ai->mode & AUMODE_PLAY_ALL); 6737 } 6738 } 6739 6740 if (ptrack) { 6741 pchanges = audio_track_setinfo_check(&pfmt, pi); 6742 if (pchanges == -1) { 6743 TRACET(1, ptrack, "check play.params failed"); 6744 return EINVAL; 6745 } 6746 if (SPECIFIED(ai->mode)) 6747 pchanges = 1; 6748 } 6749 if (rtrack) { 6750 rchanges = audio_track_setinfo_check(&rfmt, ri); 6751 if (rchanges == -1) { 6752 TRACET(1, rtrack, "check record.params failed"); 6753 return EINVAL; 6754 } 6755 if (SPECIFIED(ai->mode)) 6756 rchanges = 1; 6757 } 6758 6759 /* 6760 * Even when setting either one of playback and recording, 6761 * both track must be halted. 6762 */ 6763 if (pchanges || rchanges) { 6764 audio_file_clear(sc, file); 6765 #if defined(AUDIO_DEBUG) 6766 char fmtbuf[64]; 6767 if (pchanges) { 6768 audio_format2_tostr(fmtbuf, sizeof(fmtbuf), &pfmt); 6769 DPRINTF(1, "audio track#%d play mode: %s\n", 6770 ptrack->id, fmtbuf); 6771 } 6772 if (rchanges) { 6773 audio_format2_tostr(fmtbuf, sizeof(fmtbuf), &rfmt); 6774 DPRINTF(1, "audio track#%d rec mode: %s\n", 6775 rtrack->id, fmtbuf); 6776 } 6777 #endif 6778 } 6779 6780 /* Set mixer parameters */ 6781 error = audio_hw_setinfo(sc, ai, &saved_ai); 6782 if (error) 6783 goto abort1; 6784 6785 /* Set to track and update sticky parameters */ 6786 error = 0; 6787 file->mode = mode; 6788 if (ptrack) { 6789 if (SPECIFIED_CH(pi->pause)) { 6790 ptrack->is_pause = pi->pause; 6791 sc->sc_sound_ppause = pi->pause; 6792 } 6793 if (pchanges) { 6794 audio_track_lock_enter(ptrack); 6795 error = audio_track_set_format(ptrack, &pfmt); 6796 audio_track_lock_exit(ptrack); 6797 if (error) { 6798 TRACET(1, ptrack, "set play.params failed"); 6799 goto abort2; 6800 } 6801 sc->sc_sound_pparams = pfmt; 6802 } 6803 /* Change water marks after initializing the buffers. */ 6804 if (SPECIFIED(ai->hiwat) || SPECIFIED(ai->lowat)) 6805 audio_track_setinfo_water(ptrack, ai); 6806 } 6807 if (rtrack) { 6808 if (SPECIFIED_CH(ri->pause)) { 6809 rtrack->is_pause = ri->pause; 6810 sc->sc_sound_rpause = ri->pause; 6811 } 6812 if (rchanges) { 6813 audio_track_lock_enter(rtrack); 6814 error = audio_track_set_format(rtrack, &rfmt); 6815 audio_track_lock_exit(rtrack); 6816 if (error) { 6817 TRACET(1, rtrack, "set record.params failed"); 6818 goto abort3; 6819 } 6820 sc->sc_sound_rparams = rfmt; 6821 } 6822 } 6823 6824 return 0; 6825 6826 /* Rollback */ 6827 abort3: 6828 if (error != ENOMEM) { 6829 rtrack->is_pause = saved_ai.record.pause; 6830 audio_track_lock_enter(rtrack); 6831 audio_track_set_format(rtrack, &saved_rfmt); 6832 audio_track_lock_exit(rtrack); 6833 } 6834 abort2: 6835 if (ptrack && error != ENOMEM) { 6836 ptrack->is_pause = saved_ai.play.pause; 6837 audio_track_lock_enter(ptrack); 6838 audio_track_set_format(ptrack, &saved_pfmt); 6839 audio_track_lock_exit(ptrack); 6840 sc->sc_sound_pparams = saved_pfmt; 6841 sc->sc_sound_ppause = saved_ai.play.pause; 6842 } 6843 file->mode = saved_ai.mode; 6844 abort1: 6845 audio_hw_setinfo(sc, &saved_ai, NULL); 6846 6847 return error; 6848 } 6849 6850 /* 6851 * Write SPECIFIED() parameters within info back to fmt. 6852 * Return value of 1 indicates that fmt is modified. 6853 * Return value of 0 indicates that fmt is not modified. 6854 * Return value of -1 indicates that error EINVAL has occurred. 6855 */ 6856 static int 6857 audio_track_setinfo_check(audio_format2_t *fmt, const struct audio_prinfo *info) 6858 { 6859 int changes; 6860 6861 changes = 0; 6862 if (SPECIFIED(info->sample_rate)) { 6863 if (info->sample_rate < AUDIO_MIN_FREQUENCY) 6864 return -1; 6865 if (info->sample_rate > AUDIO_MAX_FREQUENCY) 6866 return -1; 6867 fmt->sample_rate = info->sample_rate; 6868 changes = 1; 6869 } 6870 if (SPECIFIED(info->encoding)) { 6871 fmt->encoding = info->encoding; 6872 changes = 1; 6873 } 6874 if (SPECIFIED(info->precision)) { 6875 fmt->precision = info->precision; 6876 /* we don't have API to specify stride */ 6877 fmt->stride = info->precision; 6878 changes = 1; 6879 } 6880 if (SPECIFIED(info->channels)) { 6881 fmt->channels = info->channels; 6882 changes = 1; 6883 } 6884 6885 if (changes) { 6886 if (audio_check_params(fmt) != 0) { 6887 #ifdef DIAGNOSTIC 6888 char fmtbuf[64]; 6889 audio_format2_tostr(fmtbuf, sizeof(fmtbuf), fmt); 6890 printf("%s failed: %s\n", __func__, fmtbuf); 6891 #endif 6892 return -1; 6893 } 6894 } 6895 6896 return changes; 6897 } 6898 6899 /* 6900 * Change water marks for playback track if specfied. 6901 */ 6902 static void 6903 audio_track_setinfo_water(audio_track_t *track, const struct audio_info *ai) 6904 { 6905 u_int blks; 6906 u_int maxblks; 6907 u_int blksize; 6908 6909 KASSERT(audio_track_is_playback(track)); 6910 6911 blksize = track->usrbuf_blksize; 6912 maxblks = track->usrbuf.capacity / blksize; 6913 6914 if (SPECIFIED(ai->hiwat)) { 6915 blks = ai->hiwat; 6916 if (blks > maxblks) 6917 blks = maxblks; 6918 if (blks < 2) 6919 blks = 2; 6920 track->usrbuf_usedhigh = blks * blksize; 6921 } 6922 if (SPECIFIED(ai->lowat)) { 6923 blks = ai->lowat; 6924 if (blks > maxblks - 1) 6925 blks = maxblks - 1; 6926 track->usrbuf_usedlow = blks * blksize; 6927 } 6928 if (SPECIFIED(ai->hiwat) || SPECIFIED(ai->lowat)) { 6929 if (track->usrbuf_usedlow > track->usrbuf_usedhigh - blksize) { 6930 track->usrbuf_usedlow = track->usrbuf_usedhigh - 6931 blksize; 6932 } 6933 } 6934 } 6935 6936 /* 6937 * Set hardware part of *ai. 6938 * The parameters handled here are *.port, *.gain, *.balance and monitor_gain. 6939 * If oldai is specified, previous parameters are stored. 6940 * This function itself does not roll back if error occurred. 6941 * Must be called with sc_lock and sc_exlock held. 6942 */ 6943 static int 6944 audio_hw_setinfo(struct audio_softc *sc, const struct audio_info *newai, 6945 struct audio_info *oldai) 6946 { 6947 const struct audio_prinfo *newpi; 6948 const struct audio_prinfo *newri; 6949 struct audio_prinfo *oldpi; 6950 struct audio_prinfo *oldri; 6951 u_int pgain; 6952 u_int rgain; 6953 u_char pbalance; 6954 u_char rbalance; 6955 int error; 6956 6957 KASSERT(mutex_owned(sc->sc_lock)); 6958 KASSERT(sc->sc_exlock); 6959 6960 /* XXX shut up gcc */ 6961 oldpi = NULL; 6962 oldri = NULL; 6963 6964 newpi = &newai->play; 6965 newri = &newai->record; 6966 if (oldai) { 6967 oldpi = &oldai->play; 6968 oldri = &oldai->record; 6969 } 6970 error = 0; 6971 6972 /* 6973 * It looks like unnecessary to halt HW mixers to set HW mixers. 6974 * mixer_ioctl(MIXER_WRITE) also doesn't halt. 6975 */ 6976 6977 if (SPECIFIED(newpi->port)) { 6978 if (oldai) 6979 oldpi->port = au_get_port(sc, &sc->sc_outports); 6980 error = au_set_port(sc, &sc->sc_outports, newpi->port); 6981 if (error) { 6982 device_printf(sc->sc_dev, 6983 "setting play.port=%d failed with %d\n", 6984 newpi->port, error); 6985 goto abort; 6986 } 6987 } 6988 if (SPECIFIED(newri->port)) { 6989 if (oldai) 6990 oldri->port = au_get_port(sc, &sc->sc_inports); 6991 error = au_set_port(sc, &sc->sc_inports, newri->port); 6992 if (error) { 6993 device_printf(sc->sc_dev, 6994 "setting record.port=%d failed with %d\n", 6995 newri->port, error); 6996 goto abort; 6997 } 6998 } 6999 7000 /* Backup play.{gain,balance} */ 7001 if (SPECIFIED(newpi->gain) || SPECIFIED_CH(newpi->balance)) { 7002 au_get_gain(sc, &sc->sc_outports, &pgain, &pbalance); 7003 if (oldai) { 7004 oldpi->gain = pgain; 7005 oldpi->balance = pbalance; 7006 } 7007 } 7008 /* Backup record.{gain,balance} */ 7009 if (SPECIFIED(newri->gain) || SPECIFIED_CH(newri->balance)) { 7010 au_get_gain(sc, &sc->sc_inports, &rgain, &rbalance); 7011 if (oldai) { 7012 oldri->gain = rgain; 7013 oldri->balance = rbalance; 7014 } 7015 } 7016 if (SPECIFIED(newpi->gain)) { 7017 error = au_set_gain(sc, &sc->sc_outports, 7018 newpi->gain, pbalance); 7019 if (error) { 7020 device_printf(sc->sc_dev, 7021 "setting play.gain=%d failed with %d\n", 7022 newpi->gain, error); 7023 goto abort; 7024 } 7025 } 7026 if (SPECIFIED(newri->gain)) { 7027 error = au_set_gain(sc, &sc->sc_inports, 7028 newri->gain, rbalance); 7029 if (error) { 7030 device_printf(sc->sc_dev, 7031 "setting record.gain=%d failed with %d\n", 7032 newri->gain, error); 7033 goto abort; 7034 } 7035 } 7036 if (SPECIFIED_CH(newpi->balance)) { 7037 error = au_set_gain(sc, &sc->sc_outports, 7038 pgain, newpi->balance); 7039 if (error) { 7040 device_printf(sc->sc_dev, 7041 "setting play.balance=%d failed with %d\n", 7042 newpi->balance, error); 7043 goto abort; 7044 } 7045 } 7046 if (SPECIFIED_CH(newri->balance)) { 7047 error = au_set_gain(sc, &sc->sc_inports, 7048 rgain, newri->balance); 7049 if (error) { 7050 device_printf(sc->sc_dev, 7051 "setting record.balance=%d failed with %d\n", 7052 newri->balance, error); 7053 goto abort; 7054 } 7055 } 7056 7057 if (SPECIFIED(newai->monitor_gain) && sc->sc_monitor_port != -1) { 7058 if (oldai) 7059 oldai->monitor_gain = au_get_monitor_gain(sc); 7060 error = au_set_monitor_gain(sc, newai->monitor_gain); 7061 if (error) { 7062 device_printf(sc->sc_dev, 7063 "setting monitor_gain=%d failed with %d\n", 7064 newai->monitor_gain, error); 7065 goto abort; 7066 } 7067 } 7068 7069 /* XXX TODO */ 7070 /* sc->sc_ai = *ai; */ 7071 7072 error = 0; 7073 abort: 7074 return error; 7075 } 7076 7077 /* 7078 * Setup the hardware with mixer format phwfmt, rhwfmt. 7079 * The arguments have following restrictions: 7080 * - setmode is the direction you want to set, AUMODE_PLAY or AUMODE_RECORD, 7081 * or both. 7082 * - phwfmt and rhwfmt must not be NULL regardless of setmode. 7083 * - On non-independent devices, phwfmt and rhwfmt must have the same 7084 * parameters. 7085 * - pfil and rfil must be zero-filled. 7086 * If successful, 7087 * - phwfmt, rhwfmt will be overwritten by hardware format. 7088 * - pfil, rfil will be filled with filter information specified by the 7089 * hardware driver. 7090 * and then returns 0. Otherwise returns errno. 7091 * Must be called with sc_lock held. 7092 */ 7093 static int 7094 audio_hw_set_format(struct audio_softc *sc, int setmode, 7095 audio_format2_t *phwfmt, audio_format2_t *rhwfmt, 7096 audio_filter_reg_t *pfil, audio_filter_reg_t *rfil) 7097 { 7098 audio_params_t pp, rp; 7099 int error; 7100 7101 KASSERT(mutex_owned(sc->sc_lock)); 7102 KASSERT(phwfmt != NULL); 7103 KASSERT(rhwfmt != NULL); 7104 7105 pp = format2_to_params(phwfmt); 7106 rp = format2_to_params(rhwfmt); 7107 7108 error = sc->hw_if->set_format(sc->hw_hdl, setmode, 7109 &pp, &rp, pfil, rfil); 7110 if (error) { 7111 device_printf(sc->sc_dev, 7112 "set_format failed with %d\n", error); 7113 return error; 7114 } 7115 7116 if (sc->hw_if->commit_settings) { 7117 error = sc->hw_if->commit_settings(sc->hw_hdl); 7118 if (error) { 7119 device_printf(sc->sc_dev, 7120 "commit_settings failed with %d\n", error); 7121 return error; 7122 } 7123 } 7124 7125 return 0; 7126 } 7127 7128 /* 7129 * Fill audio_info structure. If need_mixerinfo is true, it will also 7130 * fill the hardware mixer information. 7131 * Must be called with sc_lock held. 7132 * Must be called with sc_exlock held, in addition, if need_mixerinfo is 7133 * true. 7134 */ 7135 static int 7136 audiogetinfo(struct audio_softc *sc, struct audio_info *ai, int need_mixerinfo, 7137 audio_file_t *file) 7138 { 7139 struct audio_prinfo *ri, *pi; 7140 audio_track_t *track; 7141 audio_track_t *ptrack; 7142 audio_track_t *rtrack; 7143 int gain; 7144 7145 KASSERT(mutex_owned(sc->sc_lock)); 7146 7147 ri = &ai->record; 7148 pi = &ai->play; 7149 ptrack = file->ptrack; 7150 rtrack = file->rtrack; 7151 7152 memset(ai, 0, sizeof(*ai)); 7153 7154 if (ptrack) { 7155 pi->sample_rate = ptrack->usrbuf.fmt.sample_rate; 7156 pi->channels = ptrack->usrbuf.fmt.channels; 7157 pi->precision = ptrack->usrbuf.fmt.precision; 7158 pi->encoding = ptrack->usrbuf.fmt.encoding; 7159 } else { 7160 /* Set default parameters if the track is not available. */ 7161 if (ISDEVAUDIO(file->dev)) { 7162 pi->sample_rate = audio_default.sample_rate; 7163 pi->channels = audio_default.channels; 7164 pi->precision = audio_default.precision; 7165 pi->encoding = audio_default.encoding; 7166 } else { 7167 pi->sample_rate = sc->sc_sound_pparams.sample_rate; 7168 pi->channels = sc->sc_sound_pparams.channels; 7169 pi->precision = sc->sc_sound_pparams.precision; 7170 pi->encoding = sc->sc_sound_pparams.encoding; 7171 } 7172 } 7173 if (rtrack) { 7174 ri->sample_rate = rtrack->usrbuf.fmt.sample_rate; 7175 ri->channels = rtrack->usrbuf.fmt.channels; 7176 ri->precision = rtrack->usrbuf.fmt.precision; 7177 ri->encoding = rtrack->usrbuf.fmt.encoding; 7178 } else { 7179 /* Set default parameters if the track is not available. */ 7180 if (ISDEVAUDIO(file->dev)) { 7181 ri->sample_rate = audio_default.sample_rate; 7182 ri->channels = audio_default.channels; 7183 ri->precision = audio_default.precision; 7184 ri->encoding = audio_default.encoding; 7185 } else { 7186 ri->sample_rate = sc->sc_sound_rparams.sample_rate; 7187 ri->channels = sc->sc_sound_rparams.channels; 7188 ri->precision = sc->sc_sound_rparams.precision; 7189 ri->encoding = sc->sc_sound_rparams.encoding; 7190 } 7191 } 7192 7193 if (ptrack) { 7194 pi->seek = ptrack->usrbuf.used; 7195 pi->samples = ptrack->usrbuf_stamp; 7196 pi->eof = ptrack->eofcounter; 7197 pi->pause = ptrack->is_pause; 7198 pi->error = (ptrack->dropframes != 0) ? 1 : 0; 7199 pi->waiting = 0; /* open never hangs */ 7200 pi->open = 1; 7201 pi->active = sc->sc_pbusy; 7202 pi->buffer_size = ptrack->usrbuf.capacity; 7203 } 7204 if (rtrack) { 7205 ri->seek = rtrack->usrbuf.used; 7206 ri->samples = rtrack->usrbuf_stamp; 7207 ri->eof = 0; 7208 ri->pause = rtrack->is_pause; 7209 ri->error = (rtrack->dropframes != 0) ? 1 : 0; 7210 ri->waiting = 0; /* open never hangs */ 7211 ri->open = 1; 7212 ri->active = sc->sc_rbusy; 7213 ri->buffer_size = rtrack->usrbuf.capacity; 7214 } 7215 7216 /* 7217 * XXX There may be different number of channels between playback 7218 * and recording, so that blocksize also may be different. 7219 * But struct audio_info has an united blocksize... 7220 * Here, I use play info precedencely if ptrack is available, 7221 * otherwise record info. 7222 * 7223 * XXX hiwat/lowat is a playback-only parameter. What should I 7224 * return for a record-only descriptor? 7225 */ 7226 track = ptrack ? ptrack : rtrack; 7227 if (track) { 7228 ai->blocksize = track->usrbuf_blksize; 7229 ai->hiwat = track->usrbuf_usedhigh / track->usrbuf_blksize; 7230 ai->lowat = track->usrbuf_usedlow / track->usrbuf_blksize; 7231 } 7232 ai->mode = file->mode; 7233 7234 if (need_mixerinfo) { 7235 KASSERT(sc->sc_exlock); 7236 7237 pi->port = au_get_port(sc, &sc->sc_outports); 7238 ri->port = au_get_port(sc, &sc->sc_inports); 7239 7240 pi->avail_ports = sc->sc_outports.allports; 7241 ri->avail_ports = sc->sc_inports.allports; 7242 7243 au_get_gain(sc, &sc->sc_outports, &pi->gain, &pi->balance); 7244 au_get_gain(sc, &sc->sc_inports, &ri->gain, &ri->balance); 7245 7246 if (sc->sc_monitor_port != -1) { 7247 gain = au_get_monitor_gain(sc); 7248 if (gain != -1) 7249 ai->monitor_gain = gain; 7250 } 7251 } 7252 7253 return 0; 7254 } 7255 7256 /* 7257 * Must be called with sc_lock held. 7258 */ 7259 static int 7260 audio_get_props(struct audio_softc *sc) 7261 { 7262 const struct audio_hw_if *hw; 7263 int props; 7264 7265 KASSERT(mutex_owned(sc->sc_lock)); 7266 7267 hw = sc->hw_if; 7268 props = hw->get_props(sc->hw_hdl); 7269 7270 /* 7271 * For historical reasons, if neither playback nor capture 7272 * properties are reported, assume both are supported. 7273 * XXX Ideally (all) hardware driver should be updated... 7274 */ 7275 if ((props & (AUDIO_PROP_PLAYBACK|AUDIO_PROP_CAPTURE)) == 0) 7276 props |= (AUDIO_PROP_PLAYBACK | AUDIO_PROP_CAPTURE); 7277 7278 /* MMAP is now supported by upper layer. */ 7279 props |= AUDIO_PROP_MMAP; 7280 7281 return props; 7282 } 7283 7284 /* 7285 * Return true if playback is configured. 7286 * This function can be used after audioattach. 7287 */ 7288 static bool 7289 audio_can_playback(struct audio_softc *sc) 7290 { 7291 7292 return (sc->sc_pmixer != NULL); 7293 } 7294 7295 /* 7296 * Return true if recording is configured. 7297 * This function can be used after audioattach. 7298 */ 7299 static bool 7300 audio_can_capture(struct audio_softc *sc) 7301 { 7302 7303 return (sc->sc_rmixer != NULL); 7304 } 7305 7306 /* 7307 * Get the afp->index'th item from the valid one of format[]. 7308 * If found, stores it to afp->fmt and returns 0. Otherwise return EINVAL. 7309 * 7310 * This is common routines for query_format. 7311 * If your hardware driver has struct audio_format[], the simplest case 7312 * you can write your query_format interface as follows: 7313 * 7314 * struct audio_format foo_format[] = { ... }; 7315 * 7316 * int 7317 * foo_query_format(void *hdl, audio_format_query_t *afp) 7318 * { 7319 * return audio_query_format(foo_format, __arraycount(foo_format), afp); 7320 * } 7321 */ 7322 int 7323 audio_query_format(const struct audio_format *format, int nformats, 7324 audio_format_query_t *afp) 7325 { 7326 const struct audio_format *f; 7327 int idx; 7328 int i; 7329 7330 idx = 0; 7331 for (i = 0; i < nformats; i++) { 7332 f = &format[i]; 7333 if (!AUFMT_IS_VALID(f)) 7334 continue; 7335 if (afp->index == idx) { 7336 afp->fmt = *f; 7337 return 0; 7338 } 7339 idx++; 7340 } 7341 return EINVAL; 7342 } 7343 7344 /* 7345 * This function is provided for the hardware driver's set_format() to 7346 * find index matches with 'param' from array of audio_format_t 'formats'. 7347 * 'mode' is either of AUMODE_PLAY or AUMODE_RECORD. 7348 * It returns the matched index and never fails. Because param passed to 7349 * set_format() is selected from query_format(). 7350 * This function will be an alternative to auconv_set_converter() to 7351 * find index. 7352 */ 7353 int 7354 audio_indexof_format(const struct audio_format *formats, int nformats, 7355 int mode, const audio_params_t *param) 7356 { 7357 const struct audio_format *f; 7358 int index; 7359 int j; 7360 7361 for (index = 0; index < nformats; index++) { 7362 f = &formats[index]; 7363 7364 if (!AUFMT_IS_VALID(f)) 7365 continue; 7366 if ((f->mode & mode) == 0) 7367 continue; 7368 if (f->encoding != param->encoding) 7369 continue; 7370 if (f->validbits != param->precision) 7371 continue; 7372 if (f->channels != param->channels) 7373 continue; 7374 7375 if (f->frequency_type == 0) { 7376 if (param->sample_rate < f->frequency[0] || 7377 param->sample_rate > f->frequency[1]) 7378 continue; 7379 } else { 7380 for (j = 0; j < f->frequency_type; j++) { 7381 if (param->sample_rate == f->frequency[j]) 7382 break; 7383 } 7384 if (j == f->frequency_type) 7385 continue; 7386 } 7387 7388 /* Then, matched */ 7389 return index; 7390 } 7391 7392 /* Not matched. This should not be happened. */ 7393 panic("%s: cannot find matched format\n", __func__); 7394 } 7395 7396 /* 7397 * Get or set software master volume: 0..256 7398 * XXX It's for debug. 7399 */ 7400 static int 7401 audio_sysctl_volume(SYSCTLFN_ARGS) 7402 { 7403 struct sysctlnode node; 7404 struct audio_softc *sc; 7405 int t, error; 7406 7407 node = *rnode; 7408 sc = node.sysctl_data; 7409 7410 if (sc->sc_pmixer) 7411 t = sc->sc_pmixer->volume; 7412 else 7413 t = -1; 7414 node.sysctl_data = &t; 7415 error = sysctl_lookup(SYSCTLFN_CALL(&node)); 7416 if (error || newp == NULL) 7417 return error; 7418 7419 if (sc->sc_pmixer == NULL) 7420 return EINVAL; 7421 if (t < 0) 7422 return EINVAL; 7423 7424 sc->sc_pmixer->volume = t; 7425 return 0; 7426 } 7427 7428 /* 7429 * Get or set hardware blocksize in msec. 7430 * XXX It's for debug. 7431 */ 7432 static int 7433 audio_sysctl_blk_ms(SYSCTLFN_ARGS) 7434 { 7435 struct sysctlnode node; 7436 struct audio_softc *sc; 7437 audio_format2_t phwfmt; 7438 audio_format2_t rhwfmt; 7439 audio_filter_reg_t pfil; 7440 audio_filter_reg_t rfil; 7441 int t; 7442 int old_blk_ms; 7443 int mode; 7444 int error; 7445 7446 node = *rnode; 7447 sc = node.sysctl_data; 7448 7449 mutex_enter(sc->sc_lock); 7450 7451 old_blk_ms = sc->sc_blk_ms; 7452 t = old_blk_ms; 7453 node.sysctl_data = &t; 7454 error = sysctl_lookup(SYSCTLFN_CALL(&node)); 7455 if (error || newp == NULL) 7456 goto abort; 7457 7458 if (t < 0) { 7459 error = EINVAL; 7460 goto abort; 7461 } 7462 7463 if (sc->sc_popens + sc->sc_ropens > 0) { 7464 error = EBUSY; 7465 goto abort; 7466 } 7467 sc->sc_blk_ms = t; 7468 mode = 0; 7469 if (sc->sc_pmixer) { 7470 mode |= AUMODE_PLAY; 7471 phwfmt = sc->sc_pmixer->hwbuf.fmt; 7472 } 7473 if (sc->sc_rmixer) { 7474 mode |= AUMODE_RECORD; 7475 rhwfmt = sc->sc_rmixer->hwbuf.fmt; 7476 } 7477 7478 /* re-init hardware */ 7479 memset(&pfil, 0, sizeof(pfil)); 7480 memset(&rfil, 0, sizeof(rfil)); 7481 error = audio_hw_set_format(sc, mode, &phwfmt, &rhwfmt, &pfil, &rfil); 7482 if (error) { 7483 goto abort; 7484 } 7485 7486 /* re-init track mixer */ 7487 error = audio_mixers_init(sc, mode, &phwfmt, &rhwfmt, &pfil, &rfil); 7488 if (error) { 7489 /* Rollback */ 7490 sc->sc_blk_ms = old_blk_ms; 7491 audio_mixers_init(sc, mode, &phwfmt, &rhwfmt, &pfil, &rfil); 7492 goto abort; 7493 } 7494 error = 0; 7495 abort: 7496 mutex_exit(sc->sc_lock); 7497 return error; 7498 } 7499 7500 /* 7501 * Get or set multiuser mode. 7502 */ 7503 static int 7504 audio_sysctl_multiuser(SYSCTLFN_ARGS) 7505 { 7506 struct sysctlnode node; 7507 struct audio_softc *sc; 7508 int t, error; 7509 7510 node = *rnode; 7511 sc = node.sysctl_data; 7512 7513 mutex_enter(sc->sc_lock); 7514 7515 t = sc->sc_multiuser; 7516 node.sysctl_data = &t; 7517 error = sysctl_lookup(SYSCTLFN_CALL(&node)); 7518 if (error || newp == NULL) 7519 goto abort; 7520 7521 sc->sc_multiuser = t; 7522 error = 0; 7523 abort: 7524 mutex_exit(sc->sc_lock); 7525 return error; 7526 } 7527 7528 #if defined(AUDIO_DEBUG) 7529 /* 7530 * Get or set debug verbose level. (0..4) 7531 * XXX It's for debug. 7532 * XXX It is not separated per device. 7533 */ 7534 static int 7535 audio_sysctl_debug(SYSCTLFN_ARGS) 7536 { 7537 struct sysctlnode node; 7538 int t; 7539 int error; 7540 7541 node = *rnode; 7542 t = audiodebug; 7543 node.sysctl_data = &t; 7544 error = sysctl_lookup(SYSCTLFN_CALL(&node)); 7545 if (error || newp == NULL) 7546 return error; 7547 7548 if (t < 0 || t > 4) 7549 return EINVAL; 7550 audiodebug = t; 7551 printf("audio: audiodebug = %d\n", audiodebug); 7552 return 0; 7553 } 7554 #endif /* AUDIO_DEBUG */ 7555 7556 #ifdef AUDIO_PM_IDLE 7557 static void 7558 audio_idle(void *arg) 7559 { 7560 device_t dv = arg; 7561 struct audio_softc *sc = device_private(dv); 7562 7563 #ifdef PNP_DEBUG 7564 extern int pnp_debug_idle; 7565 if (pnp_debug_idle) 7566 printf("%s: idle handler called\n", device_xname(dv)); 7567 #endif 7568 7569 sc->sc_idle = true; 7570 7571 /* XXX joerg Make pmf_device_suspend handle children? */ 7572 if (!pmf_device_suspend(dv, PMF_Q_SELF)) 7573 return; 7574 7575 if (!pmf_device_suspend(sc->hw_dev, PMF_Q_SELF)) 7576 pmf_device_resume(dv, PMF_Q_SELF); 7577 } 7578 7579 static void 7580 audio_activity(device_t dv, devactive_t type) 7581 { 7582 struct audio_softc *sc = device_private(dv); 7583 7584 if (type != DVA_SYSTEM) 7585 return; 7586 7587 callout_schedule(&sc->sc_idle_counter, audio_idle_timeout * hz); 7588 7589 sc->sc_idle = false; 7590 if (!device_is_active(dv)) { 7591 /* XXX joerg How to deal with a failing resume... */ 7592 pmf_device_resume(sc->hw_dev, PMF_Q_SELF); 7593 pmf_device_resume(dv, PMF_Q_SELF); 7594 } 7595 } 7596 #endif 7597 7598 static bool 7599 audio_suspend(device_t dv, const pmf_qual_t *qual) 7600 { 7601 struct audio_softc *sc = device_private(dv); 7602 int error; 7603 7604 error = audio_enter_exclusive(sc); 7605 if (error) 7606 return error; 7607 audio_mixer_capture(sc); 7608 7609 /* Halts mixers but don't clear busy flag for resume */ 7610 if (sc->sc_pbusy) { 7611 audio_pmixer_halt(sc); 7612 sc->sc_pbusy = true; 7613 } 7614 if (sc->sc_rbusy) { 7615 audio_rmixer_halt(sc); 7616 sc->sc_rbusy = true; 7617 } 7618 7619 #ifdef AUDIO_PM_IDLE 7620 callout_halt(&sc->sc_idle_counter, sc->sc_lock); 7621 #endif 7622 audio_exit_exclusive(sc); 7623 7624 return true; 7625 } 7626 7627 static bool 7628 audio_resume(device_t dv, const pmf_qual_t *qual) 7629 { 7630 struct audio_softc *sc = device_private(dv); 7631 struct audio_info ai; 7632 int error; 7633 7634 error = audio_enter_exclusive(sc); 7635 if (error) 7636 return error; 7637 7638 audio_mixer_restore(sc); 7639 /* XXX ? */ 7640 AUDIO_INITINFO(&ai); 7641 audio_hw_setinfo(sc, &ai, NULL); 7642 7643 if (sc->sc_pbusy) 7644 audio_pmixer_start(sc, true); 7645 if (sc->sc_rbusy) 7646 audio_rmixer_start(sc); 7647 7648 audio_exit_exclusive(sc); 7649 7650 return true; 7651 } 7652 7653 #if defined(DIAGNOSTIC) || defined(AUDIO_DEBUG) 7654 static void 7655 audio_format2_tostr(char *buf, size_t bufsize, const audio_format2_t *fmt) 7656 { 7657 int n; 7658 7659 n = 0; 7660 n += snprintf(buf + n, bufsize - n, "%s", 7661 audio_encoding_name(fmt->encoding)); 7662 if (fmt->precision == fmt->stride) { 7663 n += snprintf(buf + n, bufsize - n, " %dbit", fmt->precision); 7664 } else { 7665 n += snprintf(buf + n, bufsize - n, " %d/%dbit", 7666 fmt->precision, fmt->stride); 7667 } 7668 7669 snprintf(buf + n, bufsize - n, " %uch %uHz", 7670 fmt->channels, fmt->sample_rate); 7671 } 7672 #endif 7673 7674 #if defined(AUDIO_DEBUG) 7675 static void 7676 audio_print_format2(const char *s, const audio_format2_t *fmt) 7677 { 7678 char fmtstr[64]; 7679 7680 audio_format2_tostr(fmtstr, sizeof(fmtstr), fmt); 7681 printf("%s %s\n", s, fmtstr); 7682 } 7683 #endif 7684 7685 #ifdef DIAGNOSTIC 7686 void 7687 audio_diagnostic_format2(const char *func, const audio_format2_t *fmt) 7688 { 7689 7690 KASSERTMSG(fmt, "%s: fmt == NULL", func); 7691 7692 /* XXX MSM6258 vs(4) only has 4bit stride format. */ 7693 if (fmt->encoding == AUDIO_ENCODING_ADPCM) { 7694 KASSERTMSG(fmt->stride == 4 || fmt->stride == 8, 7695 "%s: stride(%d) is invalid", func, fmt->stride); 7696 } else { 7697 KASSERTMSG(fmt->stride % NBBY == 0, 7698 "%s: stride(%d) is invalid", func, fmt->stride); 7699 } 7700 KASSERTMSG(fmt->precision <= fmt->stride, 7701 "%s: precision(%d) <= stride(%d)", 7702 func, fmt->precision, fmt->stride); 7703 KASSERTMSG(1 <= fmt->channels && fmt->channels <= AUDIO_MAX_CHANNELS, 7704 "%s: channels(%d) is out of range", 7705 func, fmt->channels); 7706 7707 /* XXX No check for encodings? */ 7708 } 7709 7710 void 7711 audio_diagnostic_filter_arg(const char *func, const audio_filter_arg_t *arg) 7712 { 7713 7714 KASSERT(arg != NULL); 7715 KASSERT(arg->src != NULL); 7716 KASSERT(arg->dst != NULL); 7717 DIAGNOSTIC_format2(arg->srcfmt); 7718 DIAGNOSTIC_format2(arg->dstfmt); 7719 KASSERTMSG(arg->count > 0, 7720 "%s: count(%d) is out of range", func, arg->count); 7721 } 7722 7723 void 7724 audio_diagnostic_ring(const char *func, const audio_ring_t *ring) 7725 { 7726 7727 KASSERTMSG(ring, "%s: ring == NULL", func); 7728 DIAGNOSTIC_format2(&ring->fmt); 7729 KASSERTMSG(0 <= ring->capacity && ring->capacity < INT_MAX / 2, 7730 "%s: capacity(%d) is out of range", func, ring->capacity); 7731 KASSERTMSG(0 <= ring->used && ring->used <= ring->capacity, 7732 "%s: used(%d) is out of range (capacity:%d)", 7733 func, ring->used, ring->capacity); 7734 if (ring->capacity == 0) { 7735 KASSERTMSG(ring->mem == NULL, 7736 "%s: capacity == 0 but mem != NULL", func); 7737 } else { 7738 KASSERTMSG(ring->mem != NULL, 7739 "%s: capacity != 0 but mem == NULL", func); 7740 KASSERTMSG(0 <= ring->head && ring->head < ring->capacity, 7741 "%s: head(%d) is out of range (capacity:%d)", 7742 func, ring->head, ring->capacity); 7743 } 7744 } 7745 #endif /* DIAGNOSTIC */ 7746 7747 7748 /* 7749 * Mixer driver 7750 */ 7751 int 7752 mixer_open(dev_t dev, struct audio_softc *sc, int flags, int ifmt, 7753 struct lwp *l) 7754 { 7755 struct file *fp; 7756 audio_file_t *af; 7757 int error, fd; 7758 7759 KASSERT(mutex_owned(sc->sc_lock)); 7760 7761 TRACE(1, "flags=0x%x", flags); 7762 7763 error = fd_allocfile(&fp, &fd); 7764 if (error) 7765 return error; 7766 7767 af = kmem_zalloc(sizeof(*af), KM_SLEEP); 7768 af->sc = sc; 7769 af->dev = dev; 7770 7771 error = fd_clone(fp, fd, flags, &audio_fileops, af); 7772 KASSERT(error == EMOVEFD); 7773 7774 return error; 7775 } 7776 7777 /* 7778 * Remove a process from those to be signalled on mixer activity. 7779 * Must be called with sc_lock held. 7780 */ 7781 static void 7782 mixer_remove(struct audio_softc *sc) 7783 { 7784 struct mixer_asyncs **pm, *m; 7785 pid_t pid; 7786 7787 KASSERT(mutex_owned(sc->sc_lock)); 7788 7789 pid = curproc->p_pid; 7790 for (pm = &sc->sc_async_mixer; *pm; pm = &(*pm)->next) { 7791 if ((*pm)->pid == pid) { 7792 m = *pm; 7793 *pm = m->next; 7794 kmem_free(m, sizeof(*m)); 7795 return; 7796 } 7797 } 7798 } 7799 7800 /* 7801 * Signal all processes waiting for the mixer. 7802 * Must be called with sc_lock held. 7803 */ 7804 static void 7805 mixer_signal(struct audio_softc *sc) 7806 { 7807 struct mixer_asyncs *m; 7808 proc_t *p; 7809 7810 for (m = sc->sc_async_mixer; m; m = m->next) { 7811 mutex_enter(proc_lock); 7812 if ((p = proc_find(m->pid)) != NULL) 7813 psignal(p, SIGIO); 7814 mutex_exit(proc_lock); 7815 } 7816 } 7817 7818 /* 7819 * Close a mixer device 7820 */ 7821 int 7822 mixer_close(struct audio_softc *sc, audio_file_t *file) 7823 { 7824 7825 mutex_enter(sc->sc_lock); 7826 TRACE(1, ""); 7827 mixer_remove(sc); 7828 mutex_exit(sc->sc_lock); 7829 7830 return 0; 7831 } 7832 7833 int 7834 mixer_ioctl(struct audio_softc *sc, u_long cmd, void *addr, int flag, 7835 struct lwp *l) 7836 { 7837 struct mixer_asyncs *ma; 7838 mixer_devinfo_t *mi; 7839 mixer_ctrl_t *mc; 7840 int error; 7841 7842 KASSERT(!mutex_owned(sc->sc_lock)); 7843 7844 TRACE(2, "(%lu,'%c',%lu)", 7845 IOCPARM_LEN(cmd), (char)IOCGROUP(cmd), cmd & 0xff); 7846 error = EINVAL; 7847 7848 /* we can return cached values if we are sleeping */ 7849 if (cmd != AUDIO_MIXER_READ) { 7850 mutex_enter(sc->sc_lock); 7851 device_active(sc->sc_dev, DVA_SYSTEM); 7852 mutex_exit(sc->sc_lock); 7853 } 7854 7855 switch (cmd) { 7856 case FIOASYNC: 7857 if (*(int *)addr) { 7858 ma = kmem_alloc(sizeof(struct mixer_asyncs), KM_SLEEP); 7859 } else { 7860 ma = NULL; 7861 } 7862 mixer_remove(sc); /* remove old entry */ 7863 if (ma != NULL) { 7864 ma->next = sc->sc_async_mixer; 7865 ma->pid = curproc->p_pid; 7866 sc->sc_async_mixer = ma; 7867 } 7868 error = 0; 7869 break; 7870 7871 case AUDIO_GETDEV: 7872 TRACE(2, "AUDIO_GETDEV"); 7873 error = audio_enter_exclusive(sc); 7874 if (error) 7875 break; 7876 error = sc->hw_if->getdev(sc->hw_hdl, (audio_device_t *)addr); 7877 audio_exit_exclusive(sc); 7878 break; 7879 7880 case AUDIO_MIXER_DEVINFO: 7881 TRACE(2, "AUDIO_MIXER_DEVINFO"); 7882 mi = (mixer_devinfo_t *)addr; 7883 7884 mi->un.v.delta = 0; /* default */ 7885 mutex_enter(sc->sc_lock); 7886 error = audio_query_devinfo(sc, mi); 7887 mutex_exit(sc->sc_lock); 7888 break; 7889 7890 case AUDIO_MIXER_READ: 7891 TRACE(2, "AUDIO_MIXER_READ"); 7892 mc = (mixer_ctrl_t *)addr; 7893 7894 error = audio_enter_exclusive(sc); 7895 if (error) 7896 break; 7897 if (device_is_active(sc->hw_dev)) 7898 error = audio_get_port(sc, mc); 7899 else if (mc->dev < 0 || mc->dev >= sc->sc_nmixer_states) 7900 error = ENXIO; 7901 else { 7902 int dev = mc->dev; 7903 memcpy(mc, &sc->sc_mixer_state[dev], 7904 sizeof(mixer_ctrl_t)); 7905 error = 0; 7906 } 7907 audio_exit_exclusive(sc); 7908 break; 7909 7910 case AUDIO_MIXER_WRITE: 7911 TRACE(2, "AUDIO_MIXER_WRITE"); 7912 error = audio_enter_exclusive(sc); 7913 if (error) 7914 break; 7915 error = audio_set_port(sc, (mixer_ctrl_t *)addr); 7916 if (error) { 7917 audio_exit_exclusive(sc); 7918 break; 7919 } 7920 7921 if (sc->hw_if->commit_settings) { 7922 error = sc->hw_if->commit_settings(sc->hw_hdl); 7923 if (error) { 7924 audio_exit_exclusive(sc); 7925 break; 7926 } 7927 } 7928 mixer_signal(sc); 7929 audio_exit_exclusive(sc); 7930 break; 7931 7932 default: 7933 if (sc->hw_if->dev_ioctl) { 7934 error = audio_enter_exclusive(sc); 7935 if (error) 7936 break; 7937 error = sc->hw_if->dev_ioctl(sc->hw_hdl, 7938 cmd, addr, flag, l); 7939 audio_exit_exclusive(sc); 7940 } else 7941 error = EINVAL; 7942 break; 7943 } 7944 TRACE(2, "(%lu,'%c',%lu) result %d", 7945 IOCPARM_LEN(cmd), (char)IOCGROUP(cmd), cmd & 0xff, error); 7946 return error; 7947 } 7948 7949 /* 7950 * Must be called with sc_lock held. 7951 */ 7952 int 7953 au_portof(struct audio_softc *sc, char *name, int class) 7954 { 7955 mixer_devinfo_t mi; 7956 7957 KASSERT(mutex_owned(sc->sc_lock)); 7958 7959 for (mi.index = 0; audio_query_devinfo(sc, &mi) == 0; mi.index++) { 7960 if (mi.mixer_class == class && strcmp(mi.label.name, name) == 0) 7961 return mi.index; 7962 } 7963 return -1; 7964 } 7965 7966 /* 7967 * Must be called with sc_lock held. 7968 */ 7969 void 7970 au_setup_ports(struct audio_softc *sc, struct au_mixer_ports *ports, 7971 mixer_devinfo_t *mi, const struct portname *tbl) 7972 { 7973 int i, j; 7974 7975 KASSERT(mutex_owned(sc->sc_lock)); 7976 7977 ports->index = mi->index; 7978 if (mi->type == AUDIO_MIXER_ENUM) { 7979 ports->isenum = true; 7980 for(i = 0; tbl[i].name; i++) 7981 for(j = 0; j < mi->un.e.num_mem; j++) 7982 if (strcmp(mi->un.e.member[j].label.name, 7983 tbl[i].name) == 0) { 7984 ports->allports |= tbl[i].mask; 7985 ports->aumask[ports->nports] = tbl[i].mask; 7986 ports->misel[ports->nports] = 7987 mi->un.e.member[j].ord; 7988 ports->miport[ports->nports] = 7989 au_portof(sc, mi->un.e.member[j].label.name, 7990 mi->mixer_class); 7991 if (ports->mixerout != -1 && 7992 ports->miport[ports->nports] != -1) 7993 ports->isdual = true; 7994 ++ports->nports; 7995 } 7996 } else if (mi->type == AUDIO_MIXER_SET) { 7997 for(i = 0; tbl[i].name; i++) 7998 for(j = 0; j < mi->un.s.num_mem; j++) 7999 if (strcmp(mi->un.s.member[j].label.name, 8000 tbl[i].name) == 0) { 8001 ports->allports |= tbl[i].mask; 8002 ports->aumask[ports->nports] = tbl[i].mask; 8003 ports->misel[ports->nports] = 8004 mi->un.s.member[j].mask; 8005 ports->miport[ports->nports] = 8006 au_portof(sc, mi->un.s.member[j].label.name, 8007 mi->mixer_class); 8008 ++ports->nports; 8009 } 8010 } 8011 } 8012 8013 /* 8014 * Must be called with sc_lock && sc_exlock held. 8015 */ 8016 int 8017 au_set_lr_value(struct audio_softc *sc, mixer_ctrl_t *ct, int l, int r) 8018 { 8019 8020 KASSERT(mutex_owned(sc->sc_lock)); 8021 KASSERT(sc->sc_exlock); 8022 8023 ct->type = AUDIO_MIXER_VALUE; 8024 ct->un.value.num_channels = 2; 8025 ct->un.value.level[AUDIO_MIXER_LEVEL_LEFT] = l; 8026 ct->un.value.level[AUDIO_MIXER_LEVEL_RIGHT] = r; 8027 if (audio_set_port(sc, ct) == 0) 8028 return 0; 8029 ct->un.value.num_channels = 1; 8030 ct->un.value.level[AUDIO_MIXER_LEVEL_MONO] = (l+r)/2; 8031 return audio_set_port(sc, ct); 8032 } 8033 8034 /* 8035 * Must be called with sc_lock && sc_exlock held. 8036 */ 8037 int 8038 au_get_lr_value(struct audio_softc *sc, mixer_ctrl_t *ct, int *l, int *r) 8039 { 8040 int error; 8041 8042 KASSERT(mutex_owned(sc->sc_lock)); 8043 KASSERT(sc->sc_exlock); 8044 8045 ct->un.value.num_channels = 2; 8046 if (audio_get_port(sc, ct) == 0) { 8047 *l = ct->un.value.level[AUDIO_MIXER_LEVEL_LEFT]; 8048 *r = ct->un.value.level[AUDIO_MIXER_LEVEL_RIGHT]; 8049 } else { 8050 ct->un.value.num_channels = 1; 8051 error = audio_get_port(sc, ct); 8052 if (error) 8053 return error; 8054 *r = *l = ct->un.value.level[AUDIO_MIXER_LEVEL_MONO]; 8055 } 8056 return 0; 8057 } 8058 8059 /* 8060 * Must be called with sc_lock && sc_exlock held. 8061 */ 8062 int 8063 au_set_gain(struct audio_softc *sc, struct au_mixer_ports *ports, 8064 int gain, int balance) 8065 { 8066 mixer_ctrl_t ct; 8067 int i, error; 8068 int l, r; 8069 u_int mask; 8070 int nset; 8071 8072 KASSERT(mutex_owned(sc->sc_lock)); 8073 KASSERT(sc->sc_exlock); 8074 8075 if (balance == AUDIO_MID_BALANCE) { 8076 l = r = gain; 8077 } else if (balance < AUDIO_MID_BALANCE) { 8078 l = gain; 8079 r = (balance * gain) / AUDIO_MID_BALANCE; 8080 } else { 8081 r = gain; 8082 l = ((AUDIO_RIGHT_BALANCE - balance) * gain) 8083 / AUDIO_MID_BALANCE; 8084 } 8085 TRACE(2, "gain=%d balance=%d, l=%d r=%d", gain, balance, l, r); 8086 8087 if (ports->index == -1) { 8088 usemaster: 8089 if (ports->master == -1) 8090 return 0; /* just ignore it silently */ 8091 ct.dev = ports->master; 8092 error = au_set_lr_value(sc, &ct, l, r); 8093 } else { 8094 ct.dev = ports->index; 8095 if (ports->isenum) { 8096 ct.type = AUDIO_MIXER_ENUM; 8097 error = audio_get_port(sc, &ct); 8098 if (error) 8099 return error; 8100 if (ports->isdual) { 8101 if (ports->cur_port == -1) 8102 ct.dev = ports->master; 8103 else 8104 ct.dev = ports->miport[ports->cur_port]; 8105 error = au_set_lr_value(sc, &ct, l, r); 8106 } else { 8107 for(i = 0; i < ports->nports; i++) 8108 if (ports->misel[i] == ct.un.ord) { 8109 ct.dev = ports->miport[i]; 8110 if (ct.dev == -1 || 8111 au_set_lr_value(sc, &ct, l, r)) 8112 goto usemaster; 8113 else 8114 break; 8115 } 8116 } 8117 } else { 8118 ct.type = AUDIO_MIXER_SET; 8119 error = audio_get_port(sc, &ct); 8120 if (error) 8121 return error; 8122 mask = ct.un.mask; 8123 nset = 0; 8124 for(i = 0; i < ports->nports; i++) { 8125 if (ports->misel[i] & mask) { 8126 ct.dev = ports->miport[i]; 8127 if (ct.dev != -1 && 8128 au_set_lr_value(sc, &ct, l, r) == 0) 8129 nset++; 8130 } 8131 } 8132 if (nset == 0) 8133 goto usemaster; 8134 } 8135 } 8136 if (!error) 8137 mixer_signal(sc); 8138 return error; 8139 } 8140 8141 /* 8142 * Must be called with sc_lock && sc_exlock held. 8143 */ 8144 void 8145 au_get_gain(struct audio_softc *sc, struct au_mixer_ports *ports, 8146 u_int *pgain, u_char *pbalance) 8147 { 8148 mixer_ctrl_t ct; 8149 int i, l, r, n; 8150 int lgain, rgain; 8151 8152 KASSERT(mutex_owned(sc->sc_lock)); 8153 KASSERT(sc->sc_exlock); 8154 8155 lgain = AUDIO_MAX_GAIN / 2; 8156 rgain = AUDIO_MAX_GAIN / 2; 8157 if (ports->index == -1) { 8158 usemaster: 8159 if (ports->master == -1) 8160 goto bad; 8161 ct.dev = ports->master; 8162 ct.type = AUDIO_MIXER_VALUE; 8163 if (au_get_lr_value(sc, &ct, &lgain, &rgain)) 8164 goto bad; 8165 } else { 8166 ct.dev = ports->index; 8167 if (ports->isenum) { 8168 ct.type = AUDIO_MIXER_ENUM; 8169 if (audio_get_port(sc, &ct)) 8170 goto bad; 8171 ct.type = AUDIO_MIXER_VALUE; 8172 if (ports->isdual) { 8173 if (ports->cur_port == -1) 8174 ct.dev = ports->master; 8175 else 8176 ct.dev = ports->miport[ports->cur_port]; 8177 au_get_lr_value(sc, &ct, &lgain, &rgain); 8178 } else { 8179 for(i = 0; i < ports->nports; i++) 8180 if (ports->misel[i] == ct.un.ord) { 8181 ct.dev = ports->miport[i]; 8182 if (ct.dev == -1 || 8183 au_get_lr_value(sc, &ct, 8184 &lgain, &rgain)) 8185 goto usemaster; 8186 else 8187 break; 8188 } 8189 } 8190 } else { 8191 ct.type = AUDIO_MIXER_SET; 8192 if (audio_get_port(sc, &ct)) 8193 goto bad; 8194 ct.type = AUDIO_MIXER_VALUE; 8195 lgain = rgain = n = 0; 8196 for(i = 0; i < ports->nports; i++) { 8197 if (ports->misel[i] & ct.un.mask) { 8198 ct.dev = ports->miport[i]; 8199 if (ct.dev == -1 || 8200 au_get_lr_value(sc, &ct, &l, &r)) 8201 goto usemaster; 8202 else { 8203 lgain += l; 8204 rgain += r; 8205 n++; 8206 } 8207 } 8208 } 8209 if (n != 0) { 8210 lgain /= n; 8211 rgain /= n; 8212 } 8213 } 8214 } 8215 bad: 8216 if (lgain == rgain) { /* handles lgain==rgain==0 */ 8217 *pgain = lgain; 8218 *pbalance = AUDIO_MID_BALANCE; 8219 } else if (lgain < rgain) { 8220 *pgain = rgain; 8221 /* balance should be > AUDIO_MID_BALANCE */ 8222 *pbalance = AUDIO_RIGHT_BALANCE - 8223 (AUDIO_MID_BALANCE * lgain) / rgain; 8224 } else /* lgain > rgain */ { 8225 *pgain = lgain; 8226 /* balance should be < AUDIO_MID_BALANCE */ 8227 *pbalance = (AUDIO_MID_BALANCE * rgain) / lgain; 8228 } 8229 } 8230 8231 /* 8232 * Must be called with sc_lock && sc_exlock held. 8233 */ 8234 int 8235 au_set_port(struct audio_softc *sc, struct au_mixer_ports *ports, u_int port) 8236 { 8237 mixer_ctrl_t ct; 8238 int i, error, use_mixerout; 8239 8240 KASSERT(mutex_owned(sc->sc_lock)); 8241 KASSERT(sc->sc_exlock); 8242 8243 use_mixerout = 1; 8244 if (port == 0) { 8245 if (ports->allports == 0) 8246 return 0; /* Allow this special case. */ 8247 else if (ports->isdual) { 8248 if (ports->cur_port == -1) { 8249 return 0; 8250 } else { 8251 port = ports->aumask[ports->cur_port]; 8252 ports->cur_port = -1; 8253 use_mixerout = 0; 8254 } 8255 } 8256 } 8257 if (ports->index == -1) 8258 return EINVAL; 8259 ct.dev = ports->index; 8260 if (ports->isenum) { 8261 if (port & (port-1)) 8262 return EINVAL; /* Only one port allowed */ 8263 ct.type = AUDIO_MIXER_ENUM; 8264 error = EINVAL; 8265 for(i = 0; i < ports->nports; i++) 8266 if (ports->aumask[i] == port) { 8267 if (ports->isdual && use_mixerout) { 8268 ct.un.ord = ports->mixerout; 8269 ports->cur_port = i; 8270 } else { 8271 ct.un.ord = ports->misel[i]; 8272 } 8273 error = audio_set_port(sc, &ct); 8274 break; 8275 } 8276 } else { 8277 ct.type = AUDIO_MIXER_SET; 8278 ct.un.mask = 0; 8279 for(i = 0; i < ports->nports; i++) 8280 if (ports->aumask[i] & port) 8281 ct.un.mask |= ports->misel[i]; 8282 if (port != 0 && ct.un.mask == 0) 8283 error = EINVAL; 8284 else 8285 error = audio_set_port(sc, &ct); 8286 } 8287 if (!error) 8288 mixer_signal(sc); 8289 return error; 8290 } 8291 8292 /* 8293 * Must be called with sc_lock && sc_exlock held. 8294 */ 8295 int 8296 au_get_port(struct audio_softc *sc, struct au_mixer_ports *ports) 8297 { 8298 mixer_ctrl_t ct; 8299 int i, aumask; 8300 8301 KASSERT(mutex_owned(sc->sc_lock)); 8302 KASSERT(sc->sc_exlock); 8303 8304 if (ports->index == -1) 8305 return 0; 8306 ct.dev = ports->index; 8307 ct.type = ports->isenum ? AUDIO_MIXER_ENUM : AUDIO_MIXER_SET; 8308 if (audio_get_port(sc, &ct)) 8309 return 0; 8310 aumask = 0; 8311 if (ports->isenum) { 8312 if (ports->isdual && ports->cur_port != -1) { 8313 if (ports->mixerout == ct.un.ord) 8314 aumask = ports->aumask[ports->cur_port]; 8315 else 8316 ports->cur_port = -1; 8317 } 8318 if (aumask == 0) 8319 for(i = 0; i < ports->nports; i++) 8320 if (ports->misel[i] == ct.un.ord) 8321 aumask = ports->aumask[i]; 8322 } else { 8323 for(i = 0; i < ports->nports; i++) 8324 if (ct.un.mask & ports->misel[i]) 8325 aumask |= ports->aumask[i]; 8326 } 8327 return aumask; 8328 } 8329 8330 /* 8331 * It returns 0 if success, otherwise errno. 8332 * Must be called only if sc->sc_monitor_port != -1. 8333 * Must be called with sc_lock && sc_exlock held. 8334 */ 8335 static int 8336 au_set_monitor_gain(struct audio_softc *sc, int monitor_gain) 8337 { 8338 mixer_ctrl_t ct; 8339 8340 KASSERT(mutex_owned(sc->sc_lock)); 8341 KASSERT(sc->sc_exlock); 8342 8343 ct.dev = sc->sc_monitor_port; 8344 ct.type = AUDIO_MIXER_VALUE; 8345 ct.un.value.num_channels = 1; 8346 ct.un.value.level[AUDIO_MIXER_LEVEL_MONO] = monitor_gain; 8347 return audio_set_port(sc, &ct); 8348 } 8349 8350 /* 8351 * It returns monitor gain if success, otherwise -1. 8352 * Must be called only if sc->sc_monitor_port != -1. 8353 * Must be called with sc_lock && sc_exlock held. 8354 */ 8355 static int 8356 au_get_monitor_gain(struct audio_softc *sc) 8357 { 8358 mixer_ctrl_t ct; 8359 8360 KASSERT(mutex_owned(sc->sc_lock)); 8361 KASSERT(sc->sc_exlock); 8362 8363 ct.dev = sc->sc_monitor_port; 8364 ct.type = AUDIO_MIXER_VALUE; 8365 ct.un.value.num_channels = 1; 8366 if (audio_get_port(sc, &ct)) 8367 return -1; 8368 return ct.un.value.level[AUDIO_MIXER_LEVEL_MONO]; 8369 } 8370 8371 /* 8372 * Must be called with sc_lock && sc_exlock held. 8373 */ 8374 static int 8375 audio_set_port(struct audio_softc *sc, mixer_ctrl_t *mc) 8376 { 8377 8378 KASSERT(mutex_owned(sc->sc_lock)); 8379 KASSERT(sc->sc_exlock); 8380 8381 return sc->hw_if->set_port(sc->hw_hdl, mc); 8382 } 8383 8384 /* 8385 * Must be called with sc_lock && sc_exlock held. 8386 */ 8387 static int 8388 audio_get_port(struct audio_softc *sc, mixer_ctrl_t *mc) 8389 { 8390 8391 KASSERT(mutex_owned(sc->sc_lock)); 8392 KASSERT(sc->sc_exlock); 8393 8394 return sc->hw_if->get_port(sc->hw_hdl, mc); 8395 } 8396 8397 /* 8398 * Must be called with sc_lock && sc_exlock held. 8399 */ 8400 static void 8401 audio_mixer_capture(struct audio_softc *sc) 8402 { 8403 mixer_devinfo_t mi; 8404 mixer_ctrl_t *mc; 8405 8406 KASSERT(mutex_owned(sc->sc_lock)); 8407 KASSERT(sc->sc_exlock); 8408 8409 for (mi.index = 0;; mi.index++) { 8410 if (audio_query_devinfo(sc, &mi) != 0) 8411 break; 8412 KASSERT(mi.index < sc->sc_nmixer_states); 8413 if (mi.type == AUDIO_MIXER_CLASS) 8414 continue; 8415 mc = &sc->sc_mixer_state[mi.index]; 8416 mc->dev = mi.index; 8417 mc->type = mi.type; 8418 mc->un.value.num_channels = mi.un.v.num_channels; 8419 (void)audio_get_port(sc, mc); 8420 } 8421 8422 return; 8423 } 8424 8425 /* 8426 * Must be called with sc_lock && sc_exlock held. 8427 */ 8428 static void 8429 audio_mixer_restore(struct audio_softc *sc) 8430 { 8431 mixer_devinfo_t mi; 8432 mixer_ctrl_t *mc; 8433 8434 KASSERT(mutex_owned(sc->sc_lock)); 8435 KASSERT(sc->sc_exlock); 8436 8437 for (mi.index = 0; ; mi.index++) { 8438 if (audio_query_devinfo(sc, &mi) != 0) 8439 break; 8440 if (mi.type == AUDIO_MIXER_CLASS) 8441 continue; 8442 mc = &sc->sc_mixer_state[mi.index]; 8443 (void)audio_set_port(sc, mc); 8444 } 8445 if (sc->hw_if->commit_settings) 8446 sc->hw_if->commit_settings(sc->hw_hdl); 8447 8448 return; 8449 } 8450 8451 static void 8452 audio_volume_down(device_t dv) 8453 { 8454 struct audio_softc *sc = device_private(dv); 8455 mixer_devinfo_t mi; 8456 int newgain; 8457 u_int gain; 8458 u_char balance; 8459 8460 if (audio_enter_exclusive(sc) != 0) 8461 return; 8462 if (sc->sc_outports.index == -1 && sc->sc_outports.master != -1) { 8463 mi.index = sc->sc_outports.master; 8464 mi.un.v.delta = 0; 8465 if (audio_query_devinfo(sc, &mi) == 0) { 8466 au_get_gain(sc, &sc->sc_outports, &gain, &balance); 8467 newgain = gain - mi.un.v.delta; 8468 if (newgain < AUDIO_MIN_GAIN) 8469 newgain = AUDIO_MIN_GAIN; 8470 au_set_gain(sc, &sc->sc_outports, newgain, balance); 8471 } 8472 } 8473 audio_exit_exclusive(sc); 8474 } 8475 8476 static void 8477 audio_volume_up(device_t dv) 8478 { 8479 struct audio_softc *sc = device_private(dv); 8480 mixer_devinfo_t mi; 8481 u_int gain, newgain; 8482 u_char balance; 8483 8484 if (audio_enter_exclusive(sc) != 0) 8485 return; 8486 if (sc->sc_outports.index == -1 && sc->sc_outports.master != -1) { 8487 mi.index = sc->sc_outports.master; 8488 mi.un.v.delta = 0; 8489 if (audio_query_devinfo(sc, &mi) == 0) { 8490 au_get_gain(sc, &sc->sc_outports, &gain, &balance); 8491 newgain = gain + mi.un.v.delta; 8492 if (newgain > AUDIO_MAX_GAIN) 8493 newgain = AUDIO_MAX_GAIN; 8494 au_set_gain(sc, &sc->sc_outports, newgain, balance); 8495 } 8496 } 8497 audio_exit_exclusive(sc); 8498 } 8499 8500 static void 8501 audio_volume_toggle(device_t dv) 8502 { 8503 struct audio_softc *sc = device_private(dv); 8504 u_int gain, newgain; 8505 u_char balance; 8506 8507 if (audio_enter_exclusive(sc) != 0) 8508 return; 8509 au_get_gain(sc, &sc->sc_outports, &gain, &balance); 8510 if (gain != 0) { 8511 sc->sc_lastgain = gain; 8512 newgain = 0; 8513 } else 8514 newgain = sc->sc_lastgain; 8515 au_set_gain(sc, &sc->sc_outports, newgain, balance); 8516 audio_exit_exclusive(sc); 8517 } 8518 8519 static int 8520 audio_query_devinfo(struct audio_softc *sc, mixer_devinfo_t *di) 8521 { 8522 8523 KASSERT(mutex_owned(sc->sc_lock)); 8524 8525 return sc->hw_if->query_devinfo(sc->hw_hdl, di); 8526 } 8527 8528 #endif /* NAUDIO > 0 */ 8529 8530 #if NAUDIO == 0 && (NMIDI > 0 || NMIDIBUS > 0) 8531 #include <sys/param.h> 8532 #include <sys/systm.h> 8533 #include <sys/device.h> 8534 #include <sys/audioio.h> 8535 #include <dev/audio/audio_if.h> 8536 #endif 8537 8538 #if NAUDIO > 0 || (NMIDI > 0 || NMIDIBUS > 0) 8539 int 8540 audioprint(void *aux, const char *pnp) 8541 { 8542 struct audio_attach_args *arg; 8543 const char *type; 8544 8545 if (pnp != NULL) { 8546 arg = aux; 8547 switch (arg->type) { 8548 case AUDIODEV_TYPE_AUDIO: 8549 type = "audio"; 8550 break; 8551 case AUDIODEV_TYPE_MIDI: 8552 type = "midi"; 8553 break; 8554 case AUDIODEV_TYPE_OPL: 8555 type = "opl"; 8556 break; 8557 case AUDIODEV_TYPE_MPU: 8558 type = "mpu"; 8559 break; 8560 default: 8561 panic("audioprint: unknown type %d", arg->type); 8562 } 8563 aprint_normal("%s at %s", type, pnp); 8564 } 8565 return UNCONF; 8566 } 8567 8568 #endif /* NAUDIO > 0 || (NMIDI > 0 || NMIDIBUS > 0) */ 8569 8570 #ifdef _MODULE 8571 8572 devmajor_t audio_bmajor = -1, audio_cmajor = -1; 8573 8574 #include "ioconf.c" 8575 8576 #endif 8577 8578 MODULE(MODULE_CLASS_DRIVER, audio, NULL); 8579 8580 static int 8581 audio_modcmd(modcmd_t cmd, void *arg) 8582 { 8583 int error = 0; 8584 8585 #ifdef _MODULE 8586 switch (cmd) { 8587 case MODULE_CMD_INIT: 8588 error = devsw_attach(audio_cd.cd_name, NULL, &audio_bmajor, 8589 &audio_cdevsw, &audio_cmajor); 8590 if (error) 8591 break; 8592 8593 error = config_init_component(cfdriver_ioconf_audio, 8594 cfattach_ioconf_audio, cfdata_ioconf_audio); 8595 if (error) { 8596 devsw_detach(NULL, &audio_cdevsw); 8597 } 8598 break; 8599 case MODULE_CMD_FINI: 8600 devsw_detach(NULL, &audio_cdevsw); 8601 error = config_fini_component(cfdriver_ioconf_audio, 8602 cfattach_ioconf_audio, cfdata_ioconf_audio); 8603 if (error) 8604 devsw_attach(audio_cd.cd_name, NULL, &audio_bmajor, 8605 &audio_cdevsw, &audio_cmajor); 8606 break; 8607 default: 8608 error = ENOTTY; 8609 break; 8610 } 8611 #endif 8612 8613 return error; 8614 } 8615
Indexes created Wed Sep 24 15:09:51 GMT 2025