audio.c revision 1.45
1 /* $NetBSD: audio.c,v 1.45 2020/02/22 06:28:10 isaki Exp $ */ 2 3 /*- 4 * Copyright (c) 2008 The NetBSD Foundation, Inc. 5 * All rights reserved. 6 * 7 * This code is derived from software contributed to The NetBSD Foundation 8 * by Andrew Doran. 9 * 10 * Redistribution and use in source and binary forms, with or without 11 * modification, are permitted provided that the following conditions 12 * are met: 13 * 1. Redistributions of source code must retain the above copyright 14 * notice, this list of conditions and the following disclaimer. 15 * 2. Redistributions in binary form must reproduce the above copyright 16 * notice, this list of conditions and the following disclaimer in the 17 * documentation and/or other materials provided with the distribution. 18 * 19 * THIS SOFTWARE IS PROVIDED BY THE NETBSD FOUNDATION, INC. AND CONTRIBUTORS 20 * ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED 21 * TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR 22 * PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE FOUNDATION OR CONTRIBUTORS 23 * BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR 24 * CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF 25 * SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS 26 * INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN 27 * CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) 28 * ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE 29 * POSSIBILITY OF SUCH DAMAGE. 30 */ 31 32 /* 33 * Copyright (c) 1991-1993 Regents of the University of California. 34 * All rights reserved. 35 * 36 * Redistribution and use in source and binary forms, with or without 37 * modification, are permitted provided that the following conditions 38 * are met: 39 * 1. Redistributions of source code must retain the above copyright 40 * notice, this list of conditions and the following disclaimer. 41 * 2. Redistributions in binary form must reproduce the above copyright 42 * notice, this list of conditions and the following disclaimer in the 43 * documentation and/or other materials provided with the distribution. 44 * 3. All advertising materials mentioning features or use of this software 45 * must display the following acknowledgement: 46 * This product includes software developed by the Computer Systems 47 * Engineering Group at Lawrence Berkeley Laboratory. 48 * 4. Neither the name of the University nor of the Laboratory may be used 49 * to endorse or promote products derived from this software without 50 * specific prior written permission. 51 * 52 * THIS SOFTWARE IS PROVIDED BY THE REGENTS AND CONTRIBUTORS ``AS IS'' AND 53 * ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE 54 * IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE 55 * ARE DISCLAIMED. IN NO EVENT SHALL THE REGENTS OR CONTRIBUTORS BE LIABLE 56 * FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL 57 * DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS 58 * OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) 59 * HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT 60 * LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY 61 * OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF 62 * SUCH DAMAGE. 63 */ 64 65 /* 66 * Locking: there are three locks per device. 67 * 68 * - sc_lock, provided by the underlying driver. This is an adaptive lock, 69 * returned in the second parameter to hw_if->get_locks(). It is known 70 * as the "thread lock". 71 * 72 * It serializes access to state in all places except the 73 * driver's interrupt service routine. This lock is taken from process 74 * context (example: access to /dev/audio). It is also taken from soft 75 * interrupt handlers in this module, primarily to serialize delivery of 76 * wakeups. This lock may be used/provided by modules external to the 77 * audio subsystem, so take care not to introduce a lock order problem. 78 * LONG TERM SLEEPS MUST NOT OCCUR WITH THIS LOCK HELD. 79 * 80 * - sc_intr_lock, provided by the underlying driver. This may be either a 81 * spinlock (at IPL_SCHED or IPL_VM) or an adaptive lock (IPL_NONE or 82 * IPL_SOFT*), returned in the first parameter to hw_if->get_locks(). It 83 * is known as the "interrupt lock". 84 * 85 * It provides atomic access to the device's hardware state, and to audio 86 * channel data that may be accessed by the hardware driver's ISR. 87 * In all places outside the ISR, sc_lock must be held before taking 88 * sc_intr_lock. This is to ensure that groups of hardware operations are 89 * made atomically. SLEEPS CANNOT OCCUR WITH THIS LOCK HELD. 90 * 91 * - sc_exlock, private to this module. This is a variable protected by 92 * sc_lock. It is known as the "critical section". 93 * Some operations release sc_lock in order to allocate memory, to wait 94 * for in-flight I/O to complete, to copy to/from user context, etc. 95 * sc_exlock provides a critical section even under the circumstance. 96 * "+" in following list indicates the interfaces which necessary to be 97 * protected by sc_exlock. 98 * 99 * List of hardware interface methods, and which locks are held when each 100 * is called by this module: 101 * 102 * METHOD INTR THREAD NOTES 103 * ----------------------- ------- ------- ------------------------- 104 * open x x + 105 * close x x + 106 * query_format - x 107 * set_format - x 108 * round_blocksize - x 109 * commit_settings - x 110 * init_output x x 111 * init_input x x 112 * start_output x x + 113 * start_input x x + 114 * halt_output x x + 115 * halt_input x x + 116 * speaker_ctl x x 117 * getdev - x 118 * set_port - x + 119 * get_port - x + 120 * query_devinfo - x 121 * allocm - - + (*1) 122 * freem - - + (*1) 123 * round_buffersize - x 124 * get_props - x Called at attach time 125 * trigger_output x x + 126 * trigger_input x x + 127 * dev_ioctl - x 128 * get_locks - - Called at attach time 129 * 130 * *1 Note: Before 8.0, since these have been called only at attach time, 131 * neither lock were necessary. Currently, on the other hand, since 132 * these may be also called after attach, the thread lock is required. 133 * 134 * In addition, there is an additional lock. 135 * 136 * - track->lock. This is an atomic variable and is similar to the 137 * "interrupt lock". This is one for each track. If any thread context 138 * (and software interrupt context) and hardware interrupt context who 139 * want to access some variables on this track, they must acquire this 140 * lock before. It protects track's consistency between hardware 141 * interrupt context and others. 142 */ 143 144 #include <sys/cdefs.h> 145 __KERNEL_RCSID(0, "$NetBSD: audio.c,v 1.45 2020/02/22 06:28:10 isaki Exp $"); 146 147 #ifdef _KERNEL_OPT 148 #include "audio.h" 149 #include "midi.h" 150 #endif 151 152 #if NAUDIO > 0 153 154 #include <sys/types.h> 155 #include <sys/param.h> 156 #include <sys/atomic.h> 157 #include <sys/audioio.h> 158 #include <sys/conf.h> 159 #include <sys/cpu.h> 160 #include <sys/device.h> 161 #include <sys/fcntl.h> 162 #include <sys/file.h> 163 #include <sys/filedesc.h> 164 #include <sys/intr.h> 165 #include <sys/ioctl.h> 166 #include <sys/kauth.h> 167 #include <sys/kernel.h> 168 #include <sys/kmem.h> 169 #include <sys/malloc.h> 170 #include <sys/mman.h> 171 #include <sys/module.h> 172 #include <sys/poll.h> 173 #include <sys/proc.h> 174 #include <sys/queue.h> 175 #include <sys/select.h> 176 #include <sys/signalvar.h> 177 #include <sys/stat.h> 178 #include <sys/sysctl.h> 179 #include <sys/systm.h> 180 #include <sys/syslog.h> 181 #include <sys/vnode.h> 182 183 #include <dev/audio/audio_if.h> 184 #include <dev/audio/audiovar.h> 185 #include <dev/audio/audiodef.h> 186 #include <dev/audio/linear.h> 187 #include <dev/audio/mulaw.h> 188 189 #include <machine/endian.h> 190 191 #include <uvm/uvm.h> 192 193 #include "ioconf.h" 194 195 /* 196 * 0: No debug logs 197 * 1: action changes like open/close/set_format... 198 * 2: + normal operations like read/write/ioctl... 199 * 3: + TRACEs except interrupt 200 * 4: + TRACEs including interrupt 201 */ 202 //#define AUDIO_DEBUG 1 203 204 #if defined(AUDIO_DEBUG) 205 206 int audiodebug = AUDIO_DEBUG; 207 static void audio_vtrace(struct audio_softc *sc, const char *, const char *, 208 const char *, va_list); 209 static void audio_trace(struct audio_softc *sc, const char *, const char *, ...) 210 __printflike(3, 4); 211 static void audio_tracet(const char *, audio_track_t *, const char *, ...) 212 __printflike(3, 4); 213 static void audio_tracef(const char *, audio_file_t *, const char *, ...) 214 __printflike(3, 4); 215 216 /* XXX sloppy memory logger */ 217 static void audio_mlog_init(void); 218 static void audio_mlog_free(void); 219 static void audio_mlog_softintr(void *); 220 extern void audio_mlog_flush(void); 221 extern void audio_mlog_printf(const char *, ...); 222 223 static int mlog_refs; /* reference counter */ 224 static char *mlog_buf[2]; /* double buffer */ 225 static int mlog_buflen; /* buffer length */ 226 static int mlog_used; /* used length */ 227 static int mlog_full; /* number of dropped lines by buffer full */ 228 static int mlog_drop; /* number of dropped lines by busy */ 229 static volatile uint32_t mlog_inuse; /* in-use */ 230 static int mlog_wpage; /* active page */ 231 static void *mlog_sih; /* softint handle */ 232 233 static void 234 audio_mlog_init(void) 235 { 236 mlog_refs++; 237 if (mlog_refs > 1) 238 return; 239 mlog_buflen = 4096; 240 mlog_buf[0] = kmem_zalloc(mlog_buflen, KM_SLEEP); 241 mlog_buf[1] = kmem_zalloc(mlog_buflen, KM_SLEEP); 242 mlog_used = 0; 243 mlog_full = 0; 244 mlog_drop = 0; 245 mlog_inuse = 0; 246 mlog_wpage = 0; 247 mlog_sih = softint_establish(SOFTINT_SERIAL, audio_mlog_softintr, NULL); 248 if (mlog_sih == NULL) 249 printf("%s: softint_establish failed\n", __func__); 250 } 251 252 static void 253 audio_mlog_free(void) 254 { 255 mlog_refs--; 256 if (mlog_refs > 0) 257 return; 258 259 audio_mlog_flush(); 260 if (mlog_sih) 261 softint_disestablish(mlog_sih); 262 kmem_free(mlog_buf[0], mlog_buflen); 263 kmem_free(mlog_buf[1], mlog_buflen); 264 } 265 266 /* 267 * Flush memory buffer. 268 * It must not be called from hardware interrupt context. 269 */ 270 void 271 audio_mlog_flush(void) 272 { 273 if (mlog_refs == 0) 274 return; 275 276 /* Nothing to do if already in use ? */ 277 if (atomic_swap_32(&mlog_inuse, 1) == 1) 278 return; 279 280 int rpage = mlog_wpage; 281 mlog_wpage ^= 1; 282 mlog_buf[mlog_wpage][0] = '\0'; 283 mlog_used = 0; 284 285 atomic_swap_32(&mlog_inuse, 0); 286 287 if (mlog_buf[rpage][0] != '\0') { 288 printf("%s", mlog_buf[rpage]); 289 if (mlog_drop > 0) 290 printf("mlog_drop %d\n", mlog_drop); 291 if (mlog_full > 0) 292 printf("mlog_full %d\n", mlog_full); 293 } 294 mlog_full = 0; 295 mlog_drop = 0; 296 } 297 298 static void 299 audio_mlog_softintr(void *cookie) 300 { 301 audio_mlog_flush(); 302 } 303 304 void 305 audio_mlog_printf(const char *fmt, ...) 306 { 307 int len; 308 va_list ap; 309 310 if (atomic_swap_32(&mlog_inuse, 1) == 1) { 311 /* already inuse */ 312 mlog_drop++; 313 return; 314 } 315 316 va_start(ap, fmt); 317 len = vsnprintf( 318 mlog_buf[mlog_wpage] + mlog_used, 319 mlog_buflen - mlog_used, 320 fmt, ap); 321 va_end(ap); 322 323 mlog_used += len; 324 if (mlog_buflen - mlog_used <= 1) { 325 mlog_full++; 326 } 327 328 atomic_swap_32(&mlog_inuse, 0); 329 330 if (mlog_sih) 331 softint_schedule(mlog_sih); 332 } 333 334 /* trace functions */ 335 static void 336 audio_vtrace(struct audio_softc *sc, const char *funcname, const char *header, 337 const char *fmt, va_list ap) 338 { 339 char buf[256]; 340 int n; 341 342 n = 0; 343 buf[0] = '\0'; 344 n += snprintf(buf + n, sizeof(buf) - n, "%s@%d %s", 345 funcname, device_unit(sc->sc_dev), header); 346 n += vsnprintf(buf + n, sizeof(buf) - n, fmt, ap); 347 348 if (cpu_intr_p()) { 349 audio_mlog_printf("%s\n", buf); 350 } else { 351 audio_mlog_flush(); 352 printf("%s\n", buf); 353 } 354 } 355 356 static void 357 audio_trace(struct audio_softc *sc, const char *funcname, const char *fmt, ...) 358 { 359 va_list ap; 360 361 va_start(ap, fmt); 362 audio_vtrace(sc, funcname, "", fmt, ap); 363 va_end(ap); 364 } 365 366 static void 367 audio_tracet(const char *funcname, audio_track_t *track, const char *fmt, ...) 368 { 369 char hdr[16]; 370 va_list ap; 371 372 snprintf(hdr, sizeof(hdr), "#%d ", track->id); 373 va_start(ap, fmt); 374 audio_vtrace(track->mixer->sc, funcname, hdr, fmt, ap); 375 va_end(ap); 376 } 377 378 static void 379 audio_tracef(const char *funcname, audio_file_t *file, const char *fmt, ...) 380 { 381 char hdr[32]; 382 char phdr[16], rhdr[16]; 383 va_list ap; 384 385 phdr[0] = '\0'; 386 rhdr[0] = '\0'; 387 if (file->ptrack) 388 snprintf(phdr, sizeof(phdr), "#%d", file->ptrack->id); 389 if (file->rtrack) 390 snprintf(rhdr, sizeof(rhdr), "#%d", file->rtrack->id); 391 snprintf(hdr, sizeof(hdr), "{%s,%s} ", phdr, rhdr); 392 393 va_start(ap, fmt); 394 audio_vtrace(file->sc, funcname, hdr, fmt, ap); 395 va_end(ap); 396 } 397 398 #define DPRINTF(n, fmt...) do { \ 399 if (audiodebug >= (n)) { \ 400 audio_mlog_flush(); \ 401 printf(fmt); \ 402 } \ 403 } while (0) 404 #define TRACE(n, fmt...) do { \ 405 if (audiodebug >= (n)) audio_trace(sc, __func__, fmt); \ 406 } while (0) 407 #define TRACET(n, t, fmt...) do { \ 408 if (audiodebug >= (n)) audio_tracet(__func__, t, fmt); \ 409 } while (0) 410 #define TRACEF(n, f, fmt...) do { \ 411 if (audiodebug >= (n)) audio_tracef(__func__, f, fmt); \ 412 } while (0) 413 414 struct audio_track_debugbuf { 415 char usrbuf[32]; 416 char codec[32]; 417 char chvol[32]; 418 char chmix[32]; 419 char freq[32]; 420 char outbuf[32]; 421 }; 422 423 static void 424 audio_track_bufstat(audio_track_t *track, struct audio_track_debugbuf *buf) 425 { 426 427 memset(buf, 0, sizeof(*buf)); 428 429 snprintf(buf->outbuf, sizeof(buf->outbuf), " out=%d/%d/%d", 430 track->outbuf.head, track->outbuf.used, track->outbuf.capacity); 431 if (track->freq.filter) 432 snprintf(buf->freq, sizeof(buf->freq), " f=%d/%d/%d", 433 track->freq.srcbuf.head, 434 track->freq.srcbuf.used, 435 track->freq.srcbuf.capacity); 436 if (track->chmix.filter) 437 snprintf(buf->chmix, sizeof(buf->chmix), " m=%d", 438 track->chmix.srcbuf.used); 439 if (track->chvol.filter) 440 snprintf(buf->chvol, sizeof(buf->chvol), " v=%d", 441 track->chvol.srcbuf.used); 442 if (track->codec.filter) 443 snprintf(buf->codec, sizeof(buf->codec), " e=%d", 444 track->codec.srcbuf.used); 445 snprintf(buf->usrbuf, sizeof(buf->usrbuf), " usr=%d/%d/H%d", 446 track->usrbuf.head, track->usrbuf.used, track->usrbuf_usedhigh); 447 } 448 #else 449 #define DPRINTF(n, fmt...) do { } while (0) 450 #define TRACE(n, fmt, ...) do { } while (0) 451 #define TRACET(n, t, fmt, ...) do { } while (0) 452 #define TRACEF(n, f, fmt, ...) do { } while (0) 453 #endif 454 455 #define SPECIFIED(x) ((x) != ~0) 456 #define SPECIFIED_CH(x) ((x) != (u_char)~0) 457 458 /* Device timeout in msec */ 459 #define AUDIO_TIMEOUT (3000) 460 461 /* #define AUDIO_PM_IDLE */ 462 #ifdef AUDIO_PM_IDLE 463 int audio_idle_timeout = 30; 464 #endif 465 466 /* Number of elements of async mixer's pid */ 467 #define AM_CAPACITY (4) 468 469 struct portname { 470 const char *name; 471 int mask; 472 }; 473 474 static int audiomatch(device_t, cfdata_t, void *); 475 static void audioattach(device_t, device_t, void *); 476 static int audiodetach(device_t, int); 477 static int audioactivate(device_t, enum devact); 478 static void audiochilddet(device_t, device_t); 479 static int audiorescan(device_t, const char *, const int *); 480 481 static int audio_modcmd(modcmd_t, void *); 482 483 #ifdef AUDIO_PM_IDLE 484 static void audio_idle(void *); 485 static void audio_activity(device_t, devactive_t); 486 #endif 487 488 static bool audio_suspend(device_t dv, const pmf_qual_t *); 489 static bool audio_resume(device_t dv, const pmf_qual_t *); 490 static void audio_volume_down(device_t); 491 static void audio_volume_up(device_t); 492 static void audio_volume_toggle(device_t); 493 494 static void audio_mixer_capture(struct audio_softc *); 495 static void audio_mixer_restore(struct audio_softc *); 496 497 static void audio_softintr_rd(void *); 498 static void audio_softintr_wr(void *); 499 500 static int audio_enter_exclusive(struct audio_softc *); 501 static void audio_exit_exclusive(struct audio_softc *); 502 static int audio_track_waitio(struct audio_softc *, audio_track_t *); 503 504 static int audioclose(struct file *); 505 static int audioread(struct file *, off_t *, struct uio *, kauth_cred_t, int); 506 static int audiowrite(struct file *, off_t *, struct uio *, kauth_cred_t, int); 507 static int audioioctl(struct file *, u_long, void *); 508 static int audiopoll(struct file *, int); 509 static int audiokqfilter(struct file *, struct knote *); 510 static int audiommap(struct file *, off_t *, size_t, int, int *, int *, 511 struct uvm_object **, int *); 512 static int audiostat(struct file *, struct stat *); 513 514 static void filt_audiowrite_detach(struct knote *); 515 static int filt_audiowrite_event(struct knote *, long); 516 static void filt_audioread_detach(struct knote *); 517 static int filt_audioread_event(struct knote *, long); 518 519 static int audio_open(dev_t, struct audio_softc *, int, int, struct lwp *, 520 audio_file_t **); 521 static int audio_close(struct audio_softc *, audio_file_t *); 522 static int audio_read(struct audio_softc *, struct uio *, int, audio_file_t *); 523 static int audio_write(struct audio_softc *, struct uio *, int, audio_file_t *); 524 static void audio_file_clear(struct audio_softc *, audio_file_t *); 525 static int audio_ioctl(dev_t, struct audio_softc *, u_long, void *, int, 526 struct lwp *, audio_file_t *); 527 static int audio_poll(struct audio_softc *, int, struct lwp *, audio_file_t *); 528 static int audio_kqfilter(struct audio_softc *, audio_file_t *, struct knote *); 529 static int audio_mmap(struct audio_softc *, off_t *, size_t, int, int *, int *, 530 struct uvm_object **, int *, audio_file_t *); 531 532 static int audioctl_open(dev_t, struct audio_softc *, int, int, struct lwp *); 533 static int audioctl_close(struct audio_softc *, audio_file_t *); 534 535 static void audio_pintr(void *); 536 static void audio_rintr(void *); 537 538 static int audio_query_devinfo(struct audio_softc *, mixer_devinfo_t *); 539 540 static __inline int audio_track_readablebytes(const audio_track_t *); 541 static int audio_file_setinfo(struct audio_softc *, audio_file_t *, 542 const struct audio_info *); 543 static int audio_track_setinfo_check(audio_format2_t *, 544 const struct audio_prinfo *, const audio_format2_t *); 545 static void audio_track_setinfo_water(audio_track_t *, 546 const struct audio_info *); 547 static int audio_hw_setinfo(struct audio_softc *, const struct audio_info *, 548 struct audio_info *); 549 static int audio_hw_set_format(struct audio_softc *, int, 550 const audio_format2_t *, const audio_format2_t *, 551 audio_filter_reg_t *, audio_filter_reg_t *); 552 static int audiogetinfo(struct audio_softc *, struct audio_info *, int, 553 audio_file_t *); 554 static bool audio_can_playback(struct audio_softc *); 555 static bool audio_can_capture(struct audio_softc *); 556 static int audio_check_params(audio_format2_t *); 557 static int audio_mixers_init(struct audio_softc *sc, int, 558 const audio_format2_t *, const audio_format2_t *, 559 const audio_filter_reg_t *, const audio_filter_reg_t *); 560 static int audio_select_freq(const struct audio_format *); 561 static int audio_hw_probe(struct audio_softc *, int, int *, 562 audio_format2_t *, audio_format2_t *); 563 static int audio_hw_probe_fmt(struct audio_softc *, audio_format2_t *, int); 564 static int audio_hw_validate_format(struct audio_softc *, int, 565 const audio_format2_t *); 566 static int audio_mixers_set_format(struct audio_softc *, 567 const struct audio_info *); 568 static void audio_mixers_get_format(struct audio_softc *, struct audio_info *); 569 static int audio_sysctl_blk_ms(SYSCTLFN_PROTO); 570 static int audio_sysctl_multiuser(SYSCTLFN_PROTO); 571 #if defined(AUDIO_DEBUG) 572 static int audio_sysctl_debug(SYSCTLFN_PROTO); 573 static void audio_format2_tostr(char *, size_t, const audio_format2_t *); 574 static void audio_print_format2(const char *, const audio_format2_t *) __unused; 575 #endif 576 577 static void *audio_realloc(void *, size_t); 578 static int audio_realloc_usrbuf(audio_track_t *, int); 579 static void audio_free_usrbuf(audio_track_t *); 580 581 static audio_track_t *audio_track_create(struct audio_softc *, 582 audio_trackmixer_t *); 583 static void audio_track_destroy(audio_track_t *); 584 static audio_filter_t audio_track_get_codec(audio_track_t *, 585 const audio_format2_t *, const audio_format2_t *); 586 static int audio_track_set_format(audio_track_t *, audio_format2_t *); 587 static void audio_track_play(audio_track_t *); 588 static int audio_track_drain(struct audio_softc *, audio_track_t *); 589 static void audio_track_record(audio_track_t *); 590 static void audio_track_clear(struct audio_softc *, audio_track_t *); 591 592 static int audio_mixer_init(struct audio_softc *, int, 593 const audio_format2_t *, const audio_filter_reg_t *); 594 static void audio_mixer_destroy(struct audio_softc *, audio_trackmixer_t *); 595 static void audio_pmixer_start(struct audio_softc *, bool); 596 static void audio_pmixer_process(struct audio_softc *); 597 static void audio_pmixer_agc(audio_trackmixer_t *, int); 598 static int audio_pmixer_mix_track(audio_trackmixer_t *, audio_track_t *, int); 599 static void audio_pmixer_output(struct audio_softc *); 600 static int audio_pmixer_halt(struct audio_softc *); 601 static void audio_rmixer_start(struct audio_softc *); 602 static void audio_rmixer_process(struct audio_softc *); 603 static void audio_rmixer_input(struct audio_softc *); 604 static int audio_rmixer_halt(struct audio_softc *); 605 606 static void mixer_init(struct audio_softc *); 607 static int mixer_open(dev_t, struct audio_softc *, int, int, struct lwp *); 608 static int mixer_close(struct audio_softc *, audio_file_t *); 609 static int mixer_ioctl(struct audio_softc *, u_long, void *, int, struct lwp *); 610 static void mixer_async_add(struct audio_softc *, pid_t); 611 static void mixer_async_remove(struct audio_softc *, pid_t); 612 static void mixer_signal(struct audio_softc *); 613 614 static int au_portof(struct audio_softc *, char *, int); 615 616 static void au_setup_ports(struct audio_softc *, struct au_mixer_ports *, 617 mixer_devinfo_t *, const struct portname *); 618 static int au_set_lr_value(struct audio_softc *, mixer_ctrl_t *, int, int); 619 static int au_get_lr_value(struct audio_softc *, mixer_ctrl_t *, int *, int *); 620 static int au_set_gain(struct audio_softc *, struct au_mixer_ports *, int, int); 621 static void au_get_gain(struct audio_softc *, struct au_mixer_ports *, 622 u_int *, u_char *); 623 static int au_set_port(struct audio_softc *, struct au_mixer_ports *, u_int); 624 static int au_get_port(struct audio_softc *, struct au_mixer_ports *); 625 static int au_set_monitor_gain(struct audio_softc *, int); 626 static int au_get_monitor_gain(struct audio_softc *); 627 static int audio_get_port(struct audio_softc *, mixer_ctrl_t *); 628 static int audio_set_port(struct audio_softc *, mixer_ctrl_t *); 629 630 static __inline struct audio_params 631 format2_to_params(const audio_format2_t *f2) 632 { 633 audio_params_t p; 634 635 /* validbits/precision <-> precision/stride */ 636 p.sample_rate = f2->sample_rate; 637 p.channels = f2->channels; 638 p.encoding = f2->encoding; 639 p.validbits = f2->precision; 640 p.precision = f2->stride; 641 return p; 642 } 643 644 static __inline audio_format2_t 645 params_to_format2(const struct audio_params *p) 646 { 647 audio_format2_t f2; 648 649 /* precision/stride <-> validbits/precision */ 650 f2.sample_rate = p->sample_rate; 651 f2.channels = p->channels; 652 f2.encoding = p->encoding; 653 f2.precision = p->validbits; 654 f2.stride = p->precision; 655 return f2; 656 } 657 658 /* Return true if this track is a playback track. */ 659 static __inline bool 660 audio_track_is_playback(const audio_track_t *track) 661 { 662 663 return ((track->mode & AUMODE_PLAY) != 0); 664 } 665 666 /* Return true if this track is a recording track. */ 667 static __inline bool 668 audio_track_is_record(const audio_track_t *track) 669 { 670 671 return ((track->mode & AUMODE_RECORD) != 0); 672 } 673 674 #if 0 /* XXX Not used yet */ 675 /* 676 * Convert 0..255 volume used in userland to internal presentation 0..256. 677 */ 678 static __inline u_int 679 audio_volume_to_inner(u_int v) 680 { 681 682 return v < 127 ? v : v + 1; 683 } 684 685 /* 686 * Convert 0..256 internal presentation to 0..255 volume used in userland. 687 */ 688 static __inline u_int 689 audio_volume_to_outer(u_int v) 690 { 691 692 return v < 127 ? v : v - 1; 693 } 694 #endif /* 0 */ 695 696 static dev_type_open(audioopen); 697 /* XXXMRG use more dev_type_xxx */ 698 699 const struct cdevsw audio_cdevsw = { 700 .d_open = audioopen, 701 .d_close = noclose, 702 .d_read = noread, 703 .d_write = nowrite, 704 .d_ioctl = noioctl, 705 .d_stop = nostop, 706 .d_tty = notty, 707 .d_poll = nopoll, 708 .d_mmap = nommap, 709 .d_kqfilter = nokqfilter, 710 .d_discard = nodiscard, 711 .d_flag = D_OTHER | D_MPSAFE 712 }; 713 714 const struct fileops audio_fileops = { 715 .fo_name = "audio", 716 .fo_read = audioread, 717 .fo_write = audiowrite, 718 .fo_ioctl = audioioctl, 719 .fo_fcntl = fnullop_fcntl, 720 .fo_stat = audiostat, 721 .fo_poll = audiopoll, 722 .fo_close = audioclose, 723 .fo_mmap = audiommap, 724 .fo_kqfilter = audiokqfilter, 725 .fo_restart = fnullop_restart 726 }; 727 728 /* The default audio mode: 8 kHz mono mu-law */ 729 static const struct audio_params audio_default = { 730 .sample_rate = 8000, 731 .encoding = AUDIO_ENCODING_ULAW, 732 .precision = 8, 733 .validbits = 8, 734 .channels = 1, 735 }; 736 737 static const char *encoding_names[] = { 738 "none", 739 AudioEmulaw, 740 AudioEalaw, 741 "pcm16", 742 "pcm8", 743 AudioEadpcm, 744 AudioEslinear_le, 745 AudioEslinear_be, 746 AudioEulinear_le, 747 AudioEulinear_be, 748 AudioEslinear, 749 AudioEulinear, 750 AudioEmpeg_l1_stream, 751 AudioEmpeg_l1_packets, 752 AudioEmpeg_l1_system, 753 AudioEmpeg_l2_stream, 754 AudioEmpeg_l2_packets, 755 AudioEmpeg_l2_system, 756 AudioEac3, 757 }; 758 759 /* 760 * Returns encoding name corresponding to AUDIO_ENCODING_*. 761 * Note that it may return a local buffer because it is mainly for debugging. 762 */ 763 const char * 764 audio_encoding_name(int encoding) 765 { 766 static char buf[16]; 767 768 if (0 <= encoding && encoding < __arraycount(encoding_names)) { 769 return encoding_names[encoding]; 770 } else { 771 snprintf(buf, sizeof(buf), "enc=%d", encoding); 772 return buf; 773 } 774 } 775 776 /* 777 * Supported encodings used by AUDIO_GETENC. 778 * index and flags are set by code. 779 * XXX is there any needs for SLINEAR_OE:>=16/ULINEAR_OE:>=16 ? 780 */ 781 static const audio_encoding_t audio_encodings[] = { 782 { 0, AudioEmulaw, AUDIO_ENCODING_ULAW, 8, 0 }, 783 { 0, AudioEalaw, AUDIO_ENCODING_ALAW, 8, 0 }, 784 { 0, AudioEslinear, AUDIO_ENCODING_SLINEAR, 8, 0 }, 785 { 0, AudioEulinear, AUDIO_ENCODING_ULINEAR, 8, 0 }, 786 { 0, AudioEslinear_le, AUDIO_ENCODING_SLINEAR_LE, 16, 0 }, 787 { 0, AudioEulinear_le, AUDIO_ENCODING_ULINEAR_LE, 16, 0 }, 788 { 0, AudioEslinear_be, AUDIO_ENCODING_SLINEAR_BE, 16, 0 }, 789 { 0, AudioEulinear_be, AUDIO_ENCODING_ULINEAR_BE, 16, 0 }, 790 #if defined(AUDIO_SUPPORT_LINEAR24) 791 { 0, AudioEslinear_le, AUDIO_ENCODING_SLINEAR_LE, 24, 0 }, 792 { 0, AudioEulinear_le, AUDIO_ENCODING_ULINEAR_LE, 24, 0 }, 793 { 0, AudioEslinear_be, AUDIO_ENCODING_SLINEAR_BE, 24, 0 }, 794 { 0, AudioEulinear_be, AUDIO_ENCODING_ULINEAR_BE, 24, 0 }, 795 #endif 796 { 0, AudioEslinear_le, AUDIO_ENCODING_SLINEAR_LE, 32, 0 }, 797 { 0, AudioEulinear_le, AUDIO_ENCODING_ULINEAR_LE, 32, 0 }, 798 { 0, AudioEslinear_be, AUDIO_ENCODING_SLINEAR_BE, 32, 0 }, 799 { 0, AudioEulinear_be, AUDIO_ENCODING_ULINEAR_BE, 32, 0 }, 800 }; 801 802 static const struct portname itable[] = { 803 { AudioNmicrophone, AUDIO_MICROPHONE }, 804 { AudioNline, AUDIO_LINE_IN }, 805 { AudioNcd, AUDIO_CD }, 806 { 0, 0 } 807 }; 808 static const struct portname otable[] = { 809 { AudioNspeaker, AUDIO_SPEAKER }, 810 { AudioNheadphone, AUDIO_HEADPHONE }, 811 { AudioNline, AUDIO_LINE_OUT }, 812 { 0, 0 } 813 }; 814 815 CFATTACH_DECL3_NEW(audio, sizeof(struct audio_softc), 816 audiomatch, audioattach, audiodetach, audioactivate, audiorescan, 817 audiochilddet, DVF_DETACH_SHUTDOWN); 818 819 static int 820 audiomatch(device_t parent, cfdata_t match, void *aux) 821 { 822 struct audio_attach_args *sa; 823 824 sa = aux; 825 DPRINTF(1, "%s: type=%d sa=%p hw=%p\n", 826 __func__, sa->type, sa, sa->hwif); 827 return (sa->type == AUDIODEV_TYPE_AUDIO) ? 1 : 0; 828 } 829 830 static void 831 audioattach(device_t parent, device_t self, void *aux) 832 { 833 struct audio_softc *sc; 834 struct audio_attach_args *sa; 835 const struct audio_hw_if *hw_if; 836 audio_format2_t phwfmt; 837 audio_format2_t rhwfmt; 838 audio_filter_reg_t pfil; 839 audio_filter_reg_t rfil; 840 const struct sysctlnode *node; 841 void *hdlp; 842 bool has_playback; 843 bool has_capture; 844 bool has_indep; 845 bool has_fulldup; 846 int mode; 847 int error; 848 849 sc = device_private(self); 850 sc->sc_dev = self; 851 sa = (struct audio_attach_args *)aux; 852 hw_if = sa->hwif; 853 hdlp = sa->hdl; 854 855 if (hw_if == NULL || hw_if->get_locks == NULL) { 856 panic("audioattach: missing hw_if method"); 857 } 858 859 hw_if->get_locks(hdlp, &sc->sc_intr_lock, &sc->sc_lock); 860 861 #ifdef DIAGNOSTIC 862 if (hw_if->query_format == NULL || 863 hw_if->set_format == NULL || 864 (hw_if->start_output == NULL && hw_if->trigger_output == NULL) || 865 (hw_if->start_input == NULL && hw_if->trigger_input == NULL) || 866 hw_if->halt_output == NULL || 867 hw_if->halt_input == NULL || 868 hw_if->getdev == NULL || 869 hw_if->set_port == NULL || 870 hw_if->get_port == NULL || 871 hw_if->query_devinfo == NULL || 872 hw_if->get_props == NULL) { 873 aprint_error(": missing method\n"); 874 return; 875 } 876 #endif 877 878 sc->hw_if = hw_if; 879 sc->hw_hdl = hdlp; 880 sc->hw_dev = parent; 881 882 sc->sc_blk_ms = AUDIO_BLK_MS; 883 SLIST_INIT(&sc->sc_files); 884 cv_init(&sc->sc_exlockcv, "audiolk"); 885 sc->sc_am_capacity = 0; 886 sc->sc_am_used = 0; 887 sc->sc_am = NULL; 888 889 mutex_enter(sc->sc_lock); 890 sc->sc_props = hw_if->get_props(sc->hw_hdl); 891 mutex_exit(sc->sc_lock); 892 893 /* MMAP is now supported by upper layer. */ 894 sc->sc_props |= AUDIO_PROP_MMAP; 895 896 has_playback = (sc->sc_props & AUDIO_PROP_PLAYBACK); 897 has_capture = (sc->sc_props & AUDIO_PROP_CAPTURE); 898 has_indep = (sc->sc_props & AUDIO_PROP_INDEPENDENT); 899 has_fulldup = (sc->sc_props & AUDIO_PROP_FULLDUPLEX); 900 901 KASSERT(has_playback || has_capture); 902 /* Unidirectional device must have neither FULLDUP nor INDEPENDENT. */ 903 if (!has_playback || !has_capture) { 904 KASSERT(!has_indep); 905 KASSERT(!has_fulldup); 906 } 907 908 mode = 0; 909 if (has_playback) { 910 aprint_normal(": playback"); 911 mode |= AUMODE_PLAY; 912 } 913 if (has_capture) { 914 aprint_normal("%c capture", has_playback ? ',' : ':'); 915 mode |= AUMODE_RECORD; 916 } 917 if (has_playback && has_capture) { 918 if (has_fulldup) 919 aprint_normal(", full duplex"); 920 else 921 aprint_normal(", half duplex"); 922 923 if (has_indep) 924 aprint_normal(", independent"); 925 } 926 927 aprint_naive("\n"); 928 aprint_normal("\n"); 929 930 /* probe hw params */ 931 memset(&phwfmt, 0, sizeof(phwfmt)); 932 memset(&rhwfmt, 0, sizeof(rhwfmt)); 933 memset(&pfil, 0, sizeof(pfil)); 934 memset(&rfil, 0, sizeof(rfil)); 935 mutex_enter(sc->sc_lock); 936 error = audio_hw_probe(sc, has_indep, &mode, &phwfmt, &rhwfmt); 937 if (error) { 938 mutex_exit(sc->sc_lock); 939 aprint_error_dev(self, "audio_hw_probe failed, " 940 "error = %d\n", error); 941 goto bad; 942 } 943 if (mode == 0) { 944 mutex_exit(sc->sc_lock); 945 aprint_error_dev(self, "audio_hw_probe failed, no mode\n"); 946 goto bad; 947 } 948 /* Init hardware. */ 949 /* hw_probe() also validates [pr]hwfmt. */ 950 error = audio_hw_set_format(sc, mode, &phwfmt, &rhwfmt, &pfil, &rfil); 951 if (error) { 952 mutex_exit(sc->sc_lock); 953 aprint_error_dev(self, "audio_hw_set_format failed, " 954 "error = %d\n", error); 955 goto bad; 956 } 957 958 /* 959 * Init track mixers. If at least one direction is available on 960 * attach time, we assume a success. 961 */ 962 error = audio_mixers_init(sc, mode, &phwfmt, &rhwfmt, &pfil, &rfil); 963 mutex_exit(sc->sc_lock); 964 if (sc->sc_pmixer == NULL && sc->sc_rmixer == NULL) { 965 aprint_error_dev(self, "audio_mixers_init failed, " 966 "error = %d\n", error); 967 goto bad; 968 } 969 970 selinit(&sc->sc_wsel); 971 selinit(&sc->sc_rsel); 972 973 /* Initial parameter of /dev/sound */ 974 sc->sc_sound_pparams = params_to_format2(&audio_default); 975 sc->sc_sound_rparams = params_to_format2(&audio_default); 976 sc->sc_sound_ppause = false; 977 sc->sc_sound_rpause = false; 978 979 /* XXX TODO: consider about sc_ai */ 980 981 mixer_init(sc); 982 TRACE(2, "inputs ports=0x%x, input master=%d, " 983 "output ports=0x%x, output master=%d", 984 sc->sc_inports.allports, sc->sc_inports.master, 985 sc->sc_outports.allports, sc->sc_outports.master); 986 987 sysctl_createv(&sc->sc_log, 0, NULL, &node, 988 0, 989 CTLTYPE_NODE, device_xname(sc->sc_dev), 990 SYSCTL_DESCR("audio test"), 991 NULL, 0, 992 NULL, 0, 993 CTL_HW, 994 CTL_CREATE, CTL_EOL); 995 996 if (node != NULL) { 997 sysctl_createv(&sc->sc_log, 0, NULL, NULL, 998 CTLFLAG_READWRITE, 999 CTLTYPE_INT, "blk_ms", 1000 SYSCTL_DESCR("blocksize in msec"), 1001 audio_sysctl_blk_ms, 0, (void *)sc, 0, 1002 CTL_HW, node->sysctl_num, CTL_CREATE, CTL_EOL); 1003 1004 sysctl_createv(&sc->sc_log, 0, NULL, NULL, 1005 CTLFLAG_READWRITE, 1006 CTLTYPE_BOOL, "multiuser", 1007 SYSCTL_DESCR("allow multiple user access"), 1008 audio_sysctl_multiuser, 0, (void *)sc, 0, 1009 CTL_HW, node->sysctl_num, CTL_CREATE, CTL_EOL); 1010 1011 #if defined(AUDIO_DEBUG) 1012 sysctl_createv(&sc->sc_log, 0, NULL, NULL, 1013 CTLFLAG_READWRITE, 1014 CTLTYPE_INT, "debug", 1015 SYSCTL_DESCR("debug level (0..4)"), 1016 audio_sysctl_debug, 0, (void *)sc, 0, 1017 CTL_HW, node->sysctl_num, CTL_CREATE, CTL_EOL); 1018 #endif 1019 } 1020 1021 #ifdef AUDIO_PM_IDLE 1022 callout_init(&sc->sc_idle_counter, 0); 1023 callout_setfunc(&sc->sc_idle_counter, audio_idle, self); 1024 #endif 1025 1026 if (!pmf_device_register(self, audio_suspend, audio_resume)) 1027 aprint_error_dev(self, "couldn't establish power handler\n"); 1028 #ifdef AUDIO_PM_IDLE 1029 if (!device_active_register(self, audio_activity)) 1030 aprint_error_dev(self, "couldn't register activity handler\n"); 1031 #endif 1032 1033 if (!pmf_event_register(self, PMFE_AUDIO_VOLUME_DOWN, 1034 audio_volume_down, true)) 1035 aprint_error_dev(self, "couldn't add volume down handler\n"); 1036 if (!pmf_event_register(self, PMFE_AUDIO_VOLUME_UP, 1037 audio_volume_up, true)) 1038 aprint_error_dev(self, "couldn't add volume up handler\n"); 1039 if (!pmf_event_register(self, PMFE_AUDIO_VOLUME_TOGGLE, 1040 audio_volume_toggle, true)) 1041 aprint_error_dev(self, "couldn't add volume toggle handler\n"); 1042 1043 #ifdef AUDIO_PM_IDLE 1044 callout_schedule(&sc->sc_idle_counter, audio_idle_timeout * hz); 1045 #endif 1046 1047 #if defined(AUDIO_DEBUG) 1048 audio_mlog_init(); 1049 #endif 1050 1051 audiorescan(self, "audio", NULL); 1052 return; 1053 1054 bad: 1055 /* Clearing hw_if means that device is attached but disabled. */ 1056 sc->hw_if = NULL; 1057 aprint_error_dev(sc->sc_dev, "disabled\n"); 1058 return; 1059 } 1060 1061 /* 1062 * Initialize hardware mixer. 1063 * This function is called from audioattach(). 1064 */ 1065 static void 1066 mixer_init(struct audio_softc *sc) 1067 { 1068 mixer_devinfo_t mi; 1069 int iclass, mclass, oclass, rclass; 1070 int record_master_found, record_source_found; 1071 1072 iclass = mclass = oclass = rclass = -1; 1073 sc->sc_inports.index = -1; 1074 sc->sc_inports.master = -1; 1075 sc->sc_inports.nports = 0; 1076 sc->sc_inports.isenum = false; 1077 sc->sc_inports.allports = 0; 1078 sc->sc_inports.isdual = false; 1079 sc->sc_inports.mixerout = -1; 1080 sc->sc_inports.cur_port = -1; 1081 sc->sc_outports.index = -1; 1082 sc->sc_outports.master = -1; 1083 sc->sc_outports.nports = 0; 1084 sc->sc_outports.isenum = false; 1085 sc->sc_outports.allports = 0; 1086 sc->sc_outports.isdual = false; 1087 sc->sc_outports.mixerout = -1; 1088 sc->sc_outports.cur_port = -1; 1089 sc->sc_monitor_port = -1; 1090 /* 1091 * Read through the underlying driver's list, picking out the class 1092 * names from the mixer descriptions. We'll need them to decode the 1093 * mixer descriptions on the next pass through the loop. 1094 */ 1095 mutex_enter(sc->sc_lock); 1096 for(mi.index = 0; ; mi.index++) { 1097 if (audio_query_devinfo(sc, &mi) != 0) 1098 break; 1099 /* 1100 * The type of AUDIO_MIXER_CLASS merely introduces a class. 1101 * All the other types describe an actual mixer. 1102 */ 1103 if (mi.type == AUDIO_MIXER_CLASS) { 1104 if (strcmp(mi.label.name, AudioCinputs) == 0) 1105 iclass = mi.mixer_class; 1106 if (strcmp(mi.label.name, AudioCmonitor) == 0) 1107 mclass = mi.mixer_class; 1108 if (strcmp(mi.label.name, AudioCoutputs) == 0) 1109 oclass = mi.mixer_class; 1110 if (strcmp(mi.label.name, AudioCrecord) == 0) 1111 rclass = mi.mixer_class; 1112 } 1113 } 1114 mutex_exit(sc->sc_lock); 1115 1116 /* Allocate save area. Ensure non-zero allocation. */ 1117 sc->sc_nmixer_states = mi.index; 1118 sc->sc_mixer_state = kmem_zalloc(sizeof(mixer_ctrl_t) * 1119 (sc->sc_nmixer_states + 1), KM_SLEEP); 1120 1121 /* 1122 * This is where we assign each control in the "audio" model, to the 1123 * underlying "mixer" control. We walk through the whole list once, 1124 * assigning likely candidates as we come across them. 1125 */ 1126 record_master_found = 0; 1127 record_source_found = 0; 1128 mutex_enter(sc->sc_lock); 1129 for(mi.index = 0; ; mi.index++) { 1130 if (audio_query_devinfo(sc, &mi) != 0) 1131 break; 1132 KASSERT(mi.index < sc->sc_nmixer_states); 1133 if (mi.type == AUDIO_MIXER_CLASS) 1134 continue; 1135 if (mi.mixer_class == iclass) { 1136 /* 1137 * AudioCinputs is only a fallback, when we don't 1138 * find what we're looking for in AudioCrecord, so 1139 * check the flags before accepting one of these. 1140 */ 1141 if (strcmp(mi.label.name, AudioNmaster) == 0 1142 && record_master_found == 0) 1143 sc->sc_inports.master = mi.index; 1144 if (strcmp(mi.label.name, AudioNsource) == 0 1145 && record_source_found == 0) { 1146 if (mi.type == AUDIO_MIXER_ENUM) { 1147 int i; 1148 for(i = 0; i < mi.un.e.num_mem; i++) 1149 if (strcmp(mi.un.e.member[i].label.name, 1150 AudioNmixerout) == 0) 1151 sc->sc_inports.mixerout = 1152 mi.un.e.member[i].ord; 1153 } 1154 au_setup_ports(sc, &sc->sc_inports, &mi, 1155 itable); 1156 } 1157 if (strcmp(mi.label.name, AudioNdac) == 0 && 1158 sc->sc_outports.master == -1) 1159 sc->sc_outports.master = mi.index; 1160 } else if (mi.mixer_class == mclass) { 1161 if (strcmp(mi.label.name, AudioNmonitor) == 0) 1162 sc->sc_monitor_port = mi.index; 1163 } else if (mi.mixer_class == oclass) { 1164 if (strcmp(mi.label.name, AudioNmaster) == 0) 1165 sc->sc_outports.master = mi.index; 1166 if (strcmp(mi.label.name, AudioNselect) == 0) 1167 au_setup_ports(sc, &sc->sc_outports, &mi, 1168 otable); 1169 } else if (mi.mixer_class == rclass) { 1170 /* 1171 * These are the preferred mixers for the audio record 1172 * controls, so set the flags here, but don't check. 1173 */ 1174 if (strcmp(mi.label.name, AudioNmaster) == 0) { 1175 sc->sc_inports.master = mi.index; 1176 record_master_found = 1; 1177 } 1178 #if 1 /* Deprecated. Use AudioNmaster. */ 1179 if (strcmp(mi.label.name, AudioNrecord) == 0) { 1180 sc->sc_inports.master = mi.index; 1181 record_master_found = 1; 1182 } 1183 if (strcmp(mi.label.name, AudioNvolume) == 0) { 1184 sc->sc_inports.master = mi.index; 1185 record_master_found = 1; 1186 } 1187 #endif 1188 if (strcmp(mi.label.name, AudioNsource) == 0) { 1189 if (mi.type == AUDIO_MIXER_ENUM) { 1190 int i; 1191 for(i = 0; i < mi.un.e.num_mem; i++) 1192 if (strcmp(mi.un.e.member[i].label.name, 1193 AudioNmixerout) == 0) 1194 sc->sc_inports.mixerout = 1195 mi.un.e.member[i].ord; 1196 } 1197 au_setup_ports(sc, &sc->sc_inports, &mi, 1198 itable); 1199 record_source_found = 1; 1200 } 1201 } 1202 } 1203 mutex_exit(sc->sc_lock); 1204 } 1205 1206 static int 1207 audioactivate(device_t self, enum devact act) 1208 { 1209 struct audio_softc *sc = device_private(self); 1210 1211 switch (act) { 1212 case DVACT_DEACTIVATE: 1213 mutex_enter(sc->sc_lock); 1214 sc->sc_dying = true; 1215 cv_broadcast(&sc->sc_exlockcv); 1216 mutex_exit(sc->sc_lock); 1217 return 0; 1218 default: 1219 return EOPNOTSUPP; 1220 } 1221 } 1222 1223 static int 1224 audiodetach(device_t self, int flags) 1225 { 1226 struct audio_softc *sc; 1227 int maj, mn; 1228 int error; 1229 1230 sc = device_private(self); 1231 TRACE(2, "flags=%d", flags); 1232 1233 /* device is not initialized */ 1234 if (sc->hw_if == NULL) 1235 return 0; 1236 1237 /* Start draining existing accessors of the device. */ 1238 error = config_detach_children(self, flags); 1239 if (error) 1240 return error; 1241 1242 mutex_enter(sc->sc_lock); 1243 sc->sc_dying = true; 1244 cv_broadcast(&sc->sc_exlockcv); 1245 if (sc->sc_pmixer) 1246 cv_broadcast(&sc->sc_pmixer->outcv); 1247 if (sc->sc_rmixer) 1248 cv_broadcast(&sc->sc_rmixer->outcv); 1249 mutex_exit(sc->sc_lock); 1250 1251 /* delete sysctl nodes */ 1252 sysctl_teardown(&sc->sc_log); 1253 1254 /* locate the major number */ 1255 maj = cdevsw_lookup_major(&audio_cdevsw); 1256 1257 /* 1258 * Nuke the vnodes for any open instances (calls close). 1259 * Will wait until any activity on the device nodes has ceased. 1260 */ 1261 mn = device_unit(self); 1262 vdevgone(maj, mn | SOUND_DEVICE, mn | SOUND_DEVICE, VCHR); 1263 vdevgone(maj, mn | AUDIO_DEVICE, mn | AUDIO_DEVICE, VCHR); 1264 vdevgone(maj, mn | AUDIOCTL_DEVICE, mn | AUDIOCTL_DEVICE, VCHR); 1265 vdevgone(maj, mn | MIXER_DEVICE, mn | MIXER_DEVICE, VCHR); 1266 1267 pmf_event_deregister(self, PMFE_AUDIO_VOLUME_DOWN, 1268 audio_volume_down, true); 1269 pmf_event_deregister(self, PMFE_AUDIO_VOLUME_UP, 1270 audio_volume_up, true); 1271 pmf_event_deregister(self, PMFE_AUDIO_VOLUME_TOGGLE, 1272 audio_volume_toggle, true); 1273 1274 #ifdef AUDIO_PM_IDLE 1275 callout_halt(&sc->sc_idle_counter, sc->sc_lock); 1276 1277 device_active_deregister(self, audio_activity); 1278 #endif 1279 1280 pmf_device_deregister(self); 1281 1282 /* Free resources */ 1283 mutex_enter(sc->sc_lock); 1284 if (sc->sc_pmixer) { 1285 audio_mixer_destroy(sc, sc->sc_pmixer); 1286 kmem_free(sc->sc_pmixer, sizeof(*sc->sc_pmixer)); 1287 } 1288 if (sc->sc_rmixer) { 1289 audio_mixer_destroy(sc, sc->sc_rmixer); 1290 kmem_free(sc->sc_rmixer, sizeof(*sc->sc_rmixer)); 1291 } 1292 mutex_exit(sc->sc_lock); 1293 if (sc->sc_am) 1294 kern_free(sc->sc_am); 1295 1296 seldestroy(&sc->sc_wsel); 1297 seldestroy(&sc->sc_rsel); 1298 1299 #ifdef AUDIO_PM_IDLE 1300 callout_destroy(&sc->sc_idle_counter); 1301 #endif 1302 1303 cv_destroy(&sc->sc_exlockcv); 1304 1305 #if defined(AUDIO_DEBUG) 1306 audio_mlog_free(); 1307 #endif 1308 1309 return 0; 1310 } 1311 1312 static void 1313 audiochilddet(device_t self, device_t child) 1314 { 1315 1316 /* we hold no child references, so do nothing */ 1317 } 1318 1319 static int 1320 audiosearch(device_t parent, cfdata_t cf, const int *locs, void *aux) 1321 { 1322 1323 if (config_match(parent, cf, aux)) 1324 config_attach_loc(parent, cf, locs, aux, NULL); 1325 1326 return 0; 1327 } 1328 1329 static int 1330 audiorescan(device_t self, const char *ifattr, const int *flags) 1331 { 1332 struct audio_softc *sc = device_private(self); 1333 1334 if (!ifattr_match(ifattr, "audio")) 1335 return 0; 1336 1337 config_search_loc(audiosearch, sc->sc_dev, "audio", NULL, NULL); 1338 1339 return 0; 1340 } 1341 1342 /* 1343 * Called from hardware driver. This is where the MI audio driver gets 1344 * probed/attached to the hardware driver. 1345 */ 1346 device_t 1347 audio_attach_mi(const struct audio_hw_if *ahwp, void *hdlp, device_t dev) 1348 { 1349 struct audio_attach_args arg; 1350 1351 #ifdef DIAGNOSTIC 1352 if (ahwp == NULL) { 1353 aprint_error("audio_attach_mi: NULL\n"); 1354 return 0; 1355 } 1356 #endif 1357 arg.type = AUDIODEV_TYPE_AUDIO; 1358 arg.hwif = ahwp; 1359 arg.hdl = hdlp; 1360 return config_found(dev, &arg, audioprint); 1361 } 1362 1363 /* 1364 * Acquire sc_lock and enter exlock critical section. 1365 * If successful, it returns 0. Otherwise returns errno. 1366 * Must be called without sc_lock held. 1367 */ 1368 static int 1369 audio_enter_exclusive(struct audio_softc *sc) 1370 { 1371 int error; 1372 1373 mutex_enter(sc->sc_lock); 1374 if (sc->sc_dying) { 1375 mutex_exit(sc->sc_lock); 1376 return EIO; 1377 } 1378 1379 while (__predict_false(sc->sc_exlock != 0)) { 1380 error = cv_wait_sig(&sc->sc_exlockcv, sc->sc_lock); 1381 if (sc->sc_dying) 1382 error = EIO; 1383 if (error) { 1384 mutex_exit(sc->sc_lock); 1385 return error; 1386 } 1387 } 1388 1389 /* Acquire */ 1390 sc->sc_exlock = 1; 1391 return 0; 1392 } 1393 1394 /* 1395 * Leave exlock critical section and release sc_lock. 1396 * Must be called with sc_lock held. 1397 */ 1398 static void 1399 audio_exit_exclusive(struct audio_softc *sc) 1400 { 1401 1402 KASSERT(mutex_owned(sc->sc_lock)); 1403 KASSERT(sc->sc_exlock); 1404 1405 /* Leave critical section */ 1406 sc->sc_exlock = 0; 1407 cv_broadcast(&sc->sc_exlockcv); 1408 mutex_exit(sc->sc_lock); 1409 } 1410 1411 /* 1412 * Wait for I/O to complete, releasing sc_lock. 1413 * Must be called with sc_lock held. 1414 */ 1415 static int 1416 audio_track_waitio(struct audio_softc *sc, audio_track_t *track) 1417 { 1418 int error; 1419 1420 KASSERT(track); 1421 KASSERT(mutex_owned(sc->sc_lock)); 1422 1423 /* Wait for pending I/O to complete. */ 1424 error = cv_timedwait_sig(&track->mixer->outcv, sc->sc_lock, 1425 mstohz(AUDIO_TIMEOUT)); 1426 if (sc->sc_dying) { 1427 error = EIO; 1428 } 1429 if (error) { 1430 TRACET(2, track, "cv_timedwait_sig failed %d", error); 1431 if (error == EWOULDBLOCK) 1432 device_printf(sc->sc_dev, "device timeout\n"); 1433 } else { 1434 TRACET(3, track, "wakeup"); 1435 } 1436 return error; 1437 } 1438 1439 /* 1440 * Try to acquire track lock. 1441 * It doesn't block if the track lock is already aquired. 1442 * Returns true if the track lock was acquired, or false if the track 1443 * lock was already acquired. 1444 */ 1445 static __inline bool 1446 audio_track_lock_tryenter(audio_track_t *track) 1447 { 1448 return (atomic_cas_uint(&track->lock, 0, 1) == 0); 1449 } 1450 1451 /* 1452 * Acquire track lock. 1453 */ 1454 static __inline void 1455 audio_track_lock_enter(audio_track_t *track) 1456 { 1457 /* Don't sleep here. */ 1458 while (audio_track_lock_tryenter(track) == false) 1459 ; 1460 } 1461 1462 /* 1463 * Release track lock. 1464 */ 1465 static __inline void 1466 audio_track_lock_exit(audio_track_t *track) 1467 { 1468 atomic_swap_uint(&track->lock, 0); 1469 } 1470 1471 1472 static int 1473 audioopen(dev_t dev, int flags, int ifmt, struct lwp *l) 1474 { 1475 struct audio_softc *sc; 1476 int error; 1477 1478 /* Find the device */ 1479 sc = device_lookup_private(&audio_cd, AUDIOUNIT(dev)); 1480 if (sc == NULL || sc->hw_if == NULL) 1481 return ENXIO; 1482 1483 error = audio_enter_exclusive(sc); 1484 if (error) 1485 return error; 1486 1487 device_active(sc->sc_dev, DVA_SYSTEM); 1488 switch (AUDIODEV(dev)) { 1489 case SOUND_DEVICE: 1490 case AUDIO_DEVICE: 1491 error = audio_open(dev, sc, flags, ifmt, l, NULL); 1492 break; 1493 case AUDIOCTL_DEVICE: 1494 error = audioctl_open(dev, sc, flags, ifmt, l); 1495 break; 1496 case MIXER_DEVICE: 1497 error = mixer_open(dev, sc, flags, ifmt, l); 1498 break; 1499 default: 1500 error = ENXIO; 1501 break; 1502 } 1503 audio_exit_exclusive(sc); 1504 1505 return error; 1506 } 1507 1508 static int 1509 audioclose(struct file *fp) 1510 { 1511 struct audio_softc *sc; 1512 audio_file_t *file; 1513 int error; 1514 dev_t dev; 1515 1516 KASSERT(fp->f_audioctx); 1517 file = fp->f_audioctx; 1518 sc = file->sc; 1519 dev = file->dev; 1520 1521 /* audio_{enter,exit}_exclusive() is called by lower audio_close() */ 1522 1523 device_active(sc->sc_dev, DVA_SYSTEM); 1524 switch (AUDIODEV(dev)) { 1525 case SOUND_DEVICE: 1526 case AUDIO_DEVICE: 1527 error = audio_close(sc, file); 1528 break; 1529 case AUDIOCTL_DEVICE: 1530 error = audioctl_close(sc, file); 1531 break; 1532 case MIXER_DEVICE: 1533 error = mixer_close(sc, file); 1534 break; 1535 default: 1536 error = ENXIO; 1537 break; 1538 } 1539 /* f_audioctx has already been freed in lower *_close() */ 1540 fp->f_audioctx = NULL; 1541 1542 return error; 1543 } 1544 1545 static int 1546 audioread(struct file *fp, off_t *offp, struct uio *uio, kauth_cred_t cred, 1547 int ioflag) 1548 { 1549 struct audio_softc *sc; 1550 audio_file_t *file; 1551 int error; 1552 dev_t dev; 1553 1554 KASSERT(fp->f_audioctx); 1555 file = fp->f_audioctx; 1556 sc = file->sc; 1557 dev = file->dev; 1558 1559 if (fp->f_flag & O_NONBLOCK) 1560 ioflag |= IO_NDELAY; 1561 1562 switch (AUDIODEV(dev)) { 1563 case SOUND_DEVICE: 1564 case AUDIO_DEVICE: 1565 error = audio_read(sc, uio, ioflag, file); 1566 break; 1567 case AUDIOCTL_DEVICE: 1568 case MIXER_DEVICE: 1569 error = ENODEV; 1570 break; 1571 default: 1572 error = ENXIO; 1573 break; 1574 } 1575 1576 return error; 1577 } 1578 1579 static int 1580 audiowrite(struct file *fp, off_t *offp, struct uio *uio, kauth_cred_t cred, 1581 int ioflag) 1582 { 1583 struct audio_softc *sc; 1584 audio_file_t *file; 1585 int error; 1586 dev_t dev; 1587 1588 KASSERT(fp->f_audioctx); 1589 file = fp->f_audioctx; 1590 sc = file->sc; 1591 dev = file->dev; 1592 1593 if (fp->f_flag & O_NONBLOCK) 1594 ioflag |= IO_NDELAY; 1595 1596 switch (AUDIODEV(dev)) { 1597 case SOUND_DEVICE: 1598 case AUDIO_DEVICE: 1599 error = audio_write(sc, uio, ioflag, file); 1600 break; 1601 case AUDIOCTL_DEVICE: 1602 case MIXER_DEVICE: 1603 error = ENODEV; 1604 break; 1605 default: 1606 error = ENXIO; 1607 break; 1608 } 1609 1610 return error; 1611 } 1612 1613 static int 1614 audioioctl(struct file *fp, u_long cmd, void *addr) 1615 { 1616 struct audio_softc *sc; 1617 audio_file_t *file; 1618 struct lwp *l = curlwp; 1619 int error; 1620 dev_t dev; 1621 1622 KASSERT(fp->f_audioctx); 1623 file = fp->f_audioctx; 1624 sc = file->sc; 1625 dev = file->dev; 1626 1627 switch (AUDIODEV(dev)) { 1628 case SOUND_DEVICE: 1629 case AUDIO_DEVICE: 1630 case AUDIOCTL_DEVICE: 1631 mutex_enter(sc->sc_lock); 1632 device_active(sc->sc_dev, DVA_SYSTEM); 1633 mutex_exit(sc->sc_lock); 1634 if (IOCGROUP(cmd) == IOCGROUP(AUDIO_MIXER_READ)) 1635 error = mixer_ioctl(sc, cmd, addr, fp->f_flag, l); 1636 else 1637 error = audio_ioctl(dev, sc, cmd, addr, fp->f_flag, l, 1638 file); 1639 break; 1640 case MIXER_DEVICE: 1641 error = mixer_ioctl(sc, cmd, addr, fp->f_flag, l); 1642 break; 1643 default: 1644 error = ENXIO; 1645 break; 1646 } 1647 1648 return error; 1649 } 1650 1651 static int 1652 audiostat(struct file *fp, struct stat *st) 1653 { 1654 audio_file_t *file; 1655 1656 KASSERT(fp->f_audioctx); 1657 file = fp->f_audioctx; 1658 1659 memset(st, 0, sizeof(*st)); 1660 1661 st->st_dev = file->dev; 1662 st->st_uid = kauth_cred_geteuid(fp->f_cred); 1663 st->st_gid = kauth_cred_getegid(fp->f_cred); 1664 st->st_mode = S_IFCHR; 1665 return 0; 1666 } 1667 1668 static int 1669 audiopoll(struct file *fp, int events) 1670 { 1671 struct audio_softc *sc; 1672 audio_file_t *file; 1673 struct lwp *l = curlwp; 1674 int revents; 1675 dev_t dev; 1676 1677 KASSERT(fp->f_audioctx); 1678 file = fp->f_audioctx; 1679 sc = file->sc; 1680 dev = file->dev; 1681 1682 switch (AUDIODEV(dev)) { 1683 case SOUND_DEVICE: 1684 case AUDIO_DEVICE: 1685 revents = audio_poll(sc, events, l, file); 1686 break; 1687 case AUDIOCTL_DEVICE: 1688 case MIXER_DEVICE: 1689 revents = 0; 1690 break; 1691 default: 1692 revents = POLLERR; 1693 break; 1694 } 1695 1696 return revents; 1697 } 1698 1699 static int 1700 audiokqfilter(struct file *fp, struct knote *kn) 1701 { 1702 struct audio_softc *sc; 1703 audio_file_t *file; 1704 dev_t dev; 1705 int error; 1706 1707 KASSERT(fp->f_audioctx); 1708 file = fp->f_audioctx; 1709 sc = file->sc; 1710 dev = file->dev; 1711 1712 switch (AUDIODEV(dev)) { 1713 case SOUND_DEVICE: 1714 case AUDIO_DEVICE: 1715 error = audio_kqfilter(sc, file, kn); 1716 break; 1717 case AUDIOCTL_DEVICE: 1718 case MIXER_DEVICE: 1719 error = ENODEV; 1720 break; 1721 default: 1722 error = ENXIO; 1723 break; 1724 } 1725 1726 return error; 1727 } 1728 1729 static int 1730 audiommap(struct file *fp, off_t *offp, size_t len, int prot, int *flagsp, 1731 int *advicep, struct uvm_object **uobjp, int *maxprotp) 1732 { 1733 struct audio_softc *sc; 1734 audio_file_t *file; 1735 dev_t dev; 1736 int error; 1737 1738 KASSERT(fp->f_audioctx); 1739 file = fp->f_audioctx; 1740 sc = file->sc; 1741 dev = file->dev; 1742 1743 mutex_enter(sc->sc_lock); 1744 device_active(sc->sc_dev, DVA_SYSTEM); /* XXXJDM */ 1745 mutex_exit(sc->sc_lock); 1746 1747 switch (AUDIODEV(dev)) { 1748 case SOUND_DEVICE: 1749 case AUDIO_DEVICE: 1750 error = audio_mmap(sc, offp, len, prot, flagsp, advicep, 1751 uobjp, maxprotp, file); 1752 break; 1753 case AUDIOCTL_DEVICE: 1754 case MIXER_DEVICE: 1755 default: 1756 error = ENOTSUP; 1757 break; 1758 } 1759 1760 return error; 1761 } 1762 1763 1764 /* Exported interfaces for audiobell. */ 1765 1766 /* 1767 * Open for audiobell. 1768 * It stores allocated file to *filep. 1769 * If successful returns 0, otherwise errno. 1770 */ 1771 int 1772 audiobellopen(dev_t dev, audio_file_t **filep) 1773 { 1774 struct audio_softc *sc; 1775 int error; 1776 1777 /* Find the device */ 1778 sc = device_lookup_private(&audio_cd, AUDIOUNIT(dev)); 1779 if (sc == NULL || sc->hw_if == NULL) 1780 return ENXIO; 1781 1782 error = audio_enter_exclusive(sc); 1783 if (error) 1784 return error; 1785 1786 device_active(sc->sc_dev, DVA_SYSTEM); 1787 error = audio_open(dev, sc, FWRITE, 0, curlwp, filep); 1788 1789 audio_exit_exclusive(sc); 1790 return error; 1791 } 1792 1793 /* Close for audiobell */ 1794 int 1795 audiobellclose(audio_file_t *file) 1796 { 1797 struct audio_softc *sc; 1798 int error; 1799 1800 sc = file->sc; 1801 1802 device_active(sc->sc_dev, DVA_SYSTEM); 1803 error = audio_close(sc, file); 1804 1805 return error; 1806 } 1807 1808 /* Set sample rate for audiobell */ 1809 int 1810 audiobellsetrate(audio_file_t *file, u_int sample_rate) 1811 { 1812 struct audio_softc *sc; 1813 struct audio_info ai; 1814 int error; 1815 1816 sc = file->sc; 1817 1818 AUDIO_INITINFO(&ai); 1819 ai.play.sample_rate = sample_rate; 1820 1821 error = audio_enter_exclusive(sc); 1822 if (error) 1823 return error; 1824 error = audio_file_setinfo(sc, file, &ai); 1825 audio_exit_exclusive(sc); 1826 1827 return error; 1828 } 1829 1830 /* Playback for audiobell */ 1831 int 1832 audiobellwrite(audio_file_t *file, struct uio *uio) 1833 { 1834 struct audio_softc *sc; 1835 int error; 1836 1837 sc = file->sc; 1838 error = audio_write(sc, uio, 0, file); 1839 return error; 1840 } 1841 1842 1843 /* 1844 * Audio driver 1845 */ 1846 int 1847 audio_open(dev_t dev, struct audio_softc *sc, int flags, int ifmt, 1848 struct lwp *l, audio_file_t **bellfile) 1849 { 1850 struct audio_info ai; 1851 struct file *fp; 1852 audio_file_t *af; 1853 audio_ring_t *hwbuf; 1854 bool fullduplex; 1855 int fd; 1856 int error; 1857 1858 KASSERT(mutex_owned(sc->sc_lock)); 1859 KASSERT(sc->sc_exlock); 1860 1861 TRACE(1, "%sdev=%s flags=0x%x po=%d ro=%d", 1862 (audiodebug >= 3) ? "start " : "", 1863 ISDEVSOUND(dev) ? "sound" : "audio", 1864 flags, sc->sc_popens, sc->sc_ropens); 1865 1866 af = kmem_zalloc(sizeof(audio_file_t), KM_SLEEP); 1867 af->sc = sc; 1868 af->dev = dev; 1869 if ((flags & FWRITE) != 0 && audio_can_playback(sc)) 1870 af->mode |= AUMODE_PLAY | AUMODE_PLAY_ALL; 1871 if ((flags & FREAD) != 0 && audio_can_capture(sc)) 1872 af->mode |= AUMODE_RECORD; 1873 if (af->mode == 0) { 1874 error = ENXIO; 1875 goto bad1; 1876 } 1877 1878 fullduplex = (sc->sc_props & AUDIO_PROP_FULLDUPLEX); 1879 1880 /* 1881 * On half duplex hardware, 1882 * 1. if mode is (PLAY | REC), let mode PLAY. 1883 * 2. if mode is PLAY, let mode PLAY if no rec tracks, otherwise error. 1884 * 3. if mode is REC, let mode REC if no play tracks, otherwise error. 1885 */ 1886 if (fullduplex == false) { 1887 if ((af->mode & AUMODE_PLAY)) { 1888 if (sc->sc_ropens != 0) { 1889 TRACE(1, "record track already exists"); 1890 error = ENODEV; 1891 goto bad1; 1892 } 1893 /* Play takes precedence */ 1894 af->mode &= ~AUMODE_RECORD; 1895 } 1896 if ((af->mode & AUMODE_RECORD)) { 1897 if (sc->sc_popens != 0) { 1898 TRACE(1, "play track already exists"); 1899 error = ENODEV; 1900 goto bad1; 1901 } 1902 } 1903 } 1904 1905 /* Create tracks */ 1906 if ((af->mode & AUMODE_PLAY)) 1907 af->ptrack = audio_track_create(sc, sc->sc_pmixer); 1908 if ((af->mode & AUMODE_RECORD)) 1909 af->rtrack = audio_track_create(sc, sc->sc_rmixer); 1910 1911 /* Set parameters */ 1912 AUDIO_INITINFO(&ai); 1913 if (bellfile) { 1914 /* If audiobell, only sample_rate will be set later. */ 1915 ai.play.sample_rate = audio_default.sample_rate; 1916 ai.play.encoding = AUDIO_ENCODING_SLINEAR_NE; 1917 ai.play.channels = 1; 1918 ai.play.precision = 16; 1919 ai.play.pause = false; 1920 } else if (ISDEVAUDIO(dev)) { 1921 /* If /dev/audio, initialize everytime. */ 1922 ai.play.sample_rate = audio_default.sample_rate; 1923 ai.play.encoding = audio_default.encoding; 1924 ai.play.channels = audio_default.channels; 1925 ai.play.precision = audio_default.precision; 1926 ai.play.pause = false; 1927 ai.record.sample_rate = audio_default.sample_rate; 1928 ai.record.encoding = audio_default.encoding; 1929 ai.record.channels = audio_default.channels; 1930 ai.record.precision = audio_default.precision; 1931 ai.record.pause = false; 1932 } else { 1933 /* If /dev/sound, take over the previous parameters. */ 1934 ai.play.sample_rate = sc->sc_sound_pparams.sample_rate; 1935 ai.play.encoding = sc->sc_sound_pparams.encoding; 1936 ai.play.channels = sc->sc_sound_pparams.channels; 1937 ai.play.precision = sc->sc_sound_pparams.precision; 1938 ai.play.pause = sc->sc_sound_ppause; 1939 ai.record.sample_rate = sc->sc_sound_rparams.sample_rate; 1940 ai.record.encoding = sc->sc_sound_rparams.encoding; 1941 ai.record.channels = sc->sc_sound_rparams.channels; 1942 ai.record.precision = sc->sc_sound_rparams.precision; 1943 ai.record.pause = sc->sc_sound_rpause; 1944 } 1945 error = audio_file_setinfo(sc, af, &ai); 1946 if (error) 1947 goto bad2; 1948 1949 if (sc->sc_popens + sc->sc_ropens == 0) { 1950 /* First open */ 1951 1952 sc->sc_cred = kauth_cred_get(); 1953 kauth_cred_hold(sc->sc_cred); 1954 1955 if (sc->hw_if->open) { 1956 int hwflags; 1957 1958 /* 1959 * Call hw_if->open() only at first open of 1960 * combination of playback and recording. 1961 * On full duplex hardware, the flags passed to 1962 * hw_if->open() is always (FREAD | FWRITE) 1963 * regardless of this open()'s flags. 1964 * see also dev/isa/aria.c 1965 * On half duplex hardware, the flags passed to 1966 * hw_if->open() is either FREAD or FWRITE. 1967 * see also arch/evbarm/mini2440/audio_mini2440.c 1968 */ 1969 if (fullduplex) { 1970 hwflags = FREAD | FWRITE; 1971 } else { 1972 /* Construct hwflags from af->mode. */ 1973 hwflags = 0; 1974 if ((af->mode & AUMODE_PLAY) != 0) 1975 hwflags |= FWRITE; 1976 if ((af->mode & AUMODE_RECORD) != 0) 1977 hwflags |= FREAD; 1978 } 1979 1980 mutex_enter(sc->sc_intr_lock); 1981 error = sc->hw_if->open(sc->hw_hdl, hwflags); 1982 mutex_exit(sc->sc_intr_lock); 1983 if (error) 1984 goto bad2; 1985 } 1986 1987 /* 1988 * Set speaker mode when a half duplex. 1989 * XXX I'm not sure this is correct. 1990 */ 1991 if (1/*XXX*/) { 1992 if (sc->hw_if->speaker_ctl) { 1993 int on; 1994 if (af->ptrack) { 1995 on = 1; 1996 } else { 1997 on = 0; 1998 } 1999 mutex_enter(sc->sc_intr_lock); 2000 error = sc->hw_if->speaker_ctl(sc->hw_hdl, on); 2001 mutex_exit(sc->sc_intr_lock); 2002 if (error) 2003 goto bad3; 2004 } 2005 } 2006 } else if (sc->sc_multiuser == false) { 2007 uid_t euid = kauth_cred_geteuid(kauth_cred_get()); 2008 if (euid != 0 && euid != kauth_cred_geteuid(sc->sc_cred)) { 2009 error = EPERM; 2010 goto bad2; 2011 } 2012 } 2013 2014 /* Call init_output if this is the first playback open. */ 2015 if (af->ptrack && sc->sc_popens == 0) { 2016 if (sc->hw_if->init_output) { 2017 hwbuf = &sc->sc_pmixer->hwbuf; 2018 mutex_enter(sc->sc_intr_lock); 2019 error = sc->hw_if->init_output(sc->hw_hdl, 2020 hwbuf->mem, 2021 hwbuf->capacity * 2022 hwbuf->fmt.channels * hwbuf->fmt.stride / NBBY); 2023 mutex_exit(sc->sc_intr_lock); 2024 if (error) 2025 goto bad3; 2026 } 2027 } 2028 /* Call init_input if this is the first recording open. */ 2029 if (af->rtrack && sc->sc_ropens == 0) { 2030 if (sc->hw_if->init_input) { 2031 hwbuf = &sc->sc_rmixer->hwbuf; 2032 mutex_enter(sc->sc_intr_lock); 2033 error = sc->hw_if->init_input(sc->hw_hdl, 2034 hwbuf->mem, 2035 hwbuf->capacity * 2036 hwbuf->fmt.channels * hwbuf->fmt.stride / NBBY); 2037 mutex_exit(sc->sc_intr_lock); 2038 if (error) 2039 goto bad3; 2040 } 2041 } 2042 2043 if (bellfile == NULL) { 2044 error = fd_allocfile(&fp, &fd); 2045 if (error) 2046 goto bad3; 2047 } 2048 2049 /* 2050 * Count up finally. 2051 * Don't fail from here. 2052 */ 2053 if (af->ptrack) 2054 sc->sc_popens++; 2055 if (af->rtrack) 2056 sc->sc_ropens++; 2057 mutex_enter(sc->sc_intr_lock); 2058 SLIST_INSERT_HEAD(&sc->sc_files, af, entry); 2059 mutex_exit(sc->sc_intr_lock); 2060 2061 if (bellfile) { 2062 *bellfile = af; 2063 } else { 2064 error = fd_clone(fp, fd, flags, &audio_fileops, af); 2065 KASSERT(error == EMOVEFD); 2066 } 2067 2068 TRACEF(3, af, "done"); 2069 return error; 2070 2071 /* 2072 * Since track here is not yet linked to sc_files, 2073 * you can call track_destroy() without sc_intr_lock. 2074 */ 2075 bad3: 2076 if (sc->sc_popens + sc->sc_ropens == 0) { 2077 if (sc->hw_if->close) { 2078 mutex_enter(sc->sc_intr_lock); 2079 sc->hw_if->close(sc->hw_hdl); 2080 mutex_exit(sc->sc_intr_lock); 2081 } 2082 } 2083 bad2: 2084 if (af->rtrack) { 2085 audio_track_destroy(af->rtrack); 2086 af->rtrack = NULL; 2087 } 2088 if (af->ptrack) { 2089 audio_track_destroy(af->ptrack); 2090 af->ptrack = NULL; 2091 } 2092 bad1: 2093 kmem_free(af, sizeof(*af)); 2094 return error; 2095 } 2096 2097 /* 2098 * Must be called without sc_lock nor sc_exlock held. 2099 */ 2100 int 2101 audio_close(struct audio_softc *sc, audio_file_t *file) 2102 { 2103 audio_track_t *oldtrack; 2104 int error; 2105 2106 TRACEF(1, file, "%spid=%d.%d po=%d ro=%d", 2107 (audiodebug >= 3) ? "start " : "", 2108 (int)curproc->p_pid, (int)curlwp->l_lid, 2109 sc->sc_popens, sc->sc_ropens); 2110 KASSERTMSG(sc->sc_popens + sc->sc_ropens > 0, 2111 "sc->sc_popens=%d, sc->sc_ropens=%d", 2112 sc->sc_popens, sc->sc_ropens); 2113 2114 /* 2115 * Drain first. 2116 * It must be done before acquiring exclusive lock. 2117 */ 2118 if (file->ptrack) { 2119 mutex_enter(sc->sc_lock); 2120 audio_track_drain(sc, file->ptrack); 2121 mutex_exit(sc->sc_lock); 2122 } 2123 2124 /* Then, acquire exclusive lock to protect counters. */ 2125 /* XXX what should I do when an error occurs? */ 2126 error = audio_enter_exclusive(sc); 2127 if (error) 2128 return error; 2129 2130 if (file->ptrack) { 2131 /* Call hw halt_output if this is the last playback track. */ 2132 if (sc->sc_popens == 1 && sc->sc_pbusy) { 2133 error = audio_pmixer_halt(sc); 2134 if (error) { 2135 device_printf(sc->sc_dev, 2136 "halt_output failed with %d\n", error); 2137 } 2138 } 2139 2140 /* Destroy the track. */ 2141 oldtrack = file->ptrack; 2142 mutex_enter(sc->sc_intr_lock); 2143 file->ptrack = NULL; 2144 mutex_exit(sc->sc_intr_lock); 2145 TRACET(3, oldtrack, "dropframes=%" PRIu64, 2146 oldtrack->dropframes); 2147 audio_track_destroy(oldtrack); 2148 2149 KASSERT(sc->sc_popens > 0); 2150 sc->sc_popens--; 2151 2152 /* Restore mixing volume if all tracks are gone. */ 2153 if (sc->sc_popens == 0) { 2154 mutex_enter(sc->sc_intr_lock); 2155 sc->sc_pmixer->volume = 256; 2156 sc->sc_pmixer->voltimer = 0; 2157 mutex_exit(sc->sc_intr_lock); 2158 } 2159 } 2160 if (file->rtrack) { 2161 /* Call hw halt_input if this is the last recording track. */ 2162 if (sc->sc_ropens == 1 && sc->sc_rbusy) { 2163 error = audio_rmixer_halt(sc); 2164 if (error) { 2165 device_printf(sc->sc_dev, 2166 "halt_input failed with %d\n", error); 2167 } 2168 } 2169 2170 /* Destroy the track. */ 2171 oldtrack = file->rtrack; 2172 mutex_enter(sc->sc_intr_lock); 2173 file->rtrack = NULL; 2174 mutex_exit(sc->sc_intr_lock); 2175 TRACET(3, oldtrack, "dropframes=%" PRIu64, 2176 oldtrack->dropframes); 2177 audio_track_destroy(oldtrack); 2178 2179 KASSERT(sc->sc_ropens > 0); 2180 sc->sc_ropens--; 2181 } 2182 2183 /* Call hw close if this is the last track. */ 2184 if (sc->sc_popens + sc->sc_ropens == 0) { 2185 if (sc->hw_if->close) { 2186 TRACE(2, "hw_if close"); 2187 mutex_enter(sc->sc_intr_lock); 2188 sc->hw_if->close(sc->hw_hdl); 2189 mutex_exit(sc->sc_intr_lock); 2190 } 2191 2192 kauth_cred_free(sc->sc_cred); 2193 } 2194 2195 mutex_enter(sc->sc_intr_lock); 2196 SLIST_REMOVE(&sc->sc_files, file, audio_file, entry); 2197 mutex_exit(sc->sc_intr_lock); 2198 2199 TRACE(3, "done"); 2200 audio_exit_exclusive(sc); 2201 2202 kmem_free(file, sizeof(*file)); 2203 return 0; 2204 } 2205 2206 /* 2207 * Must be called without sc_lock nor sc_exlock held. 2208 */ 2209 int 2210 audio_read(struct audio_softc *sc, struct uio *uio, int ioflag, 2211 audio_file_t *file) 2212 { 2213 audio_track_t *track; 2214 audio_ring_t *usrbuf; 2215 audio_ring_t *input; 2216 int error; 2217 2218 /* 2219 * On half-duplex hardware, O_RDWR is treated as O_WRONLY. 2220 * However read() system call itself can be called because it's 2221 * opened with O_RDWR. So in this case, deny this read(). 2222 */ 2223 track = file->rtrack; 2224 if (track == NULL) { 2225 return EBADF; 2226 } 2227 2228 /* I think it's better than EINVAL. */ 2229 if (track->mmapped) 2230 return EPERM; 2231 2232 TRACET(2, track, "resid=%zd", uio->uio_resid); 2233 2234 #ifdef AUDIO_PM_IDLE 2235 mutex_enter(sc->sc_lock); 2236 if (device_is_active(&sc->sc_dev) || sc->sc_idle) 2237 device_active(&sc->sc_dev, DVA_SYSTEM); 2238 mutex_exit(sc->sc_lock); 2239 #endif 2240 2241 usrbuf = &track->usrbuf; 2242 input = track->input; 2243 2244 /* 2245 * The first read starts rmixer. 2246 */ 2247 error = audio_enter_exclusive(sc); 2248 if (error) 2249 return error; 2250 if (sc->sc_rbusy == false) 2251 audio_rmixer_start(sc); 2252 audio_exit_exclusive(sc); 2253 2254 error = 0; 2255 while (uio->uio_resid > 0 && error == 0) { 2256 int bytes; 2257 2258 TRACET(3, track, 2259 "while resid=%zd input=%d/%d/%d usrbuf=%d/%d/H%d", 2260 uio->uio_resid, 2261 input->head, input->used, input->capacity, 2262 usrbuf->head, usrbuf->used, track->usrbuf_usedhigh); 2263 2264 /* Wait when buffers are empty. */ 2265 mutex_enter(sc->sc_lock); 2266 for (;;) { 2267 bool empty; 2268 audio_track_lock_enter(track); 2269 empty = (input->used == 0 && usrbuf->used == 0); 2270 audio_track_lock_exit(track); 2271 if (!empty) 2272 break; 2273 2274 if ((ioflag & IO_NDELAY)) { 2275 mutex_exit(sc->sc_lock); 2276 return EWOULDBLOCK; 2277 } 2278 2279 TRACET(3, track, "sleep"); 2280 error = audio_track_waitio(sc, track); 2281 if (error) { 2282 mutex_exit(sc->sc_lock); 2283 return error; 2284 } 2285 } 2286 mutex_exit(sc->sc_lock); 2287 2288 audio_track_lock_enter(track); 2289 audio_track_record(track); 2290 2291 /* uiomove from usrbuf as much as possible. */ 2292 bytes = uimin(usrbuf->used, uio->uio_resid); 2293 while (bytes > 0) { 2294 int head = usrbuf->head; 2295 int len = uimin(bytes, usrbuf->capacity - head); 2296 error = uiomove((uint8_t *)usrbuf->mem + head, len, 2297 uio); 2298 if (error) { 2299 audio_track_lock_exit(track); 2300 device_printf(sc->sc_dev, 2301 "uiomove(len=%d) failed with %d\n", 2302 len, error); 2303 goto abort; 2304 } 2305 auring_take(usrbuf, len); 2306 track->useriobytes += len; 2307 TRACET(3, track, "uiomove(len=%d) usrbuf=%d/%d/C%d", 2308 len, 2309 usrbuf->head, usrbuf->used, usrbuf->capacity); 2310 bytes -= len; 2311 } 2312 2313 audio_track_lock_exit(track); 2314 } 2315 2316 abort: 2317 return error; 2318 } 2319 2320 2321 /* 2322 * Clear file's playback and/or record track buffer immediately. 2323 */ 2324 static void 2325 audio_file_clear(struct audio_softc *sc, audio_file_t *file) 2326 { 2327 2328 if (file->ptrack) 2329 audio_track_clear(sc, file->ptrack); 2330 if (file->rtrack) 2331 audio_track_clear(sc, file->rtrack); 2332 } 2333 2334 /* 2335 * Must be called without sc_lock nor sc_exlock held. 2336 */ 2337 int 2338 audio_write(struct audio_softc *sc, struct uio *uio, int ioflag, 2339 audio_file_t *file) 2340 { 2341 audio_track_t *track; 2342 audio_ring_t *usrbuf; 2343 audio_ring_t *outbuf; 2344 int error; 2345 2346 track = file->ptrack; 2347 KASSERT(track); 2348 2349 /* I think it's better than EINVAL. */ 2350 if (track->mmapped) 2351 return EPERM; 2352 2353 TRACET(2, track, "%sresid=%zd pid=%d.%d ioflag=0x%x", 2354 audiodebug >= 3 ? "begin " : "", 2355 uio->uio_resid, (int)curproc->p_pid, (int)curlwp->l_lid, ioflag); 2356 2357 if (uio->uio_resid == 0) { 2358 track->eofcounter++; 2359 return 0; 2360 } 2361 2362 #ifdef AUDIO_PM_IDLE 2363 mutex_enter(sc->sc_lock); 2364 if (device_is_active(&sc->sc_dev) || sc->sc_idle) 2365 device_active(&sc->sc_dev, DVA_SYSTEM); 2366 mutex_exit(sc->sc_lock); 2367 #endif 2368 2369 usrbuf = &track->usrbuf; 2370 outbuf = &track->outbuf; 2371 2372 /* 2373 * The first write starts pmixer. 2374 */ 2375 error = audio_enter_exclusive(sc); 2376 if (error) 2377 return error; 2378 if (sc->sc_pbusy == false) 2379 audio_pmixer_start(sc, false); 2380 audio_exit_exclusive(sc); 2381 2382 track->pstate = AUDIO_STATE_RUNNING; 2383 error = 0; 2384 while (uio->uio_resid > 0 && error == 0) { 2385 int bytes; 2386 2387 TRACET(3, track, "while resid=%zd usrbuf=%d/%d/H%d", 2388 uio->uio_resid, 2389 usrbuf->head, usrbuf->used, track->usrbuf_usedhigh); 2390 2391 /* Wait when buffers are full. */ 2392 mutex_enter(sc->sc_lock); 2393 for (;;) { 2394 bool full; 2395 audio_track_lock_enter(track); 2396 full = (usrbuf->used >= track->usrbuf_usedhigh && 2397 outbuf->used >= outbuf->capacity); 2398 audio_track_lock_exit(track); 2399 if (!full) 2400 break; 2401 2402 if ((ioflag & IO_NDELAY)) { 2403 error = EWOULDBLOCK; 2404 mutex_exit(sc->sc_lock); 2405 goto abort; 2406 } 2407 2408 TRACET(3, track, "sleep usrbuf=%d/H%d", 2409 usrbuf->used, track->usrbuf_usedhigh); 2410 error = audio_track_waitio(sc, track); 2411 if (error) { 2412 mutex_exit(sc->sc_lock); 2413 goto abort; 2414 } 2415 } 2416 mutex_exit(sc->sc_lock); 2417 2418 audio_track_lock_enter(track); 2419 2420 /* uiomove to usrbuf as much as possible. */ 2421 bytes = uimin(track->usrbuf_usedhigh - usrbuf->used, 2422 uio->uio_resid); 2423 while (bytes > 0) { 2424 int tail = auring_tail(usrbuf); 2425 int len = uimin(bytes, usrbuf->capacity - tail); 2426 error = uiomove((uint8_t *)usrbuf->mem + tail, len, 2427 uio); 2428 if (error) { 2429 audio_track_lock_exit(track); 2430 device_printf(sc->sc_dev, 2431 "uiomove(len=%d) failed with %d\n", 2432 len, error); 2433 goto abort; 2434 } 2435 auring_push(usrbuf, len); 2436 track->useriobytes += len; 2437 TRACET(3, track, "uiomove(len=%d) usrbuf=%d/%d/C%d", 2438 len, 2439 usrbuf->head, usrbuf->used, usrbuf->capacity); 2440 bytes -= len; 2441 } 2442 2443 /* Convert them as much as possible. */ 2444 while (usrbuf->used >= track->usrbuf_blksize && 2445 outbuf->used < outbuf->capacity) { 2446 audio_track_play(track); 2447 } 2448 2449 audio_track_lock_exit(track); 2450 } 2451 2452 abort: 2453 TRACET(3, track, "done error=%d", error); 2454 return error; 2455 } 2456 2457 /* 2458 * Must be called without sc_lock nor sc_exlock held. 2459 */ 2460 int 2461 audio_ioctl(dev_t dev, struct audio_softc *sc, u_long cmd, void *addr, int flag, 2462 struct lwp *l, audio_file_t *file) 2463 { 2464 struct audio_offset *ao; 2465 struct audio_info ai; 2466 audio_track_t *track; 2467 audio_encoding_t *ae; 2468 audio_format_query_t *query; 2469 u_int stamp; 2470 u_int offs; 2471 int fd; 2472 int index; 2473 int error; 2474 2475 #if defined(AUDIO_DEBUG) 2476 const char *ioctlnames[] = { 2477 " AUDIO_GETINFO", /* 21 */ 2478 " AUDIO_SETINFO", /* 22 */ 2479 " AUDIO_DRAIN", /* 23 */ 2480 " AUDIO_FLUSH", /* 24 */ 2481 " AUDIO_WSEEK", /* 25 */ 2482 " AUDIO_RERROR", /* 26 */ 2483 " AUDIO_GETDEV", /* 27 */ 2484 " AUDIO_GETENC", /* 28 */ 2485 " AUDIO_GETFD", /* 29 */ 2486 " AUDIO_SETFD", /* 30 */ 2487 " AUDIO_PERROR", /* 31 */ 2488 " AUDIO_GETIOFFS", /* 32 */ 2489 " AUDIO_GETOOFFS", /* 33 */ 2490 " AUDIO_GETPROPS", /* 34 */ 2491 " AUDIO_GETBUFINFO", /* 35 */ 2492 " AUDIO_SETCHAN", /* 36 */ 2493 " AUDIO_GETCHAN", /* 37 */ 2494 " AUDIO_QUERYFORMAT", /* 38 */ 2495 " AUDIO_GETFORMAT", /* 39 */ 2496 " AUDIO_SETFORMAT", /* 40 */ 2497 }; 2498 int nameidx = (cmd & 0xff); 2499 const char *ioctlname = ""; 2500 if (21 <= nameidx && nameidx <= 21 + __arraycount(ioctlnames)) 2501 ioctlname = ioctlnames[nameidx - 21]; 2502 TRACEF(2, file, "(%lu,'%c',%lu)%s pid=%d.%d", 2503 IOCPARM_LEN(cmd), (char)IOCGROUP(cmd), cmd&0xff, ioctlname, 2504 (int)curproc->p_pid, (int)l->l_lid); 2505 #endif 2506 2507 error = 0; 2508 switch (cmd) { 2509 case FIONBIO: 2510 /* All handled in the upper FS layer. */ 2511 break; 2512 2513 case FIONREAD: 2514 /* Get the number of bytes that can be read. */ 2515 if (file->rtrack) { 2516 *(int *)addr = audio_track_readablebytes(file->rtrack); 2517 } else { 2518 *(int *)addr = 0; 2519 } 2520 break; 2521 2522 case FIOASYNC: 2523 /* Set/Clear ASYNC I/O. */ 2524 if (*(int *)addr) { 2525 file->async_audio = curproc->p_pid; 2526 TRACEF(2, file, "FIOASYNC pid %d", file->async_audio); 2527 } else { 2528 file->async_audio = 0; 2529 TRACEF(2, file, "FIOASYNC off"); 2530 } 2531 break; 2532 2533 case AUDIO_FLUSH: 2534 /* XXX TODO: clear errors and restart? */ 2535 audio_file_clear(sc, file); 2536 break; 2537 2538 case AUDIO_RERROR: 2539 /* 2540 * Number of read bytes dropped. We don't know where 2541 * or when they were dropped (including conversion stage). 2542 * Therefore, the number of accurate bytes or samples is 2543 * also unknown. 2544 */ 2545 track = file->rtrack; 2546 if (track) { 2547 *(int *)addr = frametobyte(&track->usrbuf.fmt, 2548 track->dropframes); 2549 } 2550 break; 2551 2552 case AUDIO_PERROR: 2553 /* 2554 * Number of write bytes dropped. We don't know where 2555 * or when they were dropped (including conversion stage). 2556 * Therefore, the number of accurate bytes or samples is 2557 * also unknown. 2558 */ 2559 track = file->ptrack; 2560 if (track) { 2561 *(int *)addr = frametobyte(&track->usrbuf.fmt, 2562 track->dropframes); 2563 } 2564 break; 2565 2566 case AUDIO_GETIOFFS: 2567 /* XXX TODO */ 2568 ao = (struct audio_offset *)addr; 2569 ao->samples = 0; 2570 ao->deltablks = 0; 2571 ao->offset = 0; 2572 break; 2573 2574 case AUDIO_GETOOFFS: 2575 ao = (struct audio_offset *)addr; 2576 track = file->ptrack; 2577 if (track == NULL) { 2578 ao->samples = 0; 2579 ao->deltablks = 0; 2580 ao->offset = 0; 2581 break; 2582 } 2583 mutex_enter(sc->sc_lock); 2584 mutex_enter(sc->sc_intr_lock); 2585 /* figure out where next DMA will start */ 2586 stamp = track->usrbuf_stamp; 2587 offs = track->usrbuf.head; 2588 mutex_exit(sc->sc_intr_lock); 2589 mutex_exit(sc->sc_lock); 2590 2591 ao->samples = stamp; 2592 ao->deltablks = (stamp / track->usrbuf_blksize) - 2593 (track->usrbuf_stamp_last / track->usrbuf_blksize); 2594 track->usrbuf_stamp_last = stamp; 2595 offs = rounddown(offs, track->usrbuf_blksize) 2596 + track->usrbuf_blksize; 2597 if (offs >= track->usrbuf.capacity) 2598 offs -= track->usrbuf.capacity; 2599 ao->offset = offs; 2600 2601 TRACET(3, track, "GETOOFFS: samples=%u deltablks=%u offset=%u", 2602 ao->samples, ao->deltablks, ao->offset); 2603 break; 2604 2605 case AUDIO_WSEEK: 2606 /* XXX return value does not include outbuf one. */ 2607 if (file->ptrack) 2608 *(u_long *)addr = file->ptrack->usrbuf.used; 2609 break; 2610 2611 case AUDIO_SETINFO: 2612 error = audio_enter_exclusive(sc); 2613 if (error) 2614 break; 2615 error = audio_file_setinfo(sc, file, (struct audio_info *)addr); 2616 if (error) { 2617 audio_exit_exclusive(sc); 2618 break; 2619 } 2620 /* XXX TODO: update last_ai if /dev/sound ? */ 2621 if (ISDEVSOUND(dev)) 2622 error = audiogetinfo(sc, &sc->sc_ai, 0, file); 2623 audio_exit_exclusive(sc); 2624 break; 2625 2626 case AUDIO_GETINFO: 2627 error = audio_enter_exclusive(sc); 2628 if (error) 2629 break; 2630 error = audiogetinfo(sc, (struct audio_info *)addr, 1, file); 2631 audio_exit_exclusive(sc); 2632 break; 2633 2634 case AUDIO_GETBUFINFO: 2635 mutex_enter(sc->sc_lock); 2636 error = audiogetinfo(sc, (struct audio_info *)addr, 0, file); 2637 mutex_exit(sc->sc_lock); 2638 break; 2639 2640 case AUDIO_DRAIN: 2641 if (file->ptrack) { 2642 mutex_enter(sc->sc_lock); 2643 error = audio_track_drain(sc, file->ptrack); 2644 mutex_exit(sc->sc_lock); 2645 } 2646 break; 2647 2648 case AUDIO_GETDEV: 2649 mutex_enter(sc->sc_lock); 2650 error = sc->hw_if->getdev(sc->hw_hdl, (audio_device_t *)addr); 2651 mutex_exit(sc->sc_lock); 2652 break; 2653 2654 case AUDIO_GETENC: 2655 ae = (audio_encoding_t *)addr; 2656 index = ae->index; 2657 if (index < 0 || index >= __arraycount(audio_encodings)) { 2658 error = EINVAL; 2659 break; 2660 } 2661 *ae = audio_encodings[index]; 2662 ae->index = index; 2663 /* 2664 * EMULATED always. 2665 * EMULATED flag at that time used to mean that it could 2666 * not be passed directly to the hardware as-is. But 2667 * currently, all formats including hardware native is not 2668 * passed directly to the hardware. So I set EMULATED 2669 * flag for all formats. 2670 */ 2671 ae->flags = AUDIO_ENCODINGFLAG_EMULATED; 2672 break; 2673 2674 case AUDIO_GETFD: 2675 /* 2676 * Returns the current setting of full duplex mode. 2677 * If HW has full duplex mode and there are two mixers, 2678 * it is full duplex. Otherwise half duplex. 2679 */ 2680 mutex_enter(sc->sc_lock); 2681 fd = (sc->sc_props & AUDIO_PROP_FULLDUPLEX) 2682 && (sc->sc_pmixer && sc->sc_rmixer); 2683 mutex_exit(sc->sc_lock); 2684 *(int *)addr = fd; 2685 break; 2686 2687 case AUDIO_GETPROPS: 2688 *(int *)addr = sc->sc_props; 2689 break; 2690 2691 case AUDIO_QUERYFORMAT: 2692 query = (audio_format_query_t *)addr; 2693 if (sc->hw_if->query_format) { 2694 mutex_enter(sc->sc_lock); 2695 error = sc->hw_if->query_format(sc->hw_hdl, query); 2696 mutex_exit(sc->sc_lock); 2697 /* Hide internal infomations */ 2698 query->fmt.driver_data = NULL; 2699 } else { 2700 error = ENODEV; 2701 } 2702 break; 2703 2704 case AUDIO_GETFORMAT: 2705 audio_mixers_get_format(sc, (struct audio_info *)addr); 2706 break; 2707 2708 case AUDIO_SETFORMAT: 2709 mutex_enter(sc->sc_lock); 2710 audio_mixers_get_format(sc, &ai); 2711 error = audio_mixers_set_format(sc, (struct audio_info *)addr); 2712 if (error) { 2713 /* Rollback */ 2714 audio_mixers_set_format(sc, &ai); 2715 } 2716 mutex_exit(sc->sc_lock); 2717 break; 2718 2719 case AUDIO_SETFD: 2720 case AUDIO_SETCHAN: 2721 case AUDIO_GETCHAN: 2722 /* Obsoleted */ 2723 break; 2724 2725 default: 2726 if (sc->hw_if->dev_ioctl) { 2727 error = audio_enter_exclusive(sc); 2728 if (error) 2729 break; 2730 error = sc->hw_if->dev_ioctl(sc->hw_hdl, 2731 cmd, addr, flag, l); 2732 audio_exit_exclusive(sc); 2733 } else { 2734 TRACEF(2, file, "unknown ioctl"); 2735 error = EINVAL; 2736 } 2737 break; 2738 } 2739 TRACEF(2, file, "(%lu,'%c',%lu)%s result %d", 2740 IOCPARM_LEN(cmd), (char)IOCGROUP(cmd), cmd&0xff, ioctlname, 2741 error); 2742 return error; 2743 } 2744 2745 /* 2746 * Returns the number of bytes that can be read on recording buffer. 2747 */ 2748 static __inline int 2749 audio_track_readablebytes(const audio_track_t *track) 2750 { 2751 int bytes; 2752 2753 KASSERT(track); 2754 KASSERT(track->mode == AUMODE_RECORD); 2755 2756 /* 2757 * Although usrbuf is primarily readable data, recorded data 2758 * also stays in track->input until reading. So it is necessary 2759 * to add it. track->input is in frame, usrbuf is in byte. 2760 */ 2761 bytes = track->usrbuf.used + 2762 track->input->used * frametobyte(&track->usrbuf.fmt, 1); 2763 return bytes; 2764 } 2765 2766 /* 2767 * Must be called without sc_lock nor sc_exlock held. 2768 */ 2769 int 2770 audio_poll(struct audio_softc *sc, int events, struct lwp *l, 2771 audio_file_t *file) 2772 { 2773 audio_track_t *track; 2774 int revents; 2775 bool in_is_valid; 2776 bool out_is_valid; 2777 2778 #if defined(AUDIO_DEBUG) 2779 #define POLLEV_BITMAP "\177\020" \ 2780 "b\10WRBAND\0" \ 2781 "b\7RDBAND\0" "b\6RDNORM\0" "b\5NVAL\0" "b\4HUP\0" \ 2782 "b\3ERR\0" "b\2OUT\0" "b\1PRI\0" "b\0IN\0" 2783 char evbuf[64]; 2784 snprintb(evbuf, sizeof(evbuf), POLLEV_BITMAP, events); 2785 TRACEF(2, file, "pid=%d.%d events=%s", 2786 (int)curproc->p_pid, (int)l->l_lid, evbuf); 2787 #endif 2788 2789 revents = 0; 2790 in_is_valid = false; 2791 out_is_valid = false; 2792 if (events & (POLLIN | POLLRDNORM)) { 2793 track = file->rtrack; 2794 if (track) { 2795 int used; 2796 in_is_valid = true; 2797 used = audio_track_readablebytes(track); 2798 if (used > 0) 2799 revents |= events & (POLLIN | POLLRDNORM); 2800 } 2801 } 2802 if (events & (POLLOUT | POLLWRNORM)) { 2803 track = file->ptrack; 2804 if (track) { 2805 out_is_valid = true; 2806 if (track->usrbuf.used <= track->usrbuf_usedlow) 2807 revents |= events & (POLLOUT | POLLWRNORM); 2808 } 2809 } 2810 2811 if (revents == 0) { 2812 mutex_enter(sc->sc_lock); 2813 if (in_is_valid) { 2814 TRACEF(3, file, "selrecord rsel"); 2815 selrecord(l, &sc->sc_rsel); 2816 } 2817 if (out_is_valid) { 2818 TRACEF(3, file, "selrecord wsel"); 2819 selrecord(l, &sc->sc_wsel); 2820 } 2821 mutex_exit(sc->sc_lock); 2822 } 2823 2824 #if defined(AUDIO_DEBUG) 2825 snprintb(evbuf, sizeof(evbuf), POLLEV_BITMAP, revents); 2826 TRACEF(2, file, "revents=%s", evbuf); 2827 #endif 2828 return revents; 2829 } 2830 2831 static const struct filterops audioread_filtops = { 2832 .f_isfd = 1, 2833 .f_attach = NULL, 2834 .f_detach = filt_audioread_detach, 2835 .f_event = filt_audioread_event, 2836 }; 2837 2838 static void 2839 filt_audioread_detach(struct knote *kn) 2840 { 2841 struct audio_softc *sc; 2842 audio_file_t *file; 2843 2844 file = kn->kn_hook; 2845 sc = file->sc; 2846 TRACEF(3, file, ""); 2847 2848 mutex_enter(sc->sc_lock); 2849 SLIST_REMOVE(&sc->sc_rsel.sel_klist, kn, knote, kn_selnext); 2850 mutex_exit(sc->sc_lock); 2851 } 2852 2853 static int 2854 filt_audioread_event(struct knote *kn, long hint) 2855 { 2856 audio_file_t *file; 2857 audio_track_t *track; 2858 2859 file = kn->kn_hook; 2860 track = file->rtrack; 2861 2862 /* 2863 * kn_data must contain the number of bytes can be read. 2864 * The return value indicates whether the event occurs or not. 2865 */ 2866 2867 if (track == NULL) { 2868 /* can not read with this descriptor. */ 2869 kn->kn_data = 0; 2870 return 0; 2871 } 2872 2873 kn->kn_data = audio_track_readablebytes(track); 2874 TRACEF(3, file, "data=%" PRId64, kn->kn_data); 2875 return kn->kn_data > 0; 2876 } 2877 2878 static const struct filterops audiowrite_filtops = { 2879 .f_isfd = 1, 2880 .f_attach = NULL, 2881 .f_detach = filt_audiowrite_detach, 2882 .f_event = filt_audiowrite_event, 2883 }; 2884 2885 static void 2886 filt_audiowrite_detach(struct knote *kn) 2887 { 2888 struct audio_softc *sc; 2889 audio_file_t *file; 2890 2891 file = kn->kn_hook; 2892 sc = file->sc; 2893 TRACEF(3, file, ""); 2894 2895 mutex_enter(sc->sc_lock); 2896 SLIST_REMOVE(&sc->sc_wsel.sel_klist, kn, knote, kn_selnext); 2897 mutex_exit(sc->sc_lock); 2898 } 2899 2900 static int 2901 filt_audiowrite_event(struct knote *kn, long hint) 2902 { 2903 audio_file_t *file; 2904 audio_track_t *track; 2905 2906 file = kn->kn_hook; 2907 track = file->ptrack; 2908 2909 /* 2910 * kn_data must contain the number of bytes can be write. 2911 * The return value indicates whether the event occurs or not. 2912 */ 2913 2914 if (track == NULL) { 2915 /* can not write with this descriptor. */ 2916 kn->kn_data = 0; 2917 return 0; 2918 } 2919 2920 kn->kn_data = track->usrbuf_usedhigh - track->usrbuf.used; 2921 TRACEF(3, file, "data=%" PRId64, kn->kn_data); 2922 return (track->usrbuf.used < track->usrbuf_usedlow); 2923 } 2924 2925 /* 2926 * Must be called without sc_lock nor sc_exlock held. 2927 */ 2928 int 2929 audio_kqfilter(struct audio_softc *sc, audio_file_t *file, struct knote *kn) 2930 { 2931 struct klist *klist; 2932 2933 TRACEF(3, file, "kn=%p kn_filter=%x", kn, (int)kn->kn_filter); 2934 2935 switch (kn->kn_filter) { 2936 case EVFILT_READ: 2937 klist = &sc->sc_rsel.sel_klist; 2938 kn->kn_fop = &audioread_filtops; 2939 break; 2940 2941 case EVFILT_WRITE: 2942 klist = &sc->sc_wsel.sel_klist; 2943 kn->kn_fop = &audiowrite_filtops; 2944 break; 2945 2946 default: 2947 return EINVAL; 2948 } 2949 2950 kn->kn_hook = file; 2951 2952 mutex_enter(sc->sc_lock); 2953 SLIST_INSERT_HEAD(klist, kn, kn_selnext); 2954 mutex_exit(sc->sc_lock); 2955 2956 return 0; 2957 } 2958 2959 /* 2960 * Must be called without sc_lock nor sc_exlock held. 2961 */ 2962 int 2963 audio_mmap(struct audio_softc *sc, off_t *offp, size_t len, int prot, 2964 int *flagsp, int *advicep, struct uvm_object **uobjp, int *maxprotp, 2965 audio_file_t *file) 2966 { 2967 audio_track_t *track; 2968 vsize_t vsize; 2969 int error; 2970 2971 TRACEF(2, file, "off=%lld, prot=%d", (long long)(*offp), prot); 2972 2973 if (*offp < 0) 2974 return EINVAL; 2975 2976 #if 0 2977 /* XXX 2978 * The idea here was to use the protection to determine if 2979 * we are mapping the read or write buffer, but it fails. 2980 * The VM system is broken in (at least) two ways. 2981 * 1) If you map memory VM_PROT_WRITE you SIGSEGV 2982 * when writing to it, so VM_PROT_READ|VM_PROT_WRITE 2983 * has to be used for mmapping the play buffer. 2984 * 2) Even if calling mmap() with VM_PROT_READ|VM_PROT_WRITE 2985 * audio_mmap will get called at some point with VM_PROT_READ 2986 * only. 2987 * So, alas, we always map the play buffer for now. 2988 */ 2989 if (prot == (VM_PROT_READ|VM_PROT_WRITE) || 2990 prot == VM_PROT_WRITE) 2991 track = file->ptrack; 2992 else if (prot == VM_PROT_READ) 2993 track = file->rtrack; 2994 else 2995 return EINVAL; 2996 #else 2997 track = file->ptrack; 2998 #endif 2999 if (track == NULL) 3000 return EACCES; 3001 3002 vsize = roundup2(MAX(track->usrbuf.capacity, PAGE_SIZE), PAGE_SIZE); 3003 if (len > vsize) 3004 return EOVERFLOW; 3005 if (*offp > (uint)(vsize - len)) 3006 return EOVERFLOW; 3007 3008 /* XXX TODO: what happens when mmap twice. */ 3009 if (!track->mmapped) { 3010 track->mmapped = true; 3011 3012 if (!track->is_pause) { 3013 error = audio_enter_exclusive(sc); 3014 if (error) 3015 return error; 3016 if (sc->sc_pbusy == false) 3017 audio_pmixer_start(sc, true); 3018 audio_exit_exclusive(sc); 3019 } 3020 /* XXX mmapping record buffer is not supported */ 3021 } 3022 3023 /* get ringbuffer */ 3024 *uobjp = track->uobj; 3025 3026 /* Acquire a reference for the mmap. munmap will release. */ 3027 uao_reference(*uobjp); 3028 *maxprotp = prot; 3029 *advicep = UVM_ADV_RANDOM; 3030 *flagsp = MAP_SHARED; 3031 return 0; 3032 } 3033 3034 /* 3035 * /dev/audioctl has to be able to open at any time without interference 3036 * with any /dev/audio or /dev/sound. 3037 */ 3038 static int 3039 audioctl_open(dev_t dev, struct audio_softc *sc, int flags, int ifmt, 3040 struct lwp *l) 3041 { 3042 struct file *fp; 3043 audio_file_t *af; 3044 int fd; 3045 int error; 3046 3047 KASSERT(mutex_owned(sc->sc_lock)); 3048 KASSERT(sc->sc_exlock); 3049 3050 TRACE(1, ""); 3051 3052 error = fd_allocfile(&fp, &fd); 3053 if (error) 3054 return error; 3055 3056 af = kmem_zalloc(sizeof(audio_file_t), KM_SLEEP); 3057 af->sc = sc; 3058 af->dev = dev; 3059 3060 /* Not necessary to insert sc_files. */ 3061 3062 error = fd_clone(fp, fd, flags, &audio_fileops, af); 3063 KASSERT(error == EMOVEFD); 3064 3065 return error; 3066 } 3067 3068 static int 3069 audioctl_close(struct audio_softc *sc, audio_file_t *file) 3070 { 3071 3072 kmem_free(file, sizeof(*file)); 3073 return 0; 3074 } 3075 3076 /* 3077 * Free 'mem' if available, and initialize the pointer. 3078 * For this reason, this is implemented as macro. 3079 */ 3080 #define audio_free(mem) do { \ 3081 if (mem != NULL) { \ 3082 kern_free(mem); \ 3083 mem = NULL; \ 3084 } \ 3085 } while (0) 3086 3087 /* 3088 * (Re)allocate 'memblock' with specified 'bytes'. 3089 * bytes must not be 0. 3090 * This function never returns NULL. 3091 */ 3092 static void * 3093 audio_realloc(void *memblock, size_t bytes) 3094 { 3095 3096 KASSERT(bytes != 0); 3097 audio_free(memblock); 3098 return kern_malloc(bytes, M_WAITOK); 3099 } 3100 3101 /* 3102 * (Re)allocate usrbuf with 'newbufsize' bytes. 3103 * Use this function for usrbuf because only usrbuf can be mmapped. 3104 * If successful, it updates track->usrbuf.mem, track->usrbuf.capacity and 3105 * returns 0. Otherwise, it clears track->usrbuf.mem, track->usrbuf.capacity 3106 * and returns errno. 3107 * It must be called before updating usrbuf.capacity. 3108 */ 3109 static int 3110 audio_realloc_usrbuf(audio_track_t *track, int newbufsize) 3111 { 3112 struct audio_softc *sc; 3113 vaddr_t vstart; 3114 vsize_t oldvsize; 3115 vsize_t newvsize; 3116 int error; 3117 3118 KASSERT(newbufsize > 0); 3119 sc = track->mixer->sc; 3120 3121 /* Get a nonzero multiple of PAGE_SIZE */ 3122 newvsize = roundup2(MAX(newbufsize, PAGE_SIZE), PAGE_SIZE); 3123 3124 if (track->usrbuf.mem != NULL) { 3125 oldvsize = roundup2(MAX(track->usrbuf.capacity, PAGE_SIZE), 3126 PAGE_SIZE); 3127 if (oldvsize == newvsize) { 3128 track->usrbuf.capacity = newbufsize; 3129 return 0; 3130 } 3131 vstart = (vaddr_t)track->usrbuf.mem; 3132 uvm_unmap(kernel_map, vstart, vstart + oldvsize); 3133 /* uvm_unmap also detach uobj */ 3134 track->uobj = NULL; /* paranoia */ 3135 track->usrbuf.mem = NULL; 3136 } 3137 3138 /* Create a uvm anonymous object */ 3139 track->uobj = uao_create(newvsize, 0); 3140 3141 /* Map it into the kernel virtual address space */ 3142 vstart = 0; 3143 error = uvm_map(kernel_map, &vstart, newvsize, track->uobj, 0, 0, 3144 UVM_MAPFLAG(UVM_PROT_RW, UVM_PROT_RW, UVM_INH_NONE, 3145 UVM_ADV_RANDOM, 0)); 3146 if (error) { 3147 device_printf(sc->sc_dev, "uvm_map failed with %d\n", error); 3148 uao_detach(track->uobj); /* release reference */ 3149 goto abort; 3150 } 3151 3152 error = uvm_map_pageable(kernel_map, vstart, vstart + newvsize, 3153 false, 0); 3154 if (error) { 3155 device_printf(sc->sc_dev, "uvm_map_pageable failed with %d\n", 3156 error); 3157 uvm_unmap(kernel_map, vstart, vstart + newvsize); 3158 /* uvm_unmap also detach uobj */ 3159 goto abort; 3160 } 3161 3162 track->usrbuf.mem = (void *)vstart; 3163 track->usrbuf.capacity = newbufsize; 3164 memset(track->usrbuf.mem, 0, newvsize); 3165 return 0; 3166 3167 /* failure */ 3168 abort: 3169 track->uobj = NULL; /* paranoia */ 3170 track->usrbuf.mem = NULL; 3171 track->usrbuf.capacity = 0; 3172 return error; 3173 } 3174 3175 /* 3176 * Free usrbuf (if available). 3177 */ 3178 static void 3179 audio_free_usrbuf(audio_track_t *track) 3180 { 3181 vaddr_t vstart; 3182 vsize_t vsize; 3183 3184 vstart = (vaddr_t)track->usrbuf.mem; 3185 vsize = roundup2(MAX(track->usrbuf.capacity, PAGE_SIZE), PAGE_SIZE); 3186 if (track->usrbuf.mem != NULL) { 3187 /* 3188 * Unmap the kernel mapping. uvm_unmap releases the 3189 * reference to the uvm object, and this should be the 3190 * last virtual mapping of the uvm object, so no need 3191 * to explicitly release (`detach') the object. 3192 */ 3193 uvm_unmap(kernel_map, vstart, vstart + vsize); 3194 3195 track->uobj = NULL; 3196 track->usrbuf.mem = NULL; 3197 track->usrbuf.capacity = 0; 3198 } 3199 } 3200 3201 /* 3202 * This filter changes the volume for each channel. 3203 * arg->context points track->ch_volume[]. 3204 */ 3205 static void 3206 audio_track_chvol(audio_filter_arg_t *arg) 3207 { 3208 int16_t *ch_volume; 3209 const aint_t *s; 3210 aint_t *d; 3211 u_int i; 3212 u_int ch; 3213 u_int channels; 3214 3215 DIAGNOSTIC_filter_arg(arg); 3216 KASSERT(arg->srcfmt->channels == arg->dstfmt->channels); 3217 KASSERT(arg->context != NULL); 3218 KASSERT(arg->srcfmt->channels <= AUDIO_MAX_CHANNELS); 3219 3220 s = arg->src; 3221 d = arg->dst; 3222 ch_volume = arg->context; 3223 3224 channels = arg->srcfmt->channels; 3225 for (i = 0; i < arg->count; i++) { 3226 for (ch = 0; ch < channels; ch++) { 3227 aint2_t val; 3228 val = *s++; 3229 val = AUDIO_SCALEDOWN(val * ch_volume[ch], 8); 3230 *d++ = (aint_t)val; 3231 } 3232 } 3233 } 3234 3235 /* 3236 * This filter performs conversion from stereo (or more channels) to mono. 3237 */ 3238 static void 3239 audio_track_chmix_mixLR(audio_filter_arg_t *arg) 3240 { 3241 const aint_t *s; 3242 aint_t *d; 3243 u_int i; 3244 3245 DIAGNOSTIC_filter_arg(arg); 3246 3247 s = arg->src; 3248 d = arg->dst; 3249 3250 for (i = 0; i < arg->count; i++) { 3251 *d++ = AUDIO_SCALEDOWN(s[0], 1) + AUDIO_SCALEDOWN(s[1], 1); 3252 s += arg->srcfmt->channels; 3253 } 3254 } 3255 3256 /* 3257 * This filter performs conversion from mono to stereo (or more channels). 3258 */ 3259 static void 3260 audio_track_chmix_dupLR(audio_filter_arg_t *arg) 3261 { 3262 const aint_t *s; 3263 aint_t *d; 3264 u_int i; 3265 u_int ch; 3266 u_int dstchannels; 3267 3268 DIAGNOSTIC_filter_arg(arg); 3269 3270 s = arg->src; 3271 d = arg->dst; 3272 dstchannels = arg->dstfmt->channels; 3273 3274 for (i = 0; i < arg->count; i++) { 3275 d[0] = s[0]; 3276 d[1] = s[0]; 3277 s++; 3278 d += dstchannels; 3279 } 3280 if (dstchannels > 2) { 3281 d = arg->dst; 3282 for (i = 0; i < arg->count; i++) { 3283 for (ch = 2; ch < dstchannels; ch++) { 3284 d[ch] = 0; 3285 } 3286 d += dstchannels; 3287 } 3288 } 3289 } 3290 3291 /* 3292 * This filter shrinks M channels into N channels. 3293 * Extra channels are discarded. 3294 */ 3295 static void 3296 audio_track_chmix_shrink(audio_filter_arg_t *arg) 3297 { 3298 const aint_t *s; 3299 aint_t *d; 3300 u_int i; 3301 u_int ch; 3302 3303 DIAGNOSTIC_filter_arg(arg); 3304 3305 s = arg->src; 3306 d = arg->dst; 3307 3308 for (i = 0; i < arg->count; i++) { 3309 for (ch = 0; ch < arg->dstfmt->channels; ch++) { 3310 *d++ = s[ch]; 3311 } 3312 s += arg->srcfmt->channels; 3313 } 3314 } 3315 3316 /* 3317 * This filter expands M channels into N channels. 3318 * Silence is inserted for missing channels. 3319 */ 3320 static void 3321 audio_track_chmix_expand(audio_filter_arg_t *arg) 3322 { 3323 const aint_t *s; 3324 aint_t *d; 3325 u_int i; 3326 u_int ch; 3327 u_int srcchannels; 3328 u_int dstchannels; 3329 3330 DIAGNOSTIC_filter_arg(arg); 3331 3332 s = arg->src; 3333 d = arg->dst; 3334 3335 srcchannels = arg->srcfmt->channels; 3336 dstchannels = arg->dstfmt->channels; 3337 for (i = 0; i < arg->count; i++) { 3338 for (ch = 0; ch < srcchannels; ch++) { 3339 *d++ = *s++; 3340 } 3341 for (; ch < dstchannels; ch++) { 3342 *d++ = 0; 3343 } 3344 } 3345 } 3346 3347 /* 3348 * This filter performs frequency conversion (up sampling). 3349 * It uses linear interpolation. 3350 */ 3351 static void 3352 audio_track_freq_up(audio_filter_arg_t *arg) 3353 { 3354 audio_track_t *track; 3355 audio_ring_t *src; 3356 audio_ring_t *dst; 3357 const aint_t *s; 3358 aint_t *d; 3359 aint_t prev[AUDIO_MAX_CHANNELS]; 3360 aint_t curr[AUDIO_MAX_CHANNELS]; 3361 aint_t grad[AUDIO_MAX_CHANNELS]; 3362 u_int i; 3363 u_int t; 3364 u_int step; 3365 u_int channels; 3366 u_int ch; 3367 int srcused; 3368 3369 track = arg->context; 3370 KASSERT(track); 3371 src = &track->freq.srcbuf; 3372 dst = track->freq.dst; 3373 DIAGNOSTIC_ring(dst); 3374 DIAGNOSTIC_ring(src); 3375 KASSERT(src->used > 0); 3376 KASSERT(src->fmt.channels == dst->fmt.channels); 3377 KASSERT(src->head % track->mixer->frames_per_block == 0); 3378 3379 s = arg->src; 3380 d = arg->dst; 3381 3382 /* 3383 * In order to faciliate interpolation for each block, slide (delay) 3384 * input by one sample. As a result, strictly speaking, the output 3385 * phase is delayed by 1/dstfreq. However, I believe there is no 3386 * observable impact. 3387 * 3388 * Example) 3389 * srcfreq:dstfreq = 1:3 3390 * 3391 * A - - 3392 * | 3393 * | 3394 * | B - - 3395 * +-----+-----> input timeframe 3396 * 0 1 3397 * 3398 * 0 1 3399 * +-----+-----> input timeframe 3400 * | A 3401 * | x x 3402 * | x x 3403 * x (B) 3404 * +-+-+-+-+-+-> output timeframe 3405 * 0 1 2 3 4 5 3406 */ 3407 3408 /* Last samples in previous block */ 3409 channels = src->fmt.channels; 3410 for (ch = 0; ch < channels; ch++) { 3411 prev[ch] = track->freq_prev[ch]; 3412 curr[ch] = track->freq_curr[ch]; 3413 grad[ch] = curr[ch] - prev[ch]; 3414 } 3415 3416 step = track->freq_step; 3417 t = track->freq_current; 3418 //#define FREQ_DEBUG 3419 #if defined(FREQ_DEBUG) 3420 #define PRINTF(fmt...) printf(fmt) 3421 #else 3422 #define PRINTF(fmt...) do { } while (0) 3423 #endif 3424 srcused = src->used; 3425 PRINTF("upstart step=%d leap=%d", step, track->freq_leap); 3426 PRINTF(" srcused=%d arg->count=%u", src->used, arg->count); 3427 PRINTF(" prev=%d curr=%d grad=%d", prev[0], curr[0], grad[0]); 3428 PRINTF(" t=%d\n", t); 3429 3430 for (i = 0; i < arg->count; i++) { 3431 PRINTF("i=%d t=%5d", i, t); 3432 if (t >= 65536) { 3433 for (ch = 0; ch < channels; ch++) { 3434 prev[ch] = curr[ch]; 3435 curr[ch] = *s++; 3436 grad[ch] = curr[ch] - prev[ch]; 3437 } 3438 PRINTF(" prev=%d s[%d]=%d", 3439 prev[0], src->used - srcused, curr[0]); 3440 3441 /* Update */ 3442 t -= 65536; 3443 srcused--; 3444 if (srcused < 0) { 3445 PRINTF(" break\n"); 3446 break; 3447 } 3448 } 3449 3450 for (ch = 0; ch < channels; ch++) { 3451 *d++ = prev[ch] + (aint2_t)grad[ch] * t / 65536; 3452 #if defined(FREQ_DEBUG) 3453 if (ch == 0) 3454 printf(" t=%5d *d=%d", t, d[-1]); 3455 #endif 3456 } 3457 t += step; 3458 3459 PRINTF("\n"); 3460 } 3461 PRINTF("end prev=%d curr=%d\n", prev[0], curr[0]); 3462 3463 auring_take(src, src->used); 3464 auring_push(dst, i); 3465 3466 /* Adjust */ 3467 t += track->freq_leap; 3468 3469 track->freq_current = t; 3470 for (ch = 0; ch < channels; ch++) { 3471 track->freq_prev[ch] = prev[ch]; 3472 track->freq_curr[ch] = curr[ch]; 3473 } 3474 } 3475 3476 /* 3477 * This filter performs frequency conversion (down sampling). 3478 * It uses simple thinning. 3479 */ 3480 static void 3481 audio_track_freq_down(audio_filter_arg_t *arg) 3482 { 3483 audio_track_t *track; 3484 audio_ring_t *src; 3485 audio_ring_t *dst; 3486 const aint_t *s0; 3487 aint_t *d; 3488 u_int i; 3489 u_int t; 3490 u_int step; 3491 u_int ch; 3492 u_int channels; 3493 3494 track = arg->context; 3495 KASSERT(track); 3496 src = &track->freq.srcbuf; 3497 dst = track->freq.dst; 3498 3499 DIAGNOSTIC_ring(dst); 3500 DIAGNOSTIC_ring(src); 3501 KASSERT(src->used > 0); 3502 KASSERT(src->fmt.channels == dst->fmt.channels); 3503 KASSERTMSG(src->head % track->mixer->frames_per_block == 0, 3504 "src->head=%d fpb=%d", 3505 src->head, track->mixer->frames_per_block); 3506 3507 s0 = arg->src; 3508 d = arg->dst; 3509 t = track->freq_current; 3510 step = track->freq_step; 3511 channels = dst->fmt.channels; 3512 PRINTF("downstart step=%d leap=%d", step, track->freq_leap); 3513 PRINTF(" srcused=%d arg->count=%u", src->used, arg->count); 3514 PRINTF(" t=%d\n", t); 3515 3516 for (i = 0; i < arg->count && t / 65536 < src->used; i++) { 3517 const aint_t *s; 3518 PRINTF("i=%4d t=%10d", i, t); 3519 s = s0 + (t / 65536) * channels; 3520 PRINTF(" s=%5ld", (s - s0) / channels); 3521 for (ch = 0; ch < channels; ch++) { 3522 if (ch == 0) PRINTF(" *s=%d", s[ch]); 3523 *d++ = s[ch]; 3524 } 3525 PRINTF("\n"); 3526 t += step; 3527 } 3528 t += track->freq_leap; 3529 PRINTF("end t=%d\n", t); 3530 auring_take(src, src->used); 3531 auring_push(dst, i); 3532 track->freq_current = t % 65536; 3533 } 3534 3535 /* 3536 * Creates track and returns it. 3537 */ 3538 audio_track_t * 3539 audio_track_create(struct audio_softc *sc, audio_trackmixer_t *mixer) 3540 { 3541 audio_track_t *track; 3542 static int newid = 0; 3543 3544 track = kmem_zalloc(sizeof(*track), KM_SLEEP); 3545 3546 track->id = newid++; 3547 track->mixer = mixer; 3548 track->mode = mixer->mode; 3549 3550 /* Do TRACE after id is assigned. */ 3551 TRACET(3, track, "for %s", 3552 mixer->mode == AUMODE_PLAY ? "playback" : "recording"); 3553 3554 #if defined(AUDIO_SUPPORT_TRACK_VOLUME) 3555 track->volume = 256; 3556 #endif 3557 for (int i = 0; i < AUDIO_MAX_CHANNELS; i++) { 3558 track->ch_volume[i] = 256; 3559 } 3560 3561 return track; 3562 } 3563 3564 /* 3565 * Release all resources of the track and track itself. 3566 * track must not be NULL. Don't specify the track within the file 3567 * structure linked from sc->sc_files. 3568 */ 3569 static void 3570 audio_track_destroy(audio_track_t *track) 3571 { 3572 3573 KASSERT(track); 3574 3575 audio_free_usrbuf(track); 3576 audio_free(track->codec.srcbuf.mem); 3577 audio_free(track->chvol.srcbuf.mem); 3578 audio_free(track->chmix.srcbuf.mem); 3579 audio_free(track->freq.srcbuf.mem); 3580 audio_free(track->outbuf.mem); 3581 3582 kmem_free(track, sizeof(*track)); 3583 } 3584 3585 /* 3586 * It returns encoding conversion filter according to src and dst format. 3587 * If it is not a convertible pair, it returns NULL. Either src or dst 3588 * must be internal format. 3589 */ 3590 static audio_filter_t 3591 audio_track_get_codec(audio_track_t *track, const audio_format2_t *src, 3592 const audio_format2_t *dst) 3593 { 3594 3595 if (audio_format2_is_internal(src)) { 3596 if (dst->encoding == AUDIO_ENCODING_ULAW) { 3597 return audio_internal_to_mulaw; 3598 } else if (dst->encoding == AUDIO_ENCODING_ALAW) { 3599 return audio_internal_to_alaw; 3600 } else if (audio_format2_is_linear(dst)) { 3601 switch (dst->stride) { 3602 case 8: 3603 return audio_internal_to_linear8; 3604 case 16: 3605 return audio_internal_to_linear16; 3606 #if defined(AUDIO_SUPPORT_LINEAR24) 3607 case 24: 3608 return audio_internal_to_linear24; 3609 #endif 3610 case 32: 3611 return audio_internal_to_linear32; 3612 default: 3613 TRACET(1, track, "unsupported %s stride %d", 3614 "dst", dst->stride); 3615 goto abort; 3616 } 3617 } 3618 } else if (audio_format2_is_internal(dst)) { 3619 if (src->encoding == AUDIO_ENCODING_ULAW) { 3620 return audio_mulaw_to_internal; 3621 } else if (src->encoding == AUDIO_ENCODING_ALAW) { 3622 return audio_alaw_to_internal; 3623 } else if (audio_format2_is_linear(src)) { 3624 switch (src->stride) { 3625 case 8: 3626 return audio_linear8_to_internal; 3627 case 16: 3628 return audio_linear16_to_internal; 3629 #if defined(AUDIO_SUPPORT_LINEAR24) 3630 case 24: 3631 return audio_linear24_to_internal; 3632 #endif 3633 case 32: 3634 return audio_linear32_to_internal; 3635 default: 3636 TRACET(1, track, "unsupported %s stride %d", 3637 "src", src->stride); 3638 goto abort; 3639 } 3640 } 3641 } 3642 3643 TRACET(1, track, "unsupported encoding"); 3644 abort: 3645 #if defined(AUDIO_DEBUG) 3646 if (audiodebug >= 2) { 3647 char buf[100]; 3648 audio_format2_tostr(buf, sizeof(buf), src); 3649 TRACET(2, track, "src %s", buf); 3650 audio_format2_tostr(buf, sizeof(buf), dst); 3651 TRACET(2, track, "dst %s", buf); 3652 } 3653 #endif 3654 return NULL; 3655 } 3656 3657 /* 3658 * Initialize the codec stage of this track as necessary. 3659 * If successful, it initializes the codec stage as necessary, stores updated 3660 * last_dst in *last_dstp in any case, and returns 0. 3661 * Otherwise, it returns errno without modifying *last_dstp. 3662 */ 3663 static int 3664 audio_track_init_codec(audio_track_t *track, audio_ring_t **last_dstp) 3665 { 3666 audio_ring_t *last_dst; 3667 audio_ring_t *srcbuf; 3668 audio_format2_t *srcfmt; 3669 audio_format2_t *dstfmt; 3670 audio_filter_arg_t *arg; 3671 u_int len; 3672 int error; 3673 3674 KASSERT(track); 3675 3676 last_dst = *last_dstp; 3677 dstfmt = &last_dst->fmt; 3678 srcfmt = &track->inputfmt; 3679 srcbuf = &track->codec.srcbuf; 3680 error = 0; 3681 3682 if (srcfmt->encoding != dstfmt->encoding 3683 || srcfmt->precision != dstfmt->precision 3684 || srcfmt->stride != dstfmt->stride) { 3685 track->codec.dst = last_dst; 3686 3687 srcbuf->fmt = *dstfmt; 3688 srcbuf->fmt.encoding = srcfmt->encoding; 3689 srcbuf->fmt.precision = srcfmt->precision; 3690 srcbuf->fmt.stride = srcfmt->stride; 3691 3692 track->codec.filter = audio_track_get_codec(track, 3693 &srcbuf->fmt, dstfmt); 3694 if (track->codec.filter == NULL) { 3695 error = EINVAL; 3696 goto abort; 3697 } 3698 3699 srcbuf->head = 0; 3700 srcbuf->used = 0; 3701 srcbuf->capacity = frame_per_block(track->mixer, &srcbuf->fmt); 3702 len = auring_bytelen(srcbuf); 3703 srcbuf->mem = audio_realloc(srcbuf->mem, len); 3704 3705 arg = &track->codec.arg; 3706 arg->srcfmt = &srcbuf->fmt; 3707 arg->dstfmt = dstfmt; 3708 arg->context = NULL; 3709 3710 *last_dstp = srcbuf; 3711 return 0; 3712 } 3713 3714 abort: 3715 track->codec.filter = NULL; 3716 audio_free(srcbuf->mem); 3717 return error; 3718 } 3719 3720 /* 3721 * Initialize the chvol stage of this track as necessary. 3722 * If successful, it initializes the chvol stage as necessary, stores updated 3723 * last_dst in *last_dstp in any case, and returns 0. 3724 * Otherwise, it returns errno without modifying *last_dstp. 3725 */ 3726 static int 3727 audio_track_init_chvol(audio_track_t *track, audio_ring_t **last_dstp) 3728 { 3729 audio_ring_t *last_dst; 3730 audio_ring_t *srcbuf; 3731 audio_format2_t *srcfmt; 3732 audio_format2_t *dstfmt; 3733 audio_filter_arg_t *arg; 3734 u_int len; 3735 int error; 3736 3737 KASSERT(track); 3738 3739 last_dst = *last_dstp; 3740 dstfmt = &last_dst->fmt; 3741 srcfmt = &track->inputfmt; 3742 srcbuf = &track->chvol.srcbuf; 3743 error = 0; 3744 3745 /* Check whether channel volume conversion is necessary. */ 3746 bool use_chvol = false; 3747 for (int ch = 0; ch < srcfmt->channels; ch++) { 3748 if (track->ch_volume[ch] != 256) { 3749 use_chvol = true; 3750 break; 3751 } 3752 } 3753 3754 if (use_chvol == true) { 3755 track->chvol.dst = last_dst; 3756 track->chvol.filter = audio_track_chvol; 3757 3758 srcbuf->fmt = *dstfmt; 3759 /* no format conversion occurs */ 3760 3761 srcbuf->head = 0; 3762 srcbuf->used = 0; 3763 srcbuf->capacity = frame_per_block(track->mixer, &srcbuf->fmt); 3764 len = auring_bytelen(srcbuf); 3765 srcbuf->mem = audio_realloc(srcbuf->mem, len); 3766 3767 arg = &track->chvol.arg; 3768 arg->srcfmt = &srcbuf->fmt; 3769 arg->dstfmt = dstfmt; 3770 arg->context = track->ch_volume; 3771 3772 *last_dstp = srcbuf; 3773 return 0; 3774 } 3775 3776 track->chvol.filter = NULL; 3777 audio_free(srcbuf->mem); 3778 return error; 3779 } 3780 3781 /* 3782 * Initialize the chmix stage of this track as necessary. 3783 * If successful, it initializes the chmix stage as necessary, stores updated 3784 * last_dst in *last_dstp in any case, and returns 0. 3785 * Otherwise, it returns errno without modifying *last_dstp. 3786 */ 3787 static int 3788 audio_track_init_chmix(audio_track_t *track, audio_ring_t **last_dstp) 3789 { 3790 audio_ring_t *last_dst; 3791 audio_ring_t *srcbuf; 3792 audio_format2_t *srcfmt; 3793 audio_format2_t *dstfmt; 3794 audio_filter_arg_t *arg; 3795 u_int srcch; 3796 u_int dstch; 3797 u_int len; 3798 int error; 3799 3800 KASSERT(track); 3801 3802 last_dst = *last_dstp; 3803 dstfmt = &last_dst->fmt; 3804 srcfmt = &track->inputfmt; 3805 srcbuf = &track->chmix.srcbuf; 3806 error = 0; 3807 3808 srcch = srcfmt->channels; 3809 dstch = dstfmt->channels; 3810 if (srcch != dstch) { 3811 track->chmix.dst = last_dst; 3812 3813 if (srcch >= 2 && dstch == 1) { 3814 track->chmix.filter = audio_track_chmix_mixLR; 3815 } else if (srcch == 1 && dstch >= 2) { 3816 track->chmix.filter = audio_track_chmix_dupLR; 3817 } else if (srcch > dstch) { 3818 track->chmix.filter = audio_track_chmix_shrink; 3819 } else { 3820 track->chmix.filter = audio_track_chmix_expand; 3821 } 3822 3823 srcbuf->fmt = *dstfmt; 3824 srcbuf->fmt.channels = srcch; 3825 3826 srcbuf->head = 0; 3827 srcbuf->used = 0; 3828 /* XXX The buffer size should be able to calculate. */ 3829 srcbuf->capacity = frame_per_block(track->mixer, &srcbuf->fmt); 3830 len = auring_bytelen(srcbuf); 3831 srcbuf->mem = audio_realloc(srcbuf->mem, len); 3832 3833 arg = &track->chmix.arg; 3834 arg->srcfmt = &srcbuf->fmt; 3835 arg->dstfmt = dstfmt; 3836 arg->context = NULL; 3837 3838 *last_dstp = srcbuf; 3839 return 0; 3840 } 3841 3842 track->chmix.filter = NULL; 3843 audio_free(srcbuf->mem); 3844 return error; 3845 } 3846 3847 /* 3848 * Initialize the freq stage of this track as necessary. 3849 * If successful, it initializes the freq stage as necessary, stores updated 3850 * last_dst in *last_dstp in any case, and returns 0. 3851 * Otherwise, it returns errno without modifying *last_dstp. 3852 */ 3853 static int 3854 audio_track_init_freq(audio_track_t *track, audio_ring_t **last_dstp) 3855 { 3856 audio_ring_t *last_dst; 3857 audio_ring_t *srcbuf; 3858 audio_format2_t *srcfmt; 3859 audio_format2_t *dstfmt; 3860 audio_filter_arg_t *arg; 3861 uint32_t srcfreq; 3862 uint32_t dstfreq; 3863 u_int dst_capacity; 3864 u_int mod; 3865 u_int len; 3866 int error; 3867 3868 KASSERT(track); 3869 3870 last_dst = *last_dstp; 3871 dstfmt = &last_dst->fmt; 3872 srcfmt = &track->inputfmt; 3873 srcbuf = &track->freq.srcbuf; 3874 error = 0; 3875 3876 srcfreq = srcfmt->sample_rate; 3877 dstfreq = dstfmt->sample_rate; 3878 if (srcfreq != dstfreq) { 3879 track->freq.dst = last_dst; 3880 3881 memset(track->freq_prev, 0, sizeof(track->freq_prev)); 3882 memset(track->freq_curr, 0, sizeof(track->freq_curr)); 3883 3884 /* freq_step is the ratio of src/dst when let dst 65536. */ 3885 track->freq_step = (uint64_t)srcfreq * 65536 / dstfreq; 3886 3887 dst_capacity = frame_per_block(track->mixer, dstfmt); 3888 mod = (uint64_t)srcfreq * 65536 % dstfreq; 3889 track->freq_leap = (mod * dst_capacity + dstfreq / 2) / dstfreq; 3890 3891 if (track->freq_step < 65536) { 3892 track->freq.filter = audio_track_freq_up; 3893 /* In order to carry at the first time. */ 3894 track->freq_current = 65536; 3895 } else { 3896 track->freq.filter = audio_track_freq_down; 3897 track->freq_current = 0; 3898 } 3899 3900 srcbuf->fmt = *dstfmt; 3901 srcbuf->fmt.sample_rate = srcfreq; 3902 3903 srcbuf->head = 0; 3904 srcbuf->used = 0; 3905 srcbuf->capacity = frame_per_block(track->mixer, &srcbuf->fmt); 3906 len = auring_bytelen(srcbuf); 3907 srcbuf->mem = audio_realloc(srcbuf->mem, len); 3908 3909 arg = &track->freq.arg; 3910 arg->srcfmt = &srcbuf->fmt; 3911 arg->dstfmt = dstfmt;/*&last_dst->fmt;*/ 3912 arg->context = track; 3913 3914 *last_dstp = srcbuf; 3915 return 0; 3916 } 3917 3918 track->freq.filter = NULL; 3919 audio_free(srcbuf->mem); 3920 return error; 3921 } 3922 3923 /* 3924 * When playing back: (e.g. if codec and freq stage are valid) 3925 * 3926 * write 3927 * | uiomove 3928 * v 3929 * usrbuf [...............] byte ring buffer (mmap-able) 3930 * | memcpy 3931 * v 3932 * codec.srcbuf[....] 1 block (ring) buffer <-- stage input 3933 * .dst ----+ 3934 * | convert 3935 * v 3936 * freq.srcbuf [....] 1 block (ring) buffer 3937 * .dst ----+ 3938 * | convert 3939 * v 3940 * outbuf [...............] NBLKOUT blocks ring buffer 3941 * 3942 * 3943 * When recording: 3944 * 3945 * freq.srcbuf [...............] NBLKOUT blocks ring buffer <-- stage input 3946 * .dst ----+ 3947 * | convert 3948 * v 3949 * codec.srcbuf[.....] 1 block (ring) buffer 3950 * .dst ----+ 3951 * | convert 3952 * v 3953 * outbuf [.....] 1 block (ring) buffer 3954 * | memcpy 3955 * v 3956 * usrbuf [...............] byte ring buffer (mmap-able *) 3957 * | uiomove 3958 * v 3959 * read 3960 * 3961 * *: usrbuf for recording is also mmap-able due to symmetry with 3962 * playback buffer, but for now mmap will never happen for recording. 3963 */ 3964 3965 /* 3966 * Set the userland format of this track. 3967 * usrfmt argument should be parameter verified with audio_check_params(). 3968 * It will release and reallocate all internal conversion buffers. 3969 * It returns 0 if successful. Otherwise it returns errno with clearing all 3970 * internal buffers. 3971 * It must be called without sc_intr_lock since uvm_* routines require non 3972 * intr_lock state. 3973 * It must be called with track lock held since it may release and reallocate 3974 * outbuf. 3975 */ 3976 static int 3977 audio_track_set_format(audio_track_t *track, audio_format2_t *usrfmt) 3978 { 3979 struct audio_softc *sc; 3980 u_int newbufsize; 3981 u_int oldblksize; 3982 u_int len; 3983 int error; 3984 3985 KASSERT(track); 3986 sc = track->mixer->sc; 3987 3988 /* usrbuf is the closest buffer to the userland. */ 3989 track->usrbuf.fmt = *usrfmt; 3990 3991 /* 3992 * For references, one block size (in 40msec) is: 3993 * 320 bytes = 204 blocks/64KB for mulaw/8kHz/1ch 3994 * 7680 bytes = 8 blocks/64KB for s16/48kHz/2ch 3995 * 30720 bytes = 90 KB/3blocks for s16/48kHz/8ch 3996 * 61440 bytes = 180 KB/3blocks for s16/96kHz/8ch 3997 * 245760 bytes = 720 KB/3blocks for s32/192kHz/8ch 3998 * 3999 * For example, 4000 * 1) If usrbuf_blksize = 7056 (s16/44.1k/2ch) and PAGE_SIZE = 8192, 4001 * newbufsize = rounddown(65536 / 7056) = 63504 4002 * newvsize = roundup2(63504, PAGE_SIZE) = 65536 4003 * Therefore it maps 8 * 8K pages and usrbuf->capacity = 63504. 4004 * 4005 * 2) If usrbuf_blksize = 7680 (s16/48k/2ch) and PAGE_SIZE = 4096, 4006 * newbufsize = rounddown(65536 / 7680) = 61440 4007 * newvsize = roundup2(61440, PAGE_SIZE) = 61440 (= 15 pages) 4008 * Therefore it maps 15 * 4K pages and usrbuf->capacity = 61440. 4009 */ 4010 oldblksize = track->usrbuf_blksize; 4011 track->usrbuf_blksize = frametobyte(&track->usrbuf.fmt, 4012 frame_per_block(track->mixer, &track->usrbuf.fmt)); 4013 track->usrbuf.head = 0; 4014 track->usrbuf.used = 0; 4015 newbufsize = MAX(track->usrbuf_blksize * AUMINNOBLK, 65536); 4016 newbufsize = rounddown(newbufsize, track->usrbuf_blksize); 4017 error = audio_realloc_usrbuf(track, newbufsize); 4018 if (error) { 4019 device_printf(sc->sc_dev, "malloc usrbuf(%d) failed\n", 4020 newbufsize); 4021 goto error; 4022 } 4023 4024 /* Recalc water mark. */ 4025 if (track->usrbuf_blksize != oldblksize) { 4026 if (audio_track_is_playback(track)) { 4027 /* Set high at 100%, low at 75%. */ 4028 track->usrbuf_usedhigh = track->usrbuf.capacity; 4029 track->usrbuf_usedlow = track->usrbuf.capacity * 3 / 4; 4030 } else { 4031 /* Set high at 100% minus 1block(?), low at 0% */ 4032 track->usrbuf_usedhigh = track->usrbuf.capacity - 4033 track->usrbuf_blksize; 4034 track->usrbuf_usedlow = 0; 4035 } 4036 } 4037 4038 /* Stage buffer */ 4039 audio_ring_t *last_dst = &track->outbuf; 4040 if (audio_track_is_playback(track)) { 4041 /* On playback, initialize from the mixer side in order. */ 4042 track->inputfmt = *usrfmt; 4043 track->outbuf.fmt = track->mixer->track_fmt; 4044 4045 if ((error = audio_track_init_freq(track, &last_dst)) != 0) 4046 goto error; 4047 if ((error = audio_track_init_chmix(track, &last_dst)) != 0) 4048 goto error; 4049 if ((error = audio_track_init_chvol(track, &last_dst)) != 0) 4050 goto error; 4051 if ((error = audio_track_init_codec(track, &last_dst)) != 0) 4052 goto error; 4053 } else { 4054 /* On recording, initialize from userland side in order. */ 4055 track->inputfmt = track->mixer->track_fmt; 4056 track->outbuf.fmt = *usrfmt; 4057 4058 if ((error = audio_track_init_codec(track, &last_dst)) != 0) 4059 goto error; 4060 if ((error = audio_track_init_chvol(track, &last_dst)) != 0) 4061 goto error; 4062 if ((error = audio_track_init_chmix(track, &last_dst)) != 0) 4063 goto error; 4064 if ((error = audio_track_init_freq(track, &last_dst)) != 0) 4065 goto error; 4066 } 4067 #if 0 4068 /* debug */ 4069 if (track->freq.filter) { 4070 audio_print_format2("freq src", &track->freq.srcbuf.fmt); 4071 audio_print_format2("freq dst", &track->freq.dst->fmt); 4072 } 4073 if (track->chmix.filter) { 4074 audio_print_format2("chmix src", &track->chmix.srcbuf.fmt); 4075 audio_print_format2("chmix dst", &track->chmix.dst->fmt); 4076 } 4077 if (track->chvol.filter) { 4078 audio_print_format2("chvol src", &track->chvol.srcbuf.fmt); 4079 audio_print_format2("chvol dst", &track->chvol.dst->fmt); 4080 } 4081 if (track->codec.filter) { 4082 audio_print_format2("codec src", &track->codec.srcbuf.fmt); 4083 audio_print_format2("codec dst", &track->codec.dst->fmt); 4084 } 4085 #endif 4086 4087 /* Stage input buffer */ 4088 track->input = last_dst; 4089 4090 /* 4091 * On the recording track, make the first stage a ring buffer. 4092 * XXX is there a better way? 4093 */ 4094 if (audio_track_is_record(track)) { 4095 track->input->capacity = NBLKOUT * 4096 frame_per_block(track->mixer, &track->input->fmt); 4097 len = auring_bytelen(track->input); 4098 track->input->mem = audio_realloc(track->input->mem, len); 4099 } 4100 4101 /* 4102 * Output buffer. 4103 * On the playback track, its capacity is NBLKOUT blocks. 4104 * On the recording track, its capacity is 1 block. 4105 */ 4106 track->outbuf.head = 0; 4107 track->outbuf.used = 0; 4108 track->outbuf.capacity = frame_per_block(track->mixer, 4109 &track->outbuf.fmt); 4110 if (audio_track_is_playback(track)) 4111 track->outbuf.capacity *= NBLKOUT; 4112 len = auring_bytelen(&track->outbuf); 4113 track->outbuf.mem = audio_realloc(track->outbuf.mem, len); 4114 if (track->outbuf.mem == NULL) { 4115 device_printf(sc->sc_dev, "malloc outbuf(%d) failed\n", len); 4116 error = ENOMEM; 4117 goto error; 4118 } 4119 4120 #if defined(AUDIO_DEBUG) 4121 if (audiodebug >= 3) { 4122 struct audio_track_debugbuf m; 4123 4124 memset(&m, 0, sizeof(m)); 4125 snprintf(m.outbuf, sizeof(m.outbuf), " out=%d", 4126 track->outbuf.capacity * frametobyte(&track->outbuf.fmt,1)); 4127 if (track->freq.filter) 4128 snprintf(m.freq, sizeof(m.freq), " freq=%d", 4129 track->freq.srcbuf.capacity * 4130 frametobyte(&track->freq.srcbuf.fmt, 1)); 4131 if (track->chmix.filter) 4132 snprintf(m.chmix, sizeof(m.chmix), " chmix=%d", 4133 track->chmix.srcbuf.capacity * 4134 frametobyte(&track->chmix.srcbuf.fmt, 1)); 4135 if (track->chvol.filter) 4136 snprintf(m.chvol, sizeof(m.chvol), " chvol=%d", 4137 track->chvol.srcbuf.capacity * 4138 frametobyte(&track->chvol.srcbuf.fmt, 1)); 4139 if (track->codec.filter) 4140 snprintf(m.codec, sizeof(m.codec), " codec=%d", 4141 track->codec.srcbuf.capacity * 4142 frametobyte(&track->codec.srcbuf.fmt, 1)); 4143 snprintf(m.usrbuf, sizeof(m.usrbuf), 4144 " usr=%d", track->usrbuf.capacity); 4145 4146 if (audio_track_is_playback(track)) { 4147 TRACET(0, track, "bufsize%s%s%s%s%s%s", 4148 m.outbuf, m.freq, m.chmix, 4149 m.chvol, m.codec, m.usrbuf); 4150 } else { 4151 TRACET(0, track, "bufsize%s%s%s%s%s%s", 4152 m.freq, m.chmix, m.chvol, 4153 m.codec, m.outbuf, m.usrbuf); 4154 } 4155 } 4156 #endif 4157 return 0; 4158 4159 error: 4160 audio_free_usrbuf(track); 4161 audio_free(track->codec.srcbuf.mem); 4162 audio_free(track->chvol.srcbuf.mem); 4163 audio_free(track->chmix.srcbuf.mem); 4164 audio_free(track->freq.srcbuf.mem); 4165 audio_free(track->outbuf.mem); 4166 return error; 4167 } 4168 4169 /* 4170 * Fill silence frames (as the internal format) up to 1 block 4171 * if the ring is not empty and less than 1 block. 4172 * It returns the number of appended frames. 4173 */ 4174 static int 4175 audio_append_silence(audio_track_t *track, audio_ring_t *ring) 4176 { 4177 int fpb; 4178 int n; 4179 4180 KASSERT(track); 4181 KASSERT(audio_format2_is_internal(&ring->fmt)); 4182 4183 /* XXX is n correct? */ 4184 /* XXX memset uses frametobyte()? */ 4185 4186 if (ring->used == 0) 4187 return 0; 4188 4189 fpb = frame_per_block(track->mixer, &ring->fmt); 4190 if (ring->used >= fpb) 4191 return 0; 4192 4193 n = (ring->capacity - ring->used) % fpb; 4194 4195 KASSERT(auring_get_contig_free(ring) >= n); 4196 4197 memset(auring_tailptr_aint(ring), 0, 4198 n * ring->fmt.channels * sizeof(aint_t)); 4199 auring_push(ring, n); 4200 return n; 4201 } 4202 4203 /* 4204 * Execute the conversion stage. 4205 * It prepares arg from this stage and executes stage->filter. 4206 * It must be called only if stage->filter is not NULL. 4207 * 4208 * For stages other than frequency conversion, the function increments 4209 * src and dst counters here. For frequency conversion stage, on the 4210 * other hand, the function does not touch src and dst counters and 4211 * filter side has to increment them. 4212 */ 4213 static void 4214 audio_apply_stage(audio_track_t *track, audio_stage_t *stage, bool isfreq) 4215 { 4216 audio_filter_arg_t *arg; 4217 int srccount; 4218 int dstcount; 4219 int count; 4220 4221 KASSERT(track); 4222 KASSERT(stage->filter); 4223 4224 srccount = auring_get_contig_used(&stage->srcbuf); 4225 dstcount = auring_get_contig_free(stage->dst); 4226 4227 if (isfreq) { 4228 KASSERTMSG(srccount > 0, "freq but srccount == %d", srccount); 4229 count = uimin(dstcount, track->mixer->frames_per_block); 4230 } else { 4231 count = uimin(srccount, dstcount); 4232 } 4233 4234 if (count > 0) { 4235 arg = &stage->arg; 4236 arg->src = auring_headptr(&stage->srcbuf); 4237 arg->dst = auring_tailptr(stage->dst); 4238 arg->count = count; 4239 4240 stage->filter(arg); 4241 4242 if (!isfreq) { 4243 auring_take(&stage->srcbuf, count); 4244 auring_push(stage->dst, count); 4245 } 4246 } 4247 } 4248 4249 /* 4250 * Produce output buffer for playback from user input buffer. 4251 * It must be called only if usrbuf is not empty and outbuf is 4252 * available at least one free block. 4253 */ 4254 static void 4255 audio_track_play(audio_track_t *track) 4256 { 4257 audio_ring_t *usrbuf; 4258 audio_ring_t *input; 4259 int count; 4260 int framesize; 4261 int bytes; 4262 4263 KASSERT(track); 4264 KASSERT(track->lock); 4265 TRACET(4, track, "start pstate=%d", track->pstate); 4266 4267 /* At this point usrbuf must not be empty. */ 4268 KASSERT(track->usrbuf.used > 0); 4269 /* Also, outbuf must be available at least one block. */ 4270 count = auring_get_contig_free(&track->outbuf); 4271 KASSERTMSG(count >= frame_per_block(track->mixer, &track->outbuf.fmt), 4272 "count=%d fpb=%d", 4273 count, frame_per_block(track->mixer, &track->outbuf.fmt)); 4274 4275 /* XXX TODO: is this necessary for now? */ 4276 int track_count_0 = track->outbuf.used; 4277 4278 usrbuf = &track->usrbuf; 4279 input = track->input; 4280 4281 /* 4282 * framesize is always 1 byte or more since all formats supported as 4283 * usrfmt(=input) have 8bit or more stride. 4284 */ 4285 framesize = frametobyte(&input->fmt, 1); 4286 KASSERT(framesize >= 1); 4287 4288 /* The next stage of usrbuf (=input) must be available. */ 4289 KASSERT(auring_get_contig_free(input) > 0); 4290 4291 /* 4292 * Copy usrbuf up to 1block to input buffer. 4293 * count is the number of frames to copy from usrbuf. 4294 * bytes is the number of bytes to copy from usrbuf. However it is 4295 * not copied less than one frame. 4296 */ 4297 count = uimin(usrbuf->used, track->usrbuf_blksize) / framesize; 4298 bytes = count * framesize; 4299 4300 track->usrbuf_stamp += bytes; 4301 4302 if (usrbuf->head + bytes < usrbuf->capacity) { 4303 memcpy((uint8_t *)input->mem + auring_tail(input) * framesize, 4304 (uint8_t *)usrbuf->mem + usrbuf->head, 4305 bytes); 4306 auring_push(input, count); 4307 auring_take(usrbuf, bytes); 4308 } else { 4309 int bytes1; 4310 int bytes2; 4311 4312 bytes1 = auring_get_contig_used(usrbuf); 4313 KASSERT(bytes1 % framesize == 0); 4314 memcpy((uint8_t *)input->mem + auring_tail(input) * framesize, 4315 (uint8_t *)usrbuf->mem + usrbuf->head, 4316 bytes1); 4317 auring_push(input, bytes1 / framesize); 4318 auring_take(usrbuf, bytes1); 4319 4320 bytes2 = bytes - bytes1; 4321 memcpy((uint8_t *)input->mem + auring_tail(input) * framesize, 4322 (uint8_t *)usrbuf->mem + usrbuf->head, 4323 bytes2); 4324 auring_push(input, bytes2 / framesize); 4325 auring_take(usrbuf, bytes2); 4326 } 4327 4328 /* Encoding conversion */ 4329 if (track->codec.filter) 4330 audio_apply_stage(track, &track->codec, false); 4331 4332 /* Channel volume */ 4333 if (track->chvol.filter) 4334 audio_apply_stage(track, &track->chvol, false); 4335 4336 /* Channel mix */ 4337 if (track->chmix.filter) 4338 audio_apply_stage(track, &track->chmix, false); 4339 4340 /* Frequency conversion */ 4341 /* 4342 * Since the frequency conversion needs correction for each block, 4343 * it rounds up to 1 block. 4344 */ 4345 if (track->freq.filter) { 4346 int n; 4347 n = audio_append_silence(track, &track->freq.srcbuf); 4348 if (n > 0) { 4349 TRACET(4, track, 4350 "freq.srcbuf add silence %d -> %d/%d/%d", 4351 n, 4352 track->freq.srcbuf.head, 4353 track->freq.srcbuf.used, 4354 track->freq.srcbuf.capacity); 4355 } 4356 if (track->freq.srcbuf.used > 0) { 4357 audio_apply_stage(track, &track->freq, true); 4358 } 4359 } 4360 4361 if (bytes < track->usrbuf_blksize) { 4362 /* 4363 * Clear all conversion buffer pointer if the conversion was 4364 * not exactly one block. These conversion stage buffers are 4365 * certainly circular buffers because of symmetry with the 4366 * previous and next stage buffer. However, since they are 4367 * treated as simple contiguous buffers in operation, so head 4368 * always should point 0. This may happen during drain-age. 4369 */ 4370 TRACET(4, track, "reset stage"); 4371 if (track->codec.filter) { 4372 KASSERT(track->codec.srcbuf.used == 0); 4373 track->codec.srcbuf.head = 0; 4374 } 4375 if (track->chvol.filter) { 4376 KASSERT(track->chvol.srcbuf.used == 0); 4377 track->chvol.srcbuf.head = 0; 4378 } 4379 if (track->chmix.filter) { 4380 KASSERT(track->chmix.srcbuf.used == 0); 4381 track->chmix.srcbuf.head = 0; 4382 } 4383 if (track->freq.filter) { 4384 KASSERT(track->freq.srcbuf.used == 0); 4385 track->freq.srcbuf.head = 0; 4386 } 4387 } 4388 4389 if (track->input == &track->outbuf) { 4390 track->outputcounter = track->inputcounter; 4391 } else { 4392 track->outputcounter += track->outbuf.used - track_count_0; 4393 } 4394 4395 #if defined(AUDIO_DEBUG) 4396 if (audiodebug >= 3) { 4397 struct audio_track_debugbuf m; 4398 audio_track_bufstat(track, &m); 4399 TRACET(0, track, "end%s%s%s%s%s%s", 4400 m.outbuf, m.freq, m.chvol, m.chmix, m.codec, m.usrbuf); 4401 } 4402 #endif 4403 } 4404 4405 /* 4406 * Produce user output buffer for recording from input buffer. 4407 */ 4408 static void 4409 audio_track_record(audio_track_t *track) 4410 { 4411 audio_ring_t *outbuf; 4412 audio_ring_t *usrbuf; 4413 int count; 4414 int bytes; 4415 int framesize; 4416 4417 KASSERT(track); 4418 KASSERT(track->lock); 4419 4420 /* Number of frames to process */ 4421 count = auring_get_contig_used(track->input); 4422 count = uimin(count, track->mixer->frames_per_block); 4423 if (count == 0) { 4424 TRACET(4, track, "count == 0"); 4425 return; 4426 } 4427 4428 /* Frequency conversion */ 4429 if (track->freq.filter) { 4430 if (track->freq.srcbuf.used > 0) { 4431 audio_apply_stage(track, &track->freq, true); 4432 /* XXX should input of freq be from beginning of buf? */ 4433 } 4434 } 4435 4436 /* Channel mix */ 4437 if (track->chmix.filter) 4438 audio_apply_stage(track, &track->chmix, false); 4439 4440 /* Channel volume */ 4441 if (track->chvol.filter) 4442 audio_apply_stage(track, &track->chvol, false); 4443 4444 /* Encoding conversion */ 4445 if (track->codec.filter) 4446 audio_apply_stage(track, &track->codec, false); 4447 4448 /* Copy outbuf to usrbuf */ 4449 outbuf = &track->outbuf; 4450 usrbuf = &track->usrbuf; 4451 /* 4452 * framesize is always 1 byte or more since all formats supported 4453 * as usrfmt(=output) have 8bit or more stride. 4454 */ 4455 framesize = frametobyte(&outbuf->fmt, 1); 4456 KASSERT(framesize >= 1); 4457 /* 4458 * count is the number of frames to copy to usrbuf. 4459 * bytes is the number of bytes to copy to usrbuf. 4460 */ 4461 count = outbuf->used; 4462 count = uimin(count, 4463 (track->usrbuf_usedhigh - usrbuf->used) / framesize); 4464 bytes = count * framesize; 4465 if (auring_tail(usrbuf) + bytes < usrbuf->capacity) { 4466 memcpy((uint8_t *)usrbuf->mem + auring_tail(usrbuf), 4467 (uint8_t *)outbuf->mem + outbuf->head * framesize, 4468 bytes); 4469 auring_push(usrbuf, bytes); 4470 auring_take(outbuf, count); 4471 } else { 4472 int bytes1; 4473 int bytes2; 4474 4475 bytes1 = auring_get_contig_free(usrbuf); 4476 KASSERT(bytes1 % framesize == 0); 4477 memcpy((uint8_t *)usrbuf->mem + auring_tail(usrbuf), 4478 (uint8_t *)outbuf->mem + outbuf->head * framesize, 4479 bytes1); 4480 auring_push(usrbuf, bytes1); 4481 auring_take(outbuf, bytes1 / framesize); 4482 4483 bytes2 = bytes - bytes1; 4484 memcpy((uint8_t *)usrbuf->mem + auring_tail(usrbuf), 4485 (uint8_t *)outbuf->mem + outbuf->head * framesize, 4486 bytes2); 4487 auring_push(usrbuf, bytes2); 4488 auring_take(outbuf, bytes2 / framesize); 4489 } 4490 4491 /* XXX TODO: any counters here? */ 4492 4493 #if defined(AUDIO_DEBUG) 4494 if (audiodebug >= 3) { 4495 struct audio_track_debugbuf m; 4496 audio_track_bufstat(track, &m); 4497 TRACET(0, track, "end%s%s%s%s%s%s", 4498 m.freq, m.chvol, m.chmix, m.codec, m.outbuf, m.usrbuf); 4499 } 4500 #endif 4501 } 4502 4503 /* 4504 * Calcurate blktime [msec] from mixer(.hwbuf.fmt). 4505 * Must be called with sc_lock held. 4506 */ 4507 static u_int 4508 audio_mixer_calc_blktime(struct audio_softc *sc, audio_trackmixer_t *mixer) 4509 { 4510 audio_format2_t *fmt; 4511 u_int blktime; 4512 u_int frames_per_block; 4513 4514 KASSERT(mutex_owned(sc->sc_lock)); 4515 4516 fmt = &mixer->hwbuf.fmt; 4517 blktime = sc->sc_blk_ms; 4518 4519 /* 4520 * If stride is not multiples of 8, special treatment is necessary. 4521 * For now, it is only x68k's vs(4), 4 bit/sample ADPCM. 4522 */ 4523 if (fmt->stride == 4) { 4524 frames_per_block = fmt->sample_rate * blktime / 1000; 4525 if ((frames_per_block & 1) != 0) 4526 blktime *= 2; 4527 } 4528 #ifdef DIAGNOSTIC 4529 else if (fmt->stride % NBBY != 0) { 4530 panic("unsupported HW stride %d", fmt->stride); 4531 } 4532 #endif 4533 4534 return blktime; 4535 } 4536 4537 /* 4538 * Initialize the mixer corresponding to the mode. 4539 * Set AUMODE_PLAY to the 'mode' for playback or AUMODE_RECORD for recording. 4540 * sc->sc_[pr]mixer (corresponding to the 'mode') must be zero-filled. 4541 * This function returns 0 on successful. Otherwise returns errno. 4542 * Must be called with sc_lock held. 4543 */ 4544 static int 4545 audio_mixer_init(struct audio_softc *sc, int mode, 4546 const audio_format2_t *hwfmt, const audio_filter_reg_t *reg) 4547 { 4548 char codecbuf[64]; 4549 audio_trackmixer_t *mixer; 4550 void (*softint_handler)(void *); 4551 int len; 4552 int blksize; 4553 int capacity; 4554 size_t bufsize; 4555 int hwblks; 4556 int blkms; 4557 int error; 4558 4559 KASSERT(hwfmt != NULL); 4560 KASSERT(reg != NULL); 4561 KASSERT(mutex_owned(sc->sc_lock)); 4562 4563 error = 0; 4564 if (mode == AUMODE_PLAY) 4565 mixer = sc->sc_pmixer; 4566 else 4567 mixer = sc->sc_rmixer; 4568 4569 mixer->sc = sc; 4570 mixer->mode = mode; 4571 4572 mixer->hwbuf.fmt = *hwfmt; 4573 mixer->volume = 256; 4574 mixer->blktime_d = 1000; 4575 mixer->blktime_n = audio_mixer_calc_blktime(sc, mixer); 4576 sc->sc_blk_ms = mixer->blktime_n; 4577 hwblks = NBLKHW; 4578 4579 mixer->frames_per_block = frame_per_block(mixer, &mixer->hwbuf.fmt); 4580 blksize = frametobyte(&mixer->hwbuf.fmt, mixer->frames_per_block); 4581 if (sc->hw_if->round_blocksize) { 4582 int rounded; 4583 audio_params_t p = format2_to_params(&mixer->hwbuf.fmt); 4584 rounded = sc->hw_if->round_blocksize(sc->hw_hdl, blksize, 4585 mode, &p); 4586 TRACE(1, "round_blocksize %d -> %d", blksize, rounded); 4587 if (rounded != blksize) { 4588 if ((rounded * NBBY) % (mixer->hwbuf.fmt.stride * 4589 mixer->hwbuf.fmt.channels) != 0) { 4590 device_printf(sc->sc_dev, 4591 "blksize not configured %d -> %d\n", 4592 blksize, rounded); 4593 return EINVAL; 4594 } 4595 /* Recalculation */ 4596 blksize = rounded; 4597 mixer->frames_per_block = blksize * NBBY / 4598 (mixer->hwbuf.fmt.stride * 4599 mixer->hwbuf.fmt.channels); 4600 } 4601 } 4602 mixer->blktime_n = mixer->frames_per_block; 4603 mixer->blktime_d = mixer->hwbuf.fmt.sample_rate; 4604 4605 capacity = mixer->frames_per_block * hwblks; 4606 bufsize = frametobyte(&mixer->hwbuf.fmt, capacity); 4607 if (sc->hw_if->round_buffersize) { 4608 size_t rounded; 4609 rounded = sc->hw_if->round_buffersize(sc->hw_hdl, mode, 4610 bufsize); 4611 TRACE(1, "round_buffersize %zd -> %zd", bufsize, rounded); 4612 if (rounded < bufsize) { 4613 /* buffersize needs NBLKHW blocks at least. */ 4614 device_printf(sc->sc_dev, 4615 "buffersize too small: buffersize=%zd blksize=%d\n", 4616 rounded, blksize); 4617 return EINVAL; 4618 } 4619 if (rounded % blksize != 0) { 4620 /* buffersize/blksize constraint mismatch? */ 4621 device_printf(sc->sc_dev, 4622 "buffersize must be multiple of blksize: " 4623 "buffersize=%zu blksize=%d\n", 4624 rounded, blksize); 4625 return EINVAL; 4626 } 4627 if (rounded != bufsize) { 4628 /* Recalcuration */ 4629 bufsize = rounded; 4630 hwblks = bufsize / blksize; 4631 capacity = mixer->frames_per_block * hwblks; 4632 } 4633 } 4634 TRACE(1, "buffersize for %s = %zu", 4635 (mode == AUMODE_PLAY) ? "playback" : "recording", 4636 bufsize); 4637 mixer->hwbuf.capacity = capacity; 4638 4639 /* 4640 * XXX need to release sc_lock for compatibility? 4641 */ 4642 if (sc->hw_if->allocm) { 4643 mixer->hwbuf.mem = sc->hw_if->allocm(sc->hw_hdl, mode, bufsize); 4644 if (mixer->hwbuf.mem == NULL) { 4645 device_printf(sc->sc_dev, "%s: allocm(%zu) failed\n", 4646 __func__, bufsize); 4647 return ENOMEM; 4648 } 4649 } else { 4650 mixer->hwbuf.mem = kmem_alloc(bufsize, KM_SLEEP); 4651 } 4652 4653 /* From here, audio_mixer_destroy is necessary to exit. */ 4654 if (mode == AUMODE_PLAY) { 4655 cv_init(&mixer->outcv, "audiowr"); 4656 } else { 4657 cv_init(&mixer->outcv, "audiord"); 4658 } 4659 4660 if (mode == AUMODE_PLAY) { 4661 softint_handler = audio_softintr_wr; 4662 } else { 4663 softint_handler = audio_softintr_rd; 4664 } 4665 mixer->sih = softint_establish(SOFTINT_SERIAL | SOFTINT_MPSAFE, 4666 softint_handler, sc); 4667 if (mixer->sih == NULL) { 4668 device_printf(sc->sc_dev, "softint_establish failed\n"); 4669 goto abort; 4670 } 4671 4672 mixer->track_fmt.encoding = AUDIO_ENCODING_SLINEAR_NE; 4673 mixer->track_fmt.precision = AUDIO_INTERNAL_BITS; 4674 mixer->track_fmt.stride = AUDIO_INTERNAL_BITS; 4675 mixer->track_fmt.channels = mixer->hwbuf.fmt.channels; 4676 mixer->track_fmt.sample_rate = mixer->hwbuf.fmt.sample_rate; 4677 4678 if (mixer->hwbuf.fmt.encoding == AUDIO_ENCODING_SLINEAR_OE && 4679 mixer->hwbuf.fmt.precision == AUDIO_INTERNAL_BITS) { 4680 mixer->swap_endian = true; 4681 TRACE(1, "swap_endian"); 4682 } 4683 4684 if (mode == AUMODE_PLAY) { 4685 /* Mixing buffer */ 4686 mixer->mixfmt = mixer->track_fmt; 4687 mixer->mixfmt.precision *= 2; 4688 mixer->mixfmt.stride *= 2; 4689 /* XXX TODO: use some macros? */ 4690 len = mixer->frames_per_block * mixer->mixfmt.channels * 4691 mixer->mixfmt.stride / NBBY; 4692 mixer->mixsample = audio_realloc(mixer->mixsample, len); 4693 } else { 4694 /* No mixing buffer for recording */ 4695 } 4696 4697 if (reg->codec) { 4698 mixer->codec = reg->codec; 4699 mixer->codecarg.context = reg->context; 4700 if (mode == AUMODE_PLAY) { 4701 mixer->codecarg.srcfmt = &mixer->track_fmt; 4702 mixer->codecarg.dstfmt = &mixer->hwbuf.fmt; 4703 } else { 4704 mixer->codecarg.srcfmt = &mixer->hwbuf.fmt; 4705 mixer->codecarg.dstfmt = &mixer->track_fmt; 4706 } 4707 mixer->codecbuf.fmt = mixer->track_fmt; 4708 mixer->codecbuf.capacity = mixer->frames_per_block; 4709 len = auring_bytelen(&mixer->codecbuf); 4710 mixer->codecbuf.mem = audio_realloc(mixer->codecbuf.mem, len); 4711 if (mixer->codecbuf.mem == NULL) { 4712 device_printf(sc->sc_dev, 4713 "%s: malloc codecbuf(%d) failed\n", 4714 __func__, len); 4715 error = ENOMEM; 4716 goto abort; 4717 } 4718 } 4719 4720 /* Succeeded so display it. */ 4721 codecbuf[0] = '\0'; 4722 if (mixer->codec || mixer->swap_endian) { 4723 snprintf(codecbuf, sizeof(codecbuf), " %s %s:%d", 4724 (mode == AUMODE_PLAY) ? "->" : "<-", 4725 audio_encoding_name(mixer->hwbuf.fmt.encoding), 4726 mixer->hwbuf.fmt.precision); 4727 } 4728 blkms = mixer->blktime_n * 1000 / mixer->blktime_d; 4729 aprint_normal_dev(sc->sc_dev, "%s:%d%s %dch %dHz, blk %dms for %s\n", 4730 audio_encoding_name(mixer->track_fmt.encoding), 4731 mixer->track_fmt.precision, 4732 codecbuf, 4733 mixer->track_fmt.channels, 4734 mixer->track_fmt.sample_rate, 4735 blkms, 4736 (mode == AUMODE_PLAY) ? "playback" : "recording"); 4737 4738 return 0; 4739 4740 abort: 4741 audio_mixer_destroy(sc, mixer); 4742 return error; 4743 } 4744 4745 /* 4746 * Releases all resources of 'mixer'. 4747 * Note that it does not release the memory area of 'mixer' itself. 4748 * Must be called with sc_lock held. 4749 */ 4750 static void 4751 audio_mixer_destroy(struct audio_softc *sc, audio_trackmixer_t *mixer) 4752 { 4753 int bufsize; 4754 4755 KASSERT(mutex_owned(sc->sc_lock)); 4756 4757 bufsize = frametobyte(&mixer->hwbuf.fmt, mixer->hwbuf.capacity); 4758 4759 if (mixer->hwbuf.mem != NULL) { 4760 if (sc->hw_if->freem) { 4761 sc->hw_if->freem(sc->hw_hdl, mixer->hwbuf.mem, bufsize); 4762 } else { 4763 kmem_free(mixer->hwbuf.mem, bufsize); 4764 } 4765 mixer->hwbuf.mem = NULL; 4766 } 4767 4768 audio_free(mixer->codecbuf.mem); 4769 audio_free(mixer->mixsample); 4770 4771 cv_destroy(&mixer->outcv); 4772 4773 if (mixer->sih) { 4774 softint_disestablish(mixer->sih); 4775 mixer->sih = NULL; 4776 } 4777 } 4778 4779 /* 4780 * Starts playback mixer. 4781 * Must be called only if sc_pbusy is false. 4782 * Must be called with sc_lock held. 4783 * Must not be called from the interrupt context. 4784 */ 4785 static void 4786 audio_pmixer_start(struct audio_softc *sc, bool force) 4787 { 4788 audio_trackmixer_t *mixer; 4789 int minimum; 4790 4791 KASSERT(mutex_owned(sc->sc_lock)); 4792 KASSERT(sc->sc_pbusy == false); 4793 4794 mutex_enter(sc->sc_intr_lock); 4795 4796 mixer = sc->sc_pmixer; 4797 TRACE(2, "%smixseq=%d hwseq=%d hwbuf=%d/%d/%d%s", 4798 (audiodebug >= 3) ? "begin " : "", 4799 (int)mixer->mixseq, (int)mixer->hwseq, 4800 mixer->hwbuf.head, mixer->hwbuf.used, mixer->hwbuf.capacity, 4801 force ? " force" : ""); 4802 4803 /* Need two blocks to start normally. */ 4804 minimum = (force) ? 1 : 2; 4805 while (mixer->hwbuf.used < mixer->frames_per_block * minimum) { 4806 audio_pmixer_process(sc); 4807 } 4808 4809 /* Start output */ 4810 audio_pmixer_output(sc); 4811 sc->sc_pbusy = true; 4812 4813 TRACE(3, "end mixseq=%d hwseq=%d hwbuf=%d/%d/%d", 4814 (int)mixer->mixseq, (int)mixer->hwseq, 4815 mixer->hwbuf.head, mixer->hwbuf.used, mixer->hwbuf.capacity); 4816 4817 mutex_exit(sc->sc_intr_lock); 4818 } 4819 4820 /* 4821 * When playing back with MD filter: 4822 * 4823 * track track ... 4824 * v v 4825 * + mix (with aint2_t) 4826 * | master volume (with aint2_t) 4827 * v 4828 * mixsample [::::] wide-int 1 block (ring) buffer 4829 * | 4830 * | convert aint2_t -> aint_t 4831 * v 4832 * codecbuf [....] 1 block (ring) buffer 4833 * | 4834 * | convert to hw format 4835 * v 4836 * hwbuf [............] NBLKHW blocks ring buffer 4837 * 4838 * When playing back without MD filter: 4839 * 4840 * mixsample [::::] wide-int 1 block (ring) buffer 4841 * | 4842 * | convert aint2_t -> aint_t 4843 * | (with byte swap if necessary) 4844 * v 4845 * hwbuf [............] NBLKHW blocks ring buffer 4846 * 4847 * mixsample: slinear_NE, wide internal precision, HW ch, HW freq. 4848 * codecbuf: slinear_NE, internal precision, HW ch, HW freq. 4849 * hwbuf: HW encoding, HW precision, HW ch, HW freq. 4850 */ 4851 4852 /* 4853 * Performs track mixing and converts it to hwbuf. 4854 * Note that this function doesn't transfer hwbuf to hardware. 4855 * Must be called with sc_intr_lock held. 4856 */ 4857 static void 4858 audio_pmixer_process(struct audio_softc *sc) 4859 { 4860 audio_trackmixer_t *mixer; 4861 audio_file_t *f; 4862 int frame_count; 4863 int sample_count; 4864 int mixed; 4865 int i; 4866 aint2_t *m; 4867 aint_t *h; 4868 4869 mixer = sc->sc_pmixer; 4870 4871 frame_count = mixer->frames_per_block; 4872 KASSERT(auring_get_contig_free(&mixer->hwbuf) >= frame_count); 4873 sample_count = frame_count * mixer->mixfmt.channels; 4874 4875 mixer->mixseq++; 4876 4877 /* Mix all tracks */ 4878 mixed = 0; 4879 SLIST_FOREACH(f, &sc->sc_files, entry) { 4880 audio_track_t *track = f->ptrack; 4881 4882 if (track == NULL) 4883 continue; 4884 4885 if (track->is_pause) { 4886 TRACET(4, track, "skip; paused"); 4887 continue; 4888 } 4889 4890 /* Skip if the track is used by process context. */ 4891 if (audio_track_lock_tryenter(track) == false) { 4892 TRACET(4, track, "skip; in use"); 4893 continue; 4894 } 4895 4896 /* Emulate mmap'ped track */ 4897 if (track->mmapped) { 4898 auring_push(&track->usrbuf, track->usrbuf_blksize); 4899 TRACET(4, track, "mmap; usr=%d/%d/C%d", 4900 track->usrbuf.head, 4901 track->usrbuf.used, 4902 track->usrbuf.capacity); 4903 } 4904 4905 if (track->outbuf.used < mixer->frames_per_block && 4906 track->usrbuf.used > 0) { 4907 TRACET(4, track, "process"); 4908 audio_track_play(track); 4909 } 4910 4911 if (track->outbuf.used > 0) { 4912 mixed = audio_pmixer_mix_track(mixer, track, mixed); 4913 } else { 4914 TRACET(4, track, "skip; empty"); 4915 } 4916 4917 audio_track_lock_exit(track); 4918 } 4919 4920 if (mixed == 0) { 4921 /* Silence */ 4922 memset(mixer->mixsample, 0, 4923 frametobyte(&mixer->mixfmt, frame_count)); 4924 } else { 4925 if (mixed > 1) { 4926 /* If there are multiple tracks, do auto gain control */ 4927 audio_pmixer_agc(mixer, sample_count); 4928 } 4929 4930 /* Apply master volume */ 4931 if (mixer->volume < 256) { 4932 m = mixer->mixsample; 4933 for (i = 0; i < sample_count; i++) { 4934 *m = AUDIO_SCALEDOWN(*m * mixer->volume, 8); 4935 m++; 4936 } 4937 4938 /* 4939 * Recover the volume gradually at the pace of 4940 * several times per second. If it's too fast, you 4941 * can recognize that the volume changes up and down 4942 * quickly and it's not so comfortable. 4943 */ 4944 mixer->voltimer += mixer->blktime_n; 4945 if (mixer->voltimer * 4 >= mixer->blktime_d) { 4946 mixer->volume++; 4947 mixer->voltimer = 0; 4948 #if defined(AUDIO_DEBUG_AGC) 4949 TRACE(1, "volume recover: %d", mixer->volume); 4950 #endif 4951 } 4952 } 4953 } 4954 4955 /* 4956 * The rest is the hardware part. 4957 */ 4958 4959 if (mixer->codec) { 4960 h = auring_tailptr_aint(&mixer->codecbuf); 4961 } else { 4962 h = auring_tailptr_aint(&mixer->hwbuf); 4963 } 4964 4965 m = mixer->mixsample; 4966 if (mixer->swap_endian) { 4967 for (i = 0; i < sample_count; i++) { 4968 *h++ = bswap16(*m++); 4969 } 4970 } else { 4971 for (i = 0; i < sample_count; i++) { 4972 *h++ = *m++; 4973 } 4974 } 4975 4976 /* Hardware driver's codec */ 4977 if (mixer->codec) { 4978 auring_push(&mixer->codecbuf, frame_count); 4979 mixer->codecarg.src = auring_headptr(&mixer->codecbuf); 4980 mixer->codecarg.dst = auring_tailptr(&mixer->hwbuf); 4981 mixer->codecarg.count = frame_count; 4982 mixer->codec(&mixer->codecarg); 4983 auring_take(&mixer->codecbuf, mixer->codecarg.count); 4984 } 4985 4986 auring_push(&mixer->hwbuf, frame_count); 4987 4988 TRACE(4, "done mixseq=%d hwbuf=%d/%d/%d%s", 4989 (int)mixer->mixseq, 4990 mixer->hwbuf.head, mixer->hwbuf.used, mixer->hwbuf.capacity, 4991 (mixed == 0) ? " silent" : ""); 4992 } 4993 4994 /* 4995 * Do auto gain control. 4996 * Must be called sc_intr_lock held. 4997 */ 4998 static void 4999 audio_pmixer_agc(audio_trackmixer_t *mixer, int sample_count) 5000 { 5001 struct audio_softc *sc __unused; 5002 aint2_t val; 5003 aint2_t maxval; 5004 aint2_t minval; 5005 aint2_t over_plus; 5006 aint2_t over_minus; 5007 aint2_t *m; 5008 int newvol; 5009 int i; 5010 5011 sc = mixer->sc; 5012 5013 /* Overflow detection */ 5014 maxval = AINT_T_MAX; 5015 minval = AINT_T_MIN; 5016 m = mixer->mixsample; 5017 for (i = 0; i < sample_count; i++) { 5018 val = *m++; 5019 if (val > maxval) 5020 maxval = val; 5021 else if (val < minval) 5022 minval = val; 5023 } 5024 5025 /* Absolute value of overflowed amount */ 5026 over_plus = maxval - AINT_T_MAX; 5027 over_minus = AINT_T_MIN - minval; 5028 5029 if (over_plus > 0 || over_minus > 0) { 5030 if (over_plus > over_minus) { 5031 newvol = (int)((aint2_t)AINT_T_MAX * 256 / maxval); 5032 } else { 5033 newvol = (int)((aint2_t)AINT_T_MIN * 256 / minval); 5034 } 5035 5036 /* 5037 * Change the volume only if new one is smaller. 5038 * Reset the timer even if the volume isn't changed. 5039 */ 5040 if (newvol <= mixer->volume) { 5041 mixer->volume = newvol; 5042 mixer->voltimer = 0; 5043 #if defined(AUDIO_DEBUG_AGC) 5044 TRACE(1, "auto volume adjust: %d", mixer->volume); 5045 #endif 5046 } 5047 } 5048 } 5049 5050 /* 5051 * Mix one track. 5052 * 'mixed' specifies the number of tracks mixed so far. 5053 * It returns the number of tracks mixed. In other words, it returns 5054 * mixed + 1 if this track is mixed. 5055 */ 5056 static int 5057 audio_pmixer_mix_track(audio_trackmixer_t *mixer, audio_track_t *track, 5058 int mixed) 5059 { 5060 int count; 5061 int sample_count; 5062 int remain; 5063 int i; 5064 const aint_t *s; 5065 aint2_t *d; 5066 5067 /* XXX TODO: Is this necessary for now? */ 5068 if (mixer->mixseq < track->seq) 5069 return mixed; 5070 5071 count = auring_get_contig_used(&track->outbuf); 5072 count = uimin(count, mixer->frames_per_block); 5073 5074 s = auring_headptr_aint(&track->outbuf); 5075 d = mixer->mixsample; 5076 5077 /* 5078 * Apply track volume with double-sized integer and perform 5079 * additive synthesis. 5080 * 5081 * XXX If you limit the track volume to 1.0 or less (<= 256), 5082 * it would be better to do this in the track conversion stage 5083 * rather than here. However, if you accept the volume to 5084 * be greater than 1.0 (> 256), it's better to do it here. 5085 * Because the operation here is done by double-sized integer. 5086 */ 5087 sample_count = count * mixer->mixfmt.channels; 5088 if (mixed == 0) { 5089 /* If this is the first track, assignment can be used. */ 5090 #if defined(AUDIO_SUPPORT_TRACK_VOLUME) 5091 if (track->volume != 256) { 5092 for (i = 0; i < sample_count; i++) { 5093 aint2_t v; 5094 v = *s++; 5095 *d++ = AUDIO_SCALEDOWN(v * track->volume, 8) 5096 } 5097 } else 5098 #endif 5099 { 5100 for (i = 0; i < sample_count; i++) { 5101 *d++ = ((aint2_t)*s++); 5102 } 5103 } 5104 /* Fill silence if the first track is not filled. */ 5105 for (; i < mixer->frames_per_block * mixer->mixfmt.channels; i++) 5106 *d++ = 0; 5107 } else { 5108 /* If this is the second or later, add it. */ 5109 #if defined(AUDIO_SUPPORT_TRACK_VOLUME) 5110 if (track->volume != 256) { 5111 for (i = 0; i < sample_count; i++) { 5112 aint2_t v; 5113 v = *s++; 5114 *d++ += AUDIO_SCALEDOWN(v * track->volume, 8); 5115 } 5116 } else 5117 #endif 5118 { 5119 for (i = 0; i < sample_count; i++) { 5120 *d++ += ((aint2_t)*s++); 5121 } 5122 } 5123 } 5124 5125 auring_take(&track->outbuf, count); 5126 /* 5127 * The counters have to align block even if outbuf is less than 5128 * one block. XXX Is this still necessary? 5129 */ 5130 remain = mixer->frames_per_block - count; 5131 if (__predict_false(remain != 0)) { 5132 auring_push(&track->outbuf, remain); 5133 auring_take(&track->outbuf, remain); 5134 } 5135 5136 /* 5137 * Update track sequence. 5138 * mixseq has previous value yet at this point. 5139 */ 5140 track->seq = mixer->mixseq + 1; 5141 5142 return mixed + 1; 5143 } 5144 5145 /* 5146 * Output one block from hwbuf to HW. 5147 * Must be called with sc_intr_lock held. 5148 */ 5149 static void 5150 audio_pmixer_output(struct audio_softc *sc) 5151 { 5152 audio_trackmixer_t *mixer; 5153 audio_params_t params; 5154 void *start; 5155 void *end; 5156 int blksize; 5157 int error; 5158 5159 mixer = sc->sc_pmixer; 5160 TRACE(4, "pbusy=%d hwbuf=%d/%d/%d", 5161 sc->sc_pbusy, 5162 mixer->hwbuf.head, mixer->hwbuf.used, mixer->hwbuf.capacity); 5163 KASSERT(mixer->hwbuf.used >= mixer->frames_per_block); 5164 5165 blksize = frametobyte(&mixer->hwbuf.fmt, mixer->frames_per_block); 5166 5167 if (sc->hw_if->trigger_output) { 5168 /* trigger (at once) */ 5169 if (!sc->sc_pbusy) { 5170 start = mixer->hwbuf.mem; 5171 end = (uint8_t *)start + auring_bytelen(&mixer->hwbuf); 5172 params = format2_to_params(&mixer->hwbuf.fmt); 5173 5174 error = sc->hw_if->trigger_output(sc->hw_hdl, 5175 start, end, blksize, audio_pintr, sc, ¶ms); 5176 if (error) { 5177 device_printf(sc->sc_dev, 5178 "trigger_output failed with %d\n", error); 5179 return; 5180 } 5181 } 5182 } else { 5183 /* start (everytime) */ 5184 start = auring_headptr(&mixer->hwbuf); 5185 5186 error = sc->hw_if->start_output(sc->hw_hdl, 5187 start, blksize, audio_pintr, sc); 5188 if (error) { 5189 device_printf(sc->sc_dev, 5190 "start_output failed with %d\n", error); 5191 return; 5192 } 5193 } 5194 } 5195 5196 /* 5197 * This is an interrupt handler for playback. 5198 * It is called with sc_intr_lock held. 5199 * 5200 * It is usually called from hardware interrupt. However, note that 5201 * for some drivers (e.g. uaudio) it is called from software interrupt. 5202 */ 5203 static void 5204 audio_pintr(void *arg) 5205 { 5206 struct audio_softc *sc; 5207 audio_trackmixer_t *mixer; 5208 5209 sc = arg; 5210 KASSERT(mutex_owned(sc->sc_intr_lock)); 5211 5212 if (sc->sc_dying) 5213 return; 5214 #if defined(DIAGNOSTIC) 5215 if (sc->sc_pbusy == false) { 5216 device_printf(sc->sc_dev, "stray interrupt\n"); 5217 return; 5218 } 5219 #endif 5220 5221 mixer = sc->sc_pmixer; 5222 mixer->hw_complete_counter += mixer->frames_per_block; 5223 mixer->hwseq++; 5224 5225 auring_take(&mixer->hwbuf, mixer->frames_per_block); 5226 5227 TRACE(4, 5228 "HW_INT ++hwseq=%" PRIu64 " cmplcnt=%" PRIu64 " hwbuf=%d/%d/%d", 5229 mixer->hwseq, mixer->hw_complete_counter, 5230 mixer->hwbuf.head, mixer->hwbuf.used, mixer->hwbuf.capacity); 5231 5232 #if defined(AUDIO_HW_SINGLE_BUFFER) 5233 /* 5234 * Create a new block here and output it immediately. 5235 * It makes a latency lower but needs machine power. 5236 */ 5237 audio_pmixer_process(sc); 5238 audio_pmixer_output(sc); 5239 #else 5240 /* 5241 * It is called when block N output is done. 5242 * Output immediately block N+1 created by the last interrupt. 5243 * And then create block N+2 for the next interrupt. 5244 * This method makes playback robust even on slower machines. 5245 * Instead the latency is increased by one block. 5246 */ 5247 5248 /* At first, output ready block. */ 5249 if (mixer->hwbuf.used >= mixer->frames_per_block) { 5250 audio_pmixer_output(sc); 5251 } 5252 5253 bool later = false; 5254 5255 if (mixer->hwbuf.used < mixer->frames_per_block) { 5256 later = true; 5257 } 5258 5259 /* Then, process next block. */ 5260 audio_pmixer_process(sc); 5261 5262 if (later) { 5263 audio_pmixer_output(sc); 5264 } 5265 #endif 5266 5267 /* 5268 * When this interrupt is the real hardware interrupt, disabling 5269 * preemption here is not necessary. But some drivers (e.g. uaudio) 5270 * emulate it by software interrupt, so kpreempt_disable is necessary. 5271 */ 5272 kpreempt_disable(); 5273 softint_schedule(mixer->sih); 5274 kpreempt_enable(); 5275 } 5276 5277 /* 5278 * Starts record mixer. 5279 * Must be called only if sc_rbusy is false. 5280 * Must be called with sc_lock held. 5281 * Must not be called from the interrupt context. 5282 */ 5283 static void 5284 audio_rmixer_start(struct audio_softc *sc) 5285 { 5286 5287 KASSERT(mutex_owned(sc->sc_lock)); 5288 KASSERT(sc->sc_rbusy == false); 5289 5290 mutex_enter(sc->sc_intr_lock); 5291 5292 TRACE(2, "%s", (audiodebug >= 3) ? "begin" : ""); 5293 audio_rmixer_input(sc); 5294 sc->sc_rbusy = true; 5295 TRACE(3, "end"); 5296 5297 mutex_exit(sc->sc_intr_lock); 5298 } 5299 5300 /* 5301 * When recording with MD filter: 5302 * 5303 * hwbuf [............] NBLKHW blocks ring buffer 5304 * | 5305 * | convert from hw format 5306 * v 5307 * codecbuf [....] 1 block (ring) buffer 5308 * | | 5309 * v v 5310 * track track ... 5311 * 5312 * When recording without MD filter: 5313 * 5314 * hwbuf [............] NBLKHW blocks ring buffer 5315 * | | 5316 * v v 5317 * track track ... 5318 * 5319 * hwbuf: HW encoding, HW precision, HW ch, HW freq. 5320 * codecbuf: slinear_NE, internal precision, HW ch, HW freq. 5321 */ 5322 5323 /* 5324 * Distribute a recorded block to all recording tracks. 5325 */ 5326 static void 5327 audio_rmixer_process(struct audio_softc *sc) 5328 { 5329 audio_trackmixer_t *mixer; 5330 audio_ring_t *mixersrc; 5331 audio_file_t *f; 5332 aint_t *p; 5333 int count; 5334 int bytes; 5335 int i; 5336 5337 mixer = sc->sc_rmixer; 5338 5339 /* 5340 * count is the number of frames to be retrieved this time. 5341 * count should be one block. 5342 */ 5343 count = auring_get_contig_used(&mixer->hwbuf); 5344 count = uimin(count, mixer->frames_per_block); 5345 if (count <= 0) { 5346 TRACE(4, "count %d: too short", count); 5347 return; 5348 } 5349 bytes = frametobyte(&mixer->track_fmt, count); 5350 5351 /* Hardware driver's codec */ 5352 if (mixer->codec) { 5353 mixer->codecarg.src = auring_headptr(&mixer->hwbuf); 5354 mixer->codecarg.dst = auring_tailptr(&mixer->codecbuf); 5355 mixer->codecarg.count = count; 5356 mixer->codec(&mixer->codecarg); 5357 auring_take(&mixer->hwbuf, mixer->codecarg.count); 5358 auring_push(&mixer->codecbuf, mixer->codecarg.count); 5359 mixersrc = &mixer->codecbuf; 5360 } else { 5361 mixersrc = &mixer->hwbuf; 5362 } 5363 5364 if (mixer->swap_endian) { 5365 /* inplace conversion */ 5366 p = auring_headptr_aint(mixersrc); 5367 for (i = 0; i < count * mixer->track_fmt.channels; i++, p++) { 5368 *p = bswap16(*p); 5369 } 5370 } 5371 5372 /* Distribute to all tracks. */ 5373 SLIST_FOREACH(f, &sc->sc_files, entry) { 5374 audio_track_t *track = f->rtrack; 5375 audio_ring_t *input; 5376 5377 if (track == NULL) 5378 continue; 5379 5380 if (track->is_pause) { 5381 TRACET(4, track, "skip; paused"); 5382 continue; 5383 } 5384 5385 if (audio_track_lock_tryenter(track) == false) { 5386 TRACET(4, track, "skip; in use"); 5387 continue; 5388 } 5389 5390 /* If the track buffer is full, discard the oldest one? */ 5391 input = track->input; 5392 if (input->capacity - input->used < mixer->frames_per_block) { 5393 int drops = mixer->frames_per_block - 5394 (input->capacity - input->used); 5395 track->dropframes += drops; 5396 TRACET(4, track, "drop %d frames: inp=%d/%d/%d", 5397 drops, 5398 input->head, input->used, input->capacity); 5399 auring_take(input, drops); 5400 } 5401 KASSERT(input->used % mixer->frames_per_block == 0); 5402 5403 memcpy(auring_tailptr_aint(input), 5404 auring_headptr_aint(mixersrc), 5405 bytes); 5406 auring_push(input, count); 5407 5408 /* XXX sequence counter? */ 5409 5410 audio_track_lock_exit(track); 5411 } 5412 5413 auring_take(mixersrc, count); 5414 } 5415 5416 /* 5417 * Input one block from HW to hwbuf. 5418 * Must be called with sc_intr_lock held. 5419 */ 5420 static void 5421 audio_rmixer_input(struct audio_softc *sc) 5422 { 5423 audio_trackmixer_t *mixer; 5424 audio_params_t params; 5425 void *start; 5426 void *end; 5427 int blksize; 5428 int error; 5429 5430 mixer = sc->sc_rmixer; 5431 blksize = frametobyte(&mixer->hwbuf.fmt, mixer->frames_per_block); 5432 5433 if (sc->hw_if->trigger_input) { 5434 /* trigger (at once) */ 5435 if (!sc->sc_rbusy) { 5436 start = mixer->hwbuf.mem; 5437 end = (uint8_t *)start + auring_bytelen(&mixer->hwbuf); 5438 params = format2_to_params(&mixer->hwbuf.fmt); 5439 5440 error = sc->hw_if->trigger_input(sc->hw_hdl, 5441 start, end, blksize, audio_rintr, sc, ¶ms); 5442 if (error) { 5443 device_printf(sc->sc_dev, 5444 "trigger_input failed with %d\n", error); 5445 return; 5446 } 5447 } 5448 } else { 5449 /* start (everytime) */ 5450 start = auring_tailptr(&mixer->hwbuf); 5451 5452 error = sc->hw_if->start_input(sc->hw_hdl, 5453 start, blksize, audio_rintr, sc); 5454 if (error) { 5455 device_printf(sc->sc_dev, 5456 "start_input failed with %d\n", error); 5457 return; 5458 } 5459 } 5460 } 5461 5462 /* 5463 * This is an interrupt handler for recording. 5464 * It is called with sc_intr_lock. 5465 * 5466 * It is usually called from hardware interrupt. However, note that 5467 * for some drivers (e.g. uaudio) it is called from software interrupt. 5468 */ 5469 static void 5470 audio_rintr(void *arg) 5471 { 5472 struct audio_softc *sc; 5473 audio_trackmixer_t *mixer; 5474 5475 sc = arg; 5476 KASSERT(mutex_owned(sc->sc_intr_lock)); 5477 5478 if (sc->sc_dying) 5479 return; 5480 #if defined(DIAGNOSTIC) 5481 if (sc->sc_rbusy == false) { 5482 device_printf(sc->sc_dev, "stray interrupt\n"); 5483 return; 5484 } 5485 #endif 5486 5487 mixer = sc->sc_rmixer; 5488 mixer->hw_complete_counter += mixer->frames_per_block; 5489 mixer->hwseq++; 5490 5491 auring_push(&mixer->hwbuf, mixer->frames_per_block); 5492 5493 TRACE(4, 5494 "HW_INT ++hwseq=%" PRIu64 " cmplcnt=%" PRIu64 " hwbuf=%d/%d/%d", 5495 mixer->hwseq, mixer->hw_complete_counter, 5496 mixer->hwbuf.head, mixer->hwbuf.used, mixer->hwbuf.capacity); 5497 5498 /* Distrubute recorded block */ 5499 audio_rmixer_process(sc); 5500 5501 /* Request next block */ 5502 audio_rmixer_input(sc); 5503 5504 /* 5505 * When this interrupt is the real hardware interrupt, disabling 5506 * preemption here is not necessary. But some drivers (e.g. uaudio) 5507 * emulate it by software interrupt, so kpreempt_disable is necessary. 5508 */ 5509 kpreempt_disable(); 5510 softint_schedule(mixer->sih); 5511 kpreempt_enable(); 5512 } 5513 5514 /* 5515 * Halts playback mixer. 5516 * This function also clears related parameters, so call this function 5517 * instead of calling halt_output directly. 5518 * Must be called only if sc_pbusy is true. 5519 * Must be called with sc_lock && sc_exlock held. 5520 */ 5521 static int 5522 audio_pmixer_halt(struct audio_softc *sc) 5523 { 5524 int error; 5525 5526 TRACE(2, ""); 5527 KASSERT(mutex_owned(sc->sc_lock)); 5528 KASSERT(sc->sc_exlock); 5529 5530 mutex_enter(sc->sc_intr_lock); 5531 error = sc->hw_if->halt_output(sc->hw_hdl); 5532 mutex_exit(sc->sc_intr_lock); 5533 5534 /* Halts anyway even if some error has occurred. */ 5535 sc->sc_pbusy = false; 5536 sc->sc_pmixer->hwbuf.head = 0; 5537 sc->sc_pmixer->hwbuf.used = 0; 5538 sc->sc_pmixer->mixseq = 0; 5539 sc->sc_pmixer->hwseq = 0; 5540 5541 return error; 5542 } 5543 5544 /* 5545 * Halts recording mixer. 5546 * This function also clears related parameters, so call this function 5547 * instead of calling halt_input directly. 5548 * Must be called only if sc_rbusy is true. 5549 * Must be called with sc_lock && sc_exlock held. 5550 */ 5551 static int 5552 audio_rmixer_halt(struct audio_softc *sc) 5553 { 5554 int error; 5555 5556 TRACE(2, ""); 5557 KASSERT(mutex_owned(sc->sc_lock)); 5558 KASSERT(sc->sc_exlock); 5559 5560 mutex_enter(sc->sc_intr_lock); 5561 error = sc->hw_if->halt_input(sc->hw_hdl); 5562 mutex_exit(sc->sc_intr_lock); 5563 5564 /* Halts anyway even if some error has occurred. */ 5565 sc->sc_rbusy = false; 5566 sc->sc_rmixer->hwbuf.head = 0; 5567 sc->sc_rmixer->hwbuf.used = 0; 5568 sc->sc_rmixer->mixseq = 0; 5569 sc->sc_rmixer->hwseq = 0; 5570 5571 return error; 5572 } 5573 5574 /* 5575 * Flush this track. 5576 * Halts all operations, clears all buffers, reset error counters. 5577 * XXX I'm not sure... 5578 */ 5579 static void 5580 audio_track_clear(struct audio_softc *sc, audio_track_t *track) 5581 { 5582 5583 KASSERT(track); 5584 TRACET(3, track, "clear"); 5585 5586 audio_track_lock_enter(track); 5587 5588 track->usrbuf.used = 0; 5589 /* Clear all internal parameters. */ 5590 if (track->codec.filter) { 5591 track->codec.srcbuf.used = 0; 5592 track->codec.srcbuf.head = 0; 5593 } 5594 if (track->chvol.filter) { 5595 track->chvol.srcbuf.used = 0; 5596 track->chvol.srcbuf.head = 0; 5597 } 5598 if (track->chmix.filter) { 5599 track->chmix.srcbuf.used = 0; 5600 track->chmix.srcbuf.head = 0; 5601 } 5602 if (track->freq.filter) { 5603 track->freq.srcbuf.used = 0; 5604 track->freq.srcbuf.head = 0; 5605 if (track->freq_step < 65536) 5606 track->freq_current = 65536; 5607 else 5608 track->freq_current = 0; 5609 memset(track->freq_prev, 0, sizeof(track->freq_prev)); 5610 memset(track->freq_curr, 0, sizeof(track->freq_curr)); 5611 } 5612 /* Clear buffer, then operation halts naturally. */ 5613 track->outbuf.used = 0; 5614 5615 /* Clear counters. */ 5616 track->dropframes = 0; 5617 5618 audio_track_lock_exit(track); 5619 } 5620 5621 /* 5622 * Drain the track. 5623 * track must be present and for playback. 5624 * If successful, it returns 0. Otherwise returns errno. 5625 * Must be called with sc_lock held. 5626 */ 5627 static int 5628 audio_track_drain(struct audio_softc *sc, audio_track_t *track) 5629 { 5630 audio_trackmixer_t *mixer; 5631 int done; 5632 int error; 5633 5634 KASSERT(track); 5635 TRACET(3, track, "start"); 5636 mixer = track->mixer; 5637 KASSERT(mutex_owned(sc->sc_lock)); 5638 5639 /* Ignore them if pause. */ 5640 if (track->is_pause) { 5641 TRACET(3, track, "pause -> clear"); 5642 track->pstate = AUDIO_STATE_CLEAR; 5643 } 5644 /* Terminate early here if there is no data in the track. */ 5645 if (track->pstate == AUDIO_STATE_CLEAR) { 5646 TRACET(3, track, "no need to drain"); 5647 return 0; 5648 } 5649 track->pstate = AUDIO_STATE_DRAINING; 5650 5651 for (;;) { 5652 /* I want to display it before condition evaluation. */ 5653 TRACET(3, track, "pid=%d.%d trkseq=%d hwseq=%d out=%d/%d/%d", 5654 (int)curproc->p_pid, (int)curlwp->l_lid, 5655 (int)track->seq, (int)mixer->hwseq, 5656 track->outbuf.head, track->outbuf.used, 5657 track->outbuf.capacity); 5658 5659 /* Condition to terminate */ 5660 audio_track_lock_enter(track); 5661 done = (track->usrbuf.used < frametobyte(&track->inputfmt, 1) && 5662 track->outbuf.used == 0 && 5663 track->seq <= mixer->hwseq); 5664 audio_track_lock_exit(track); 5665 if (done) 5666 break; 5667 5668 TRACET(3, track, "sleep"); 5669 error = audio_track_waitio(sc, track); 5670 if (error) 5671 return error; 5672 5673 /* XXX call audio_track_play here ? */ 5674 } 5675 5676 track->pstate = AUDIO_STATE_CLEAR; 5677 TRACET(3, track, "done trk_inp=%d trk_out=%d", 5678 (int)track->inputcounter, (int)track->outputcounter); 5679 return 0; 5680 } 5681 5682 /* 5683 * Send signal to process. 5684 * This is intended to be called only from audio_softintr_{rd,wr}. 5685 * Must be called with sc_lock && sc_intr_lock held. 5686 */ 5687 static inline void 5688 audio_psignal(struct audio_softc *sc, pid_t pid, int signum) 5689 { 5690 proc_t *p; 5691 5692 KASSERT(mutex_owned(sc->sc_lock)); 5693 KASSERT(mutex_owned(sc->sc_intr_lock)); 5694 KASSERT(pid != 0); 5695 5696 /* 5697 * psignal() must be called without spin lock held. 5698 * So leave intr_lock temporarily here. 5699 */ 5700 mutex_exit(sc->sc_intr_lock); 5701 5702 mutex_enter(proc_lock); 5703 p = proc_find(pid); 5704 if (p) 5705 psignal(p, signum); 5706 mutex_exit(proc_lock); 5707 5708 /* Enter intr_lock again */ 5709 mutex_enter(sc->sc_intr_lock); 5710 } 5711 5712 /* 5713 * This is software interrupt handler for record. 5714 * It is called from recording hardware interrupt everytime. 5715 * It does: 5716 * - Deliver SIGIO for all async processes. 5717 * - Notify to audio_read() that data has arrived. 5718 * - selnotify() for select/poll-ing processes. 5719 */ 5720 /* 5721 * XXX If a process issues FIOASYNC between hardware interrupt and 5722 * software interrupt, (stray) SIGIO will be sent to the process 5723 * despite the fact that it has not receive recorded data yet. 5724 */ 5725 static void 5726 audio_softintr_rd(void *cookie) 5727 { 5728 struct audio_softc *sc = cookie; 5729 audio_file_t *f; 5730 pid_t pid; 5731 5732 mutex_enter(sc->sc_lock); 5733 mutex_enter(sc->sc_intr_lock); 5734 5735 SLIST_FOREACH(f, &sc->sc_files, entry) { 5736 audio_track_t *track = f->rtrack; 5737 5738 if (track == NULL) 5739 continue; 5740 5741 TRACET(4, track, "broadcast; inp=%d/%d/%d", 5742 track->input->head, 5743 track->input->used, 5744 track->input->capacity); 5745 5746 pid = f->async_audio; 5747 if (pid != 0) { 5748 TRACEF(4, f, "sending SIGIO %d", pid); 5749 audio_psignal(sc, pid, SIGIO); 5750 } 5751 } 5752 mutex_exit(sc->sc_intr_lock); 5753 5754 /* Notify that data has arrived. */ 5755 selnotify(&sc->sc_rsel, 0, NOTE_SUBMIT); 5756 KNOTE(&sc->sc_rsel.sel_klist, 0); 5757 cv_broadcast(&sc->sc_rmixer->outcv); 5758 5759 mutex_exit(sc->sc_lock); 5760 } 5761 5762 /* 5763 * This is software interrupt handler for playback. 5764 * It is called from playback hardware interrupt everytime. 5765 * It does: 5766 * - Deliver SIGIO for all async and writable (used < lowat) processes. 5767 * - Notify to audio_write() that outbuf block available. 5768 * - selnotify() for select/poll-ing processes if there are any writable 5769 * (used < lowat) processes. Checking each descriptor will be done by 5770 * filt_audiowrite_event(). 5771 */ 5772 static void 5773 audio_softintr_wr(void *cookie) 5774 { 5775 struct audio_softc *sc = cookie; 5776 audio_file_t *f; 5777 bool found; 5778 pid_t pid; 5779 5780 TRACE(4, "called"); 5781 found = false; 5782 5783 mutex_enter(sc->sc_lock); 5784 mutex_enter(sc->sc_intr_lock); 5785 5786 SLIST_FOREACH(f, &sc->sc_files, entry) { 5787 audio_track_t *track = f->ptrack; 5788 5789 if (track == NULL) 5790 continue; 5791 5792 TRACET(4, track, "broadcast; trseq=%d out=%d/%d/%d", 5793 (int)track->seq, 5794 track->outbuf.head, 5795 track->outbuf.used, 5796 track->outbuf.capacity); 5797 5798 /* 5799 * Send a signal if the process is async mode and 5800 * used is lower than lowat. 5801 */ 5802 if (track->usrbuf.used <= track->usrbuf_usedlow && 5803 !track->is_pause) { 5804 /* For selnotify */ 5805 found = true; 5806 /* For SIGIO */ 5807 pid = f->async_audio; 5808 if (pid != 0) { 5809 TRACEF(4, f, "sending SIGIO %d", pid); 5810 audio_psignal(sc, pid, SIGIO); 5811 } 5812 } 5813 } 5814 mutex_exit(sc->sc_intr_lock); 5815 5816 /* 5817 * Notify for select/poll when someone become writable. 5818 * It needs sc_lock (and not sc_intr_lock). 5819 */ 5820 if (found) { 5821 TRACE(4, "selnotify"); 5822 selnotify(&sc->sc_wsel, 0, NOTE_SUBMIT); 5823 KNOTE(&sc->sc_wsel.sel_klist, 0); 5824 } 5825 5826 /* Notify to audio_write() that outbuf available. */ 5827 cv_broadcast(&sc->sc_pmixer->outcv); 5828 5829 mutex_exit(sc->sc_lock); 5830 } 5831 5832 /* 5833 * Check (and convert) the format *p came from userland. 5834 * If successful, it writes back the converted format to *p if necessary 5835 * and returns 0. Otherwise returns errno (*p may change even this case). 5836 */ 5837 static int 5838 audio_check_params(audio_format2_t *p) 5839 { 5840 5841 /* Convert obsoleted AUDIO_ENCODING_PCM* */ 5842 /* XXX Is this conversion right? */ 5843 if (p->encoding == AUDIO_ENCODING_PCM16) { 5844 if (p->precision == 8) 5845 p->encoding = AUDIO_ENCODING_ULINEAR; 5846 else 5847 p->encoding = AUDIO_ENCODING_SLINEAR; 5848 } else if (p->encoding == AUDIO_ENCODING_PCM8) { 5849 if (p->precision == 8) 5850 p->encoding = AUDIO_ENCODING_ULINEAR; 5851 else 5852 return EINVAL; 5853 } 5854 5855 /* 5856 * Convert obsoleted AUDIO_ENCODING_[SU]LINEAR without endianness 5857 * suffix. 5858 */ 5859 if (p->encoding == AUDIO_ENCODING_SLINEAR) 5860 p->encoding = AUDIO_ENCODING_SLINEAR_NE; 5861 if (p->encoding == AUDIO_ENCODING_ULINEAR) 5862 p->encoding = AUDIO_ENCODING_ULINEAR_NE; 5863 5864 switch (p->encoding) { 5865 case AUDIO_ENCODING_ULAW: 5866 case AUDIO_ENCODING_ALAW: 5867 if (p->precision != 8) 5868 return EINVAL; 5869 break; 5870 case AUDIO_ENCODING_ADPCM: 5871 if (p->precision != 4 && p->precision != 8) 5872 return EINVAL; 5873 break; 5874 case AUDIO_ENCODING_SLINEAR_LE: 5875 case AUDIO_ENCODING_SLINEAR_BE: 5876 case AUDIO_ENCODING_ULINEAR_LE: 5877 case AUDIO_ENCODING_ULINEAR_BE: 5878 if (p->precision != 8 && p->precision != 16 && 5879 p->precision != 24 && p->precision != 32) 5880 return EINVAL; 5881 5882 /* 8bit format does not have endianness. */ 5883 if (p->precision == 8) { 5884 if (p->encoding == AUDIO_ENCODING_SLINEAR_OE) 5885 p->encoding = AUDIO_ENCODING_SLINEAR_NE; 5886 if (p->encoding == AUDIO_ENCODING_ULINEAR_OE) 5887 p->encoding = AUDIO_ENCODING_ULINEAR_NE; 5888 } 5889 5890 if (p->precision > p->stride) 5891 return EINVAL; 5892 break; 5893 case AUDIO_ENCODING_MPEG_L1_STREAM: 5894 case AUDIO_ENCODING_MPEG_L1_PACKETS: 5895 case AUDIO_ENCODING_MPEG_L1_SYSTEM: 5896 case AUDIO_ENCODING_MPEG_L2_STREAM: 5897 case AUDIO_ENCODING_MPEG_L2_PACKETS: 5898 case AUDIO_ENCODING_MPEG_L2_SYSTEM: 5899 case AUDIO_ENCODING_AC3: 5900 break; 5901 default: 5902 return EINVAL; 5903 } 5904 5905 /* sanity check # of channels*/ 5906 if (p->channels < 1 || p->channels > AUDIO_MAX_CHANNELS) 5907 return EINVAL; 5908 5909 return 0; 5910 } 5911 5912 /* 5913 * Initialize playback and record mixers. 5914 * mode (AUMODE_{PLAY,RECORD}) indicates the mixer to be initialized. 5915 * phwfmt and rhwfmt indicate the hardware format. pfil and rfil indicate 5916 * the filter registration information. These four must not be NULL. 5917 * If successful returns 0. Otherwise returns errno. 5918 * Must be called with sc_lock held. 5919 * Must not be called if there are any tracks. 5920 * Caller should check that the initialization succeed by whether 5921 * sc_[pr]mixer is not NULL. 5922 */ 5923 static int 5924 audio_mixers_init(struct audio_softc *sc, int mode, 5925 const audio_format2_t *phwfmt, const audio_format2_t *rhwfmt, 5926 const audio_filter_reg_t *pfil, const audio_filter_reg_t *rfil) 5927 { 5928 int error; 5929 5930 KASSERT(phwfmt != NULL); 5931 KASSERT(rhwfmt != NULL); 5932 KASSERT(pfil != NULL); 5933 KASSERT(rfil != NULL); 5934 KASSERT(mutex_owned(sc->sc_lock)); 5935 5936 if ((mode & AUMODE_PLAY)) { 5937 if (sc->sc_pmixer == NULL) { 5938 sc->sc_pmixer = kmem_zalloc(sizeof(*sc->sc_pmixer), 5939 KM_SLEEP); 5940 } else { 5941 /* destroy() doesn't free memory. */ 5942 audio_mixer_destroy(sc, sc->sc_pmixer); 5943 memset(sc->sc_pmixer, 0, sizeof(*sc->sc_pmixer)); 5944 } 5945 error = audio_mixer_init(sc, AUMODE_PLAY, phwfmt, pfil); 5946 if (error) { 5947 aprint_error_dev(sc->sc_dev, 5948 "configuring playback mode failed\n"); 5949 kmem_free(sc->sc_pmixer, sizeof(*sc->sc_pmixer)); 5950 sc->sc_pmixer = NULL; 5951 return error; 5952 } 5953 } 5954 if ((mode & AUMODE_RECORD)) { 5955 if (sc->sc_rmixer == NULL) { 5956 sc->sc_rmixer = kmem_zalloc(sizeof(*sc->sc_rmixer), 5957 KM_SLEEP); 5958 } else { 5959 /* destroy() doesn't free memory. */ 5960 audio_mixer_destroy(sc, sc->sc_rmixer); 5961 memset(sc->sc_rmixer, 0, sizeof(*sc->sc_rmixer)); 5962 } 5963 error = audio_mixer_init(sc, AUMODE_RECORD, rhwfmt, rfil); 5964 if (error) { 5965 aprint_error_dev(sc->sc_dev, 5966 "configuring record mode failed\n"); 5967 kmem_free(sc->sc_rmixer, sizeof(*sc->sc_rmixer)); 5968 sc->sc_rmixer = NULL; 5969 return error; 5970 } 5971 } 5972 5973 return 0; 5974 } 5975 5976 /* 5977 * Select a frequency. 5978 * Prioritize 48kHz and 44.1kHz. Otherwise choose the highest one. 5979 * XXX Better algorithm? 5980 */ 5981 static int 5982 audio_select_freq(const struct audio_format *fmt) 5983 { 5984 int freq; 5985 int high; 5986 int low; 5987 int j; 5988 5989 if (fmt->frequency_type == 0) { 5990 low = fmt->frequency[0]; 5991 high = fmt->frequency[1]; 5992 freq = 48000; 5993 if (low <= freq && freq <= high) { 5994 return freq; 5995 } 5996 freq = 44100; 5997 if (low <= freq && freq <= high) { 5998 return freq; 5999 } 6000 return high; 6001 } else { 6002 for (j = 0; j < fmt->frequency_type; j++) { 6003 if (fmt->frequency[j] == 48000) { 6004 return fmt->frequency[j]; 6005 } 6006 } 6007 high = 0; 6008 for (j = 0; j < fmt->frequency_type; j++) { 6009 if (fmt->frequency[j] == 44100) { 6010 return fmt->frequency[j]; 6011 } 6012 if (fmt->frequency[j] > high) { 6013 high = fmt->frequency[j]; 6014 } 6015 } 6016 return high; 6017 } 6018 } 6019 6020 /* 6021 * Probe playback and/or recording format (depending on *modep). 6022 * *modep is an in-out parameter. It indicates the direction to configure 6023 * as an argument, and the direction configured is written back as out 6024 * parameter. 6025 * If successful, probed hardware format is stored into *phwfmt, *rhwfmt 6026 * depending on *modep, and return 0. Otherwise it returns errno. 6027 * Must be called with sc_lock held. 6028 */ 6029 static int 6030 audio_hw_probe(struct audio_softc *sc, int is_indep, int *modep, 6031 audio_format2_t *phwfmt, audio_format2_t *rhwfmt) 6032 { 6033 audio_format2_t fmt; 6034 int mode; 6035 int error = 0; 6036 6037 KASSERT(mutex_owned(sc->sc_lock)); 6038 6039 mode = *modep; 6040 KASSERTMSG((mode & (AUMODE_PLAY | AUMODE_RECORD)) != 0, 6041 "invalid mode = %x", mode); 6042 6043 if (is_indep) { 6044 int errorp = 0, errorr = 0; 6045 6046 /* On independent devices, probe separately. */ 6047 if ((mode & AUMODE_PLAY) != 0) { 6048 errorp = audio_hw_probe_fmt(sc, phwfmt, AUMODE_PLAY); 6049 if (errorp) 6050 mode &= ~AUMODE_PLAY; 6051 } 6052 if ((mode & AUMODE_RECORD) != 0) { 6053 errorr = audio_hw_probe_fmt(sc, rhwfmt, AUMODE_RECORD); 6054 if (errorr) 6055 mode &= ~AUMODE_RECORD; 6056 } 6057 6058 /* Return error if both play and record probes failed. */ 6059 if (errorp && errorr) 6060 error = errorp; 6061 } else { 6062 /* On non independent devices, probe simultaneously. */ 6063 error = audio_hw_probe_fmt(sc, &fmt, mode); 6064 if (error) { 6065 mode = 0; 6066 } else { 6067 *phwfmt = fmt; 6068 *rhwfmt = fmt; 6069 } 6070 } 6071 6072 *modep = mode; 6073 return error; 6074 } 6075 6076 /* 6077 * Choose the most preferred hardware format. 6078 * If successful, it will store the chosen format into *cand and return 0. 6079 * Otherwise, return errno. 6080 * Must be called with sc_lock held. 6081 */ 6082 static int 6083 audio_hw_probe_fmt(struct audio_softc *sc, audio_format2_t *cand, int mode) 6084 { 6085 audio_format_query_t query; 6086 int cand_score; 6087 int score; 6088 int i; 6089 int error; 6090 6091 KASSERT(mutex_owned(sc->sc_lock)); 6092 6093 /* 6094 * Score each formats and choose the highest one. 6095 * 6096 * +---- priority(0-3) 6097 * |+--- encoding/precision 6098 * ||+-- channels 6099 * score = 0x000000PEC 6100 */ 6101 6102 cand_score = 0; 6103 for (i = 0; ; i++) { 6104 memset(&query, 0, sizeof(query)); 6105 query.index = i; 6106 6107 error = sc->hw_if->query_format(sc->hw_hdl, &query); 6108 if (error == EINVAL) 6109 break; 6110 if (error) 6111 return error; 6112 6113 #if defined(AUDIO_DEBUG) 6114 DPRINTF(1, "fmt[%d] %c%c pri=%d %s,%d/%dbit,%dch,", i, 6115 (query.fmt.mode & AUMODE_PLAY) ? 'P' : '-', 6116 (query.fmt.mode & AUMODE_RECORD) ? 'R' : '-', 6117 query.fmt.priority, 6118 audio_encoding_name(query.fmt.encoding), 6119 query.fmt.validbits, 6120 query.fmt.precision, 6121 query.fmt.channels); 6122 if (query.fmt.frequency_type == 0) { 6123 DPRINTF(1, "{%d-%d", 6124 query.fmt.frequency[0], query.fmt.frequency[1]); 6125 } else { 6126 int j; 6127 for (j = 0; j < query.fmt.frequency_type; j++) { 6128 DPRINTF(1, "%c%d", 6129 (j == 0) ? '{' : ',', 6130 query.fmt.frequency[j]); 6131 } 6132 } 6133 DPRINTF(1, "}\n"); 6134 #endif 6135 6136 if ((query.fmt.mode & mode) == 0) { 6137 DPRINTF(1, "fmt[%d] skip; mode not match %d\n", i, 6138 mode); 6139 continue; 6140 } 6141 6142 if (query.fmt.priority < 0) { 6143 DPRINTF(1, "fmt[%d] skip; unsupported encoding\n", i); 6144 continue; 6145 } 6146 6147 /* Score */ 6148 score = (query.fmt.priority & 3) * 0x100; 6149 if (query.fmt.encoding == AUDIO_ENCODING_SLINEAR_NE && 6150 query.fmt.validbits == AUDIO_INTERNAL_BITS && 6151 query.fmt.precision == AUDIO_INTERNAL_BITS) { 6152 score += 0x20; 6153 } else if (query.fmt.encoding == AUDIO_ENCODING_SLINEAR_OE && 6154 query.fmt.validbits == AUDIO_INTERNAL_BITS && 6155 query.fmt.precision == AUDIO_INTERNAL_BITS) { 6156 score += 0x10; 6157 } 6158 score += query.fmt.channels; 6159 6160 if (score < cand_score) { 6161 DPRINTF(1, "fmt[%d] skip; score 0x%x < 0x%x\n", i, 6162 score, cand_score); 6163 continue; 6164 } 6165 6166 /* Update candidate */ 6167 cand_score = score; 6168 cand->encoding = query.fmt.encoding; 6169 cand->precision = query.fmt.validbits; 6170 cand->stride = query.fmt.precision; 6171 cand->channels = query.fmt.channels; 6172 cand->sample_rate = audio_select_freq(&query.fmt); 6173 DPRINTF(1, "fmt[%d] candidate (score=0x%x)" 6174 " pri=%d %s,%d/%d,%dch,%dHz\n", i, 6175 cand_score, query.fmt.priority, 6176 audio_encoding_name(query.fmt.encoding), 6177 cand->precision, cand->stride, 6178 cand->channels, cand->sample_rate); 6179 } 6180 6181 if (cand_score == 0) { 6182 DPRINTF(1, "%s no fmt\n", __func__); 6183 return ENXIO; 6184 } 6185 DPRINTF(1, "%s selected: %s,%d/%d,%dch,%dHz\n", __func__, 6186 audio_encoding_name(cand->encoding), 6187 cand->precision, cand->stride, cand->channels, cand->sample_rate); 6188 return 0; 6189 } 6190 6191 /* 6192 * Validate fmt with query_format. 6193 * If fmt is included in the result of query_format, returns 0. 6194 * Otherwise returns EINVAL. 6195 * Must be called with sc_lock held. 6196 */ 6197 static int 6198 audio_hw_validate_format(struct audio_softc *sc, int mode, 6199 const audio_format2_t *fmt) 6200 { 6201 audio_format_query_t query; 6202 struct audio_format *q; 6203 int index; 6204 int error; 6205 int j; 6206 6207 KASSERT(mutex_owned(sc->sc_lock)); 6208 6209 /* 6210 * If query_format is not supported by hardware driver, 6211 * a rough check instead will be performed. 6212 * XXX This will gone in the future. 6213 */ 6214 if (sc->hw_if->query_format == NULL) { 6215 if (fmt->encoding != AUDIO_ENCODING_SLINEAR_NE) 6216 return EINVAL; 6217 if (fmt->precision != AUDIO_INTERNAL_BITS) 6218 return EINVAL; 6219 if (fmt->stride != AUDIO_INTERNAL_BITS) 6220 return EINVAL; 6221 return 0; 6222 } 6223 6224 for (index = 0; ; index++) { 6225 query.index = index; 6226 error = sc->hw_if->query_format(sc->hw_hdl, &query); 6227 if (error == EINVAL) 6228 break; 6229 if (error) 6230 return error; 6231 6232 q = &query.fmt; 6233 /* 6234 * Note that fmt is audio_format2_t (precision/stride) but 6235 * q is audio_format_t (validbits/precision). 6236 */ 6237 if ((q->mode & mode) == 0) { 6238 continue; 6239 } 6240 if (fmt->encoding != q->encoding) { 6241 continue; 6242 } 6243 if (fmt->precision != q->validbits) { 6244 continue; 6245 } 6246 if (fmt->stride != q->precision) { 6247 continue; 6248 } 6249 if (fmt->channels != q->channels) { 6250 continue; 6251 } 6252 if (q->frequency_type == 0) { 6253 if (fmt->sample_rate < q->frequency[0] || 6254 fmt->sample_rate > q->frequency[1]) { 6255 continue; 6256 } 6257 } else { 6258 for (j = 0; j < q->frequency_type; j++) { 6259 if (fmt->sample_rate == q->frequency[j]) 6260 break; 6261 } 6262 if (j == query.fmt.frequency_type) { 6263 continue; 6264 } 6265 } 6266 6267 /* Matched. */ 6268 return 0; 6269 } 6270 6271 return EINVAL; 6272 } 6273 6274 /* 6275 * Set track mixer's format depending on ai->mode. 6276 * If AUMODE_PLAY is set in ai->mode, it set up the playback mixer 6277 * with ai.play.*. 6278 * If AUMODE_RECORD is set in ai->mode, it set up the recording mixer 6279 * with ai.record.*. 6280 * All other fields in ai are ignored. 6281 * If successful returns 0. Otherwise returns errno. 6282 * This function does not roll back even if it fails. 6283 * Must be called with sc_lock held. 6284 */ 6285 static int 6286 audio_mixers_set_format(struct audio_softc *sc, const struct audio_info *ai) 6287 { 6288 audio_format2_t phwfmt; 6289 audio_format2_t rhwfmt; 6290 audio_filter_reg_t pfil; 6291 audio_filter_reg_t rfil; 6292 int mode; 6293 int error; 6294 6295 KASSERT(mutex_owned(sc->sc_lock)); 6296 6297 /* 6298 * Even when setting either one of playback and recording, 6299 * both must be halted. 6300 */ 6301 if (sc->sc_popens + sc->sc_ropens > 0) 6302 return EBUSY; 6303 6304 if (!SPECIFIED(ai->mode) || ai->mode == 0) 6305 return ENOTTY; 6306 6307 mode = ai->mode; 6308 if ((mode & AUMODE_PLAY)) { 6309 phwfmt.encoding = ai->play.encoding; 6310 phwfmt.precision = ai->play.precision; 6311 phwfmt.stride = ai->play.precision; 6312 phwfmt.channels = ai->play.channels; 6313 phwfmt.sample_rate = ai->play.sample_rate; 6314 } 6315 if ((mode & AUMODE_RECORD)) { 6316 rhwfmt.encoding = ai->record.encoding; 6317 rhwfmt.precision = ai->record.precision; 6318 rhwfmt.stride = ai->record.precision; 6319 rhwfmt.channels = ai->record.channels; 6320 rhwfmt.sample_rate = ai->record.sample_rate; 6321 } 6322 6323 /* On non-independent devices, use the same format for both. */ 6324 if ((sc->sc_props & AUDIO_PROP_INDEPENDENT) == 0) { 6325 if (mode == AUMODE_RECORD) { 6326 phwfmt = rhwfmt; 6327 } else { 6328 rhwfmt = phwfmt; 6329 } 6330 mode = AUMODE_PLAY | AUMODE_RECORD; 6331 } 6332 6333 /* Then, unset the direction not exist on the hardware. */ 6334 if ((sc->sc_props & AUDIO_PROP_PLAYBACK) == 0) 6335 mode &= ~AUMODE_PLAY; 6336 if ((sc->sc_props & AUDIO_PROP_CAPTURE) == 0) 6337 mode &= ~AUMODE_RECORD; 6338 6339 /* debug */ 6340 if ((mode & AUMODE_PLAY)) { 6341 TRACE(1, "play=%s/%d/%d/%dch/%dHz", 6342 audio_encoding_name(phwfmt.encoding), 6343 phwfmt.precision, 6344 phwfmt.stride, 6345 phwfmt.channels, 6346 phwfmt.sample_rate); 6347 } 6348 if ((mode & AUMODE_RECORD)) { 6349 TRACE(1, "rec =%s/%d/%d/%dch/%dHz", 6350 audio_encoding_name(rhwfmt.encoding), 6351 rhwfmt.precision, 6352 rhwfmt.stride, 6353 rhwfmt.channels, 6354 rhwfmt.sample_rate); 6355 } 6356 6357 /* Check the format */ 6358 if ((mode & AUMODE_PLAY)) { 6359 if (audio_hw_validate_format(sc, AUMODE_PLAY, &phwfmt)) { 6360 TRACE(1, "invalid format"); 6361 return EINVAL; 6362 } 6363 } 6364 if ((mode & AUMODE_RECORD)) { 6365 if (audio_hw_validate_format(sc, AUMODE_RECORD, &rhwfmt)) { 6366 TRACE(1, "invalid format"); 6367 return EINVAL; 6368 } 6369 } 6370 6371 /* Configure the mixers. */ 6372 memset(&pfil, 0, sizeof(pfil)); 6373 memset(&rfil, 0, sizeof(rfil)); 6374 error = audio_hw_set_format(sc, mode, &phwfmt, &rhwfmt, &pfil, &rfil); 6375 if (error) 6376 return error; 6377 6378 error = audio_mixers_init(sc, mode, &phwfmt, &rhwfmt, &pfil, &rfil); 6379 if (error) 6380 return error; 6381 6382 return 0; 6383 } 6384 6385 /* 6386 * Store current mixers format into *ai. 6387 */ 6388 static void 6389 audio_mixers_get_format(struct audio_softc *sc, struct audio_info *ai) 6390 { 6391 /* 6392 * There is no stride information in audio_info but it doesn't matter. 6393 * trackmixer always treats stride and precision as the same. 6394 */ 6395 AUDIO_INITINFO(ai); 6396 ai->mode = 0; 6397 if (sc->sc_pmixer) { 6398 audio_format2_t *fmt = &sc->sc_pmixer->track_fmt; 6399 ai->play.encoding = fmt->encoding; 6400 ai->play.precision = fmt->precision; 6401 ai->play.channels = fmt->channels; 6402 ai->play.sample_rate = fmt->sample_rate; 6403 ai->mode |= AUMODE_PLAY; 6404 } 6405 if (sc->sc_rmixer) { 6406 audio_format2_t *fmt = &sc->sc_rmixer->track_fmt; 6407 ai->record.encoding = fmt->encoding; 6408 ai->record.precision = fmt->precision; 6409 ai->record.channels = fmt->channels; 6410 ai->record.sample_rate = fmt->sample_rate; 6411 ai->mode |= AUMODE_RECORD; 6412 } 6413 } 6414 6415 /* 6416 * audio_info details: 6417 * 6418 * ai.{play,record}.sample_rate (R/W) 6419 * ai.{play,record}.encoding (R/W) 6420 * ai.{play,record}.precision (R/W) 6421 * ai.{play,record}.channels (R/W) 6422 * These specify the playback or recording format. 6423 * Ignore members within an inactive track. 6424 * 6425 * ai.mode (R/W) 6426 * It specifies the playback or recording mode, AUMODE_*. 6427 * Currently, a mode change operation by ai.mode after opening is 6428 * prohibited. In addition, AUMODE_PLAY_ALL no longer makes sense. 6429 * However, it's possible to get or to set for backward compatibility. 6430 * 6431 * ai.{hiwat,lowat} (R/W) 6432 * These specify the high water mark and low water mark for playback 6433 * track. The unit is block. 6434 * 6435 * ai.{play,record}.gain (R/W) 6436 * It specifies the HW mixer volume in 0-255. 6437 * It is historical reason that the gain is connected to HW mixer. 6438 * 6439 * ai.{play,record}.balance (R/W) 6440 * It specifies the left-right balance of HW mixer in 0-64. 6441 * 32 means the center. 6442 * It is historical reason that the balance is connected to HW mixer. 6443 * 6444 * ai.{play,record}.port (R/W) 6445 * It specifies the input/output port of HW mixer. 6446 * 6447 * ai.monitor_gain (R/W) 6448 * It specifies the recording monitor gain(?) of HW mixer. 6449 * 6450 * ai.{play,record}.pause (R/W) 6451 * Non-zero means the track is paused. 6452 * 6453 * ai.play.seek (R/-) 6454 * It indicates the number of bytes written but not processed. 6455 * ai.record.seek (R/-) 6456 * It indicates the number of bytes to be able to read. 6457 * 6458 * ai.{play,record}.avail_ports (R/-) 6459 * Mixer info. 6460 * 6461 * ai.{play,record}.buffer_size (R/-) 6462 * It indicates the buffer size in bytes. Internally it means usrbuf. 6463 * 6464 * ai.{play,record}.samples (R/-) 6465 * It indicates the total number of bytes played or recorded. 6466 * 6467 * ai.{play,record}.eof (R/-) 6468 * It indicates the number of times reached EOF(?). 6469 * 6470 * ai.{play,record}.error (R/-) 6471 * Non-zero indicates overflow/underflow has occured. 6472 * 6473 * ai.{play,record}.waiting (R/-) 6474 * Non-zero indicates that other process waits to open. 6475 * It will never happen anymore. 6476 * 6477 * ai.{play,record}.open (R/-) 6478 * Non-zero indicates the direction is opened by this process(?). 6479 * XXX Is this better to indicate that "the device is opened by 6480 * at least one process"? 6481 * 6482 * ai.{play,record}.active (R/-) 6483 * Non-zero indicates that I/O is currently active. 6484 * 6485 * ai.blocksize (R/-) 6486 * It indicates the block size in bytes. 6487 * XXX The blocksize of playback and recording may be different. 6488 */ 6489 6490 /* 6491 * Pause consideration: 6492 * 6493 * The introduction of these two behavior makes pause/unpause operation 6494 * simple. 6495 * 1. The first read/write access of the first track makes mixer start. 6496 * 2. A pause of the last track doesn't make mixer stop. 6497 */ 6498 6499 /* 6500 * Set both track's parameters within a file depending on ai. 6501 * Update sc_sound_[pr]* if set. 6502 * Must be called with sc_lock and sc_exlock held. 6503 */ 6504 static int 6505 audio_file_setinfo(struct audio_softc *sc, audio_file_t *file, 6506 const struct audio_info *ai) 6507 { 6508 const struct audio_prinfo *pi; 6509 const struct audio_prinfo *ri; 6510 audio_track_t *ptrack; 6511 audio_track_t *rtrack; 6512 audio_format2_t pfmt; 6513 audio_format2_t rfmt; 6514 int pchanges; 6515 int rchanges; 6516 int mode; 6517 struct audio_info saved_ai; 6518 audio_format2_t saved_pfmt; 6519 audio_format2_t saved_rfmt; 6520 int error; 6521 6522 KASSERT(mutex_owned(sc->sc_lock)); 6523 KASSERT(sc->sc_exlock); 6524 6525 pi = &ai->play; 6526 ri = &ai->record; 6527 pchanges = 0; 6528 rchanges = 0; 6529 6530 ptrack = file->ptrack; 6531 rtrack = file->rtrack; 6532 6533 #if defined(AUDIO_DEBUG) 6534 if (audiodebug >= 2) { 6535 char buf[256]; 6536 char p[64]; 6537 int buflen; 6538 int plen; 6539 #define SPRINTF(var, fmt...) do { \ 6540 var##len += snprintf(var + var##len, sizeof(var) - var##len, fmt); \ 6541 } while (0) 6542 6543 buflen = 0; 6544 plen = 0; 6545 if (SPECIFIED(pi->encoding)) 6546 SPRINTF(p, "/%s", audio_encoding_name(pi->encoding)); 6547 if (SPECIFIED(pi->precision)) 6548 SPRINTF(p, "/%dbit", pi->precision); 6549 if (SPECIFIED(pi->channels)) 6550 SPRINTF(p, "/%dch", pi->channels); 6551 if (SPECIFIED(pi->sample_rate)) 6552 SPRINTF(p, "/%dHz", pi->sample_rate); 6553 if (plen > 0) 6554 SPRINTF(buf, ",play.param=%s", p + 1); 6555 6556 plen = 0; 6557 if (SPECIFIED(ri->encoding)) 6558 SPRINTF(p, "/%s", audio_encoding_name(ri->encoding)); 6559 if (SPECIFIED(ri->precision)) 6560 SPRINTF(p, "/%dbit", ri->precision); 6561 if (SPECIFIED(ri->channels)) 6562 SPRINTF(p, "/%dch", ri->channels); 6563 if (SPECIFIED(ri->sample_rate)) 6564 SPRINTF(p, "/%dHz", ri->sample_rate); 6565 if (plen > 0) 6566 SPRINTF(buf, ",record.param=%s", p + 1); 6567 6568 if (SPECIFIED(ai->mode)) 6569 SPRINTF(buf, ",mode=%d", ai->mode); 6570 if (SPECIFIED(ai->hiwat)) 6571 SPRINTF(buf, ",hiwat=%d", ai->hiwat); 6572 if (SPECIFIED(ai->lowat)) 6573 SPRINTF(buf, ",lowat=%d", ai->lowat); 6574 if (SPECIFIED(ai->play.gain)) 6575 SPRINTF(buf, ",play.gain=%d", ai->play.gain); 6576 if (SPECIFIED(ai->record.gain)) 6577 SPRINTF(buf, ",record.gain=%d", ai->record.gain); 6578 if (SPECIFIED_CH(ai->play.balance)) 6579 SPRINTF(buf, ",play.balance=%d", ai->play.balance); 6580 if (SPECIFIED_CH(ai->record.balance)) 6581 SPRINTF(buf, ",record.balance=%d", ai->record.balance); 6582 if (SPECIFIED(ai->play.port)) 6583 SPRINTF(buf, ",play.port=%d", ai->play.port); 6584 if (SPECIFIED(ai->record.port)) 6585 SPRINTF(buf, ",record.port=%d", ai->record.port); 6586 if (SPECIFIED(ai->monitor_gain)) 6587 SPRINTF(buf, ",monitor_gain=%d", ai->monitor_gain); 6588 if (SPECIFIED_CH(ai->play.pause)) 6589 SPRINTF(buf, ",play.pause=%d", ai->play.pause); 6590 if (SPECIFIED_CH(ai->record.pause)) 6591 SPRINTF(buf, ",record.pause=%d", ai->record.pause); 6592 6593 if (buflen > 0) 6594 TRACE(2, "specified %s", buf + 1); 6595 } 6596 #endif 6597 6598 AUDIO_INITINFO(&saved_ai); 6599 /* XXX shut up gcc */ 6600 memset(&saved_pfmt, 0, sizeof(saved_pfmt)); 6601 memset(&saved_rfmt, 0, sizeof(saved_rfmt)); 6602 6603 /* Set default value and save current parameters */ 6604 if (ptrack) { 6605 pfmt = ptrack->usrbuf.fmt; 6606 saved_pfmt = ptrack->usrbuf.fmt; 6607 saved_ai.play.pause = ptrack->is_pause; 6608 } 6609 if (rtrack) { 6610 rfmt = rtrack->usrbuf.fmt; 6611 saved_rfmt = rtrack->usrbuf.fmt; 6612 saved_ai.record.pause = rtrack->is_pause; 6613 } 6614 saved_ai.mode = file->mode; 6615 6616 /* Overwrite if specified */ 6617 mode = file->mode; 6618 if (SPECIFIED(ai->mode)) { 6619 /* 6620 * Setting ai->mode no longer does anything because it's 6621 * prohibited to change playback/recording mode after open 6622 * and AUMODE_PLAY_ALL is obsoleted. However, it still 6623 * keeps the state of AUMODE_PLAY_ALL itself for backward 6624 * compatibility. 6625 * In the internal, only file->mode has the state of 6626 * AUMODE_PLAY_ALL flag and track->mode in both track does 6627 * not have. 6628 */ 6629 if ((file->mode & AUMODE_PLAY)) { 6630 mode = (file->mode & (AUMODE_PLAY | AUMODE_RECORD)) 6631 | (ai->mode & AUMODE_PLAY_ALL); 6632 } 6633 } 6634 6635 if (ptrack) { 6636 pchanges = audio_track_setinfo_check(&pfmt, pi, 6637 &sc->sc_pmixer->hwbuf.fmt); 6638 if (pchanges == -1) { 6639 #if defined(AUDIO_DEBUG) 6640 char fmtbuf[64]; 6641 audio_format2_tostr(fmtbuf, sizeof(fmtbuf), &pfmt); 6642 TRACET(1, ptrack, "check play.params failed: %s", 6643 fmtbuf); 6644 #endif 6645 return EINVAL; 6646 } 6647 if (SPECIFIED(ai->mode)) 6648 pchanges = 1; 6649 } 6650 if (rtrack) { 6651 rchanges = audio_track_setinfo_check(&rfmt, ri, 6652 &sc->sc_rmixer->hwbuf.fmt); 6653 if (rchanges == -1) { 6654 #if defined(AUDIO_DEBUG) 6655 char fmtbuf[64]; 6656 audio_format2_tostr(fmtbuf, sizeof(fmtbuf), &rfmt); 6657 TRACET(1, rtrack, "check record.params failed: %s", 6658 fmtbuf); 6659 #endif 6660 return EINVAL; 6661 } 6662 if (SPECIFIED(ai->mode)) 6663 rchanges = 1; 6664 } 6665 6666 /* 6667 * Even when setting either one of playback and recording, 6668 * both track must be halted. 6669 */ 6670 if (pchanges || rchanges) { 6671 audio_file_clear(sc, file); 6672 #if defined(AUDIO_DEBUG) 6673 char fmtbuf[64]; 6674 if (pchanges) { 6675 audio_format2_tostr(fmtbuf, sizeof(fmtbuf), &pfmt); 6676 DPRINTF(1, "audio track#%d play mode: %s\n", 6677 ptrack->id, fmtbuf); 6678 } 6679 if (rchanges) { 6680 audio_format2_tostr(fmtbuf, sizeof(fmtbuf), &rfmt); 6681 DPRINTF(1, "audio track#%d rec mode: %s\n", 6682 rtrack->id, fmtbuf); 6683 } 6684 #endif 6685 } 6686 6687 /* Set mixer parameters */ 6688 error = audio_hw_setinfo(sc, ai, &saved_ai); 6689 if (error) 6690 goto abort1; 6691 6692 /* Set to track and update sticky parameters */ 6693 error = 0; 6694 file->mode = mode; 6695 if (ptrack) { 6696 if (SPECIFIED_CH(pi->pause)) { 6697 ptrack->is_pause = pi->pause; 6698 sc->sc_sound_ppause = pi->pause; 6699 } 6700 if (pchanges) { 6701 audio_track_lock_enter(ptrack); 6702 error = audio_track_set_format(ptrack, &pfmt); 6703 audio_track_lock_exit(ptrack); 6704 if (error) { 6705 TRACET(1, ptrack, "set play.params failed"); 6706 goto abort2; 6707 } 6708 sc->sc_sound_pparams = pfmt; 6709 } 6710 /* Change water marks after initializing the buffers. */ 6711 if (SPECIFIED(ai->hiwat) || SPECIFIED(ai->lowat)) 6712 audio_track_setinfo_water(ptrack, ai); 6713 } 6714 if (rtrack) { 6715 if (SPECIFIED_CH(ri->pause)) { 6716 rtrack->is_pause = ri->pause; 6717 sc->sc_sound_rpause = ri->pause; 6718 } 6719 if (rchanges) { 6720 audio_track_lock_enter(rtrack); 6721 error = audio_track_set_format(rtrack, &rfmt); 6722 audio_track_lock_exit(rtrack); 6723 if (error) { 6724 TRACET(1, rtrack, "set record.params failed"); 6725 goto abort3; 6726 } 6727 sc->sc_sound_rparams = rfmt; 6728 } 6729 } 6730 6731 return 0; 6732 6733 /* Rollback */ 6734 abort3: 6735 if (error != ENOMEM) { 6736 rtrack->is_pause = saved_ai.record.pause; 6737 audio_track_lock_enter(rtrack); 6738 audio_track_set_format(rtrack, &saved_rfmt); 6739 audio_track_lock_exit(rtrack); 6740 } 6741 abort2: 6742 if (ptrack && error != ENOMEM) { 6743 ptrack->is_pause = saved_ai.play.pause; 6744 audio_track_lock_enter(ptrack); 6745 audio_track_set_format(ptrack, &saved_pfmt); 6746 audio_track_lock_exit(ptrack); 6747 sc->sc_sound_pparams = saved_pfmt; 6748 sc->sc_sound_ppause = saved_ai.play.pause; 6749 } 6750 file->mode = saved_ai.mode; 6751 abort1: 6752 audio_hw_setinfo(sc, &saved_ai, NULL); 6753 6754 return error; 6755 } 6756 6757 /* 6758 * Write SPECIFIED() parameters within info back to fmt. 6759 * Return value of 1 indicates that fmt is modified. 6760 * Return value of 0 indicates that fmt is not modified. 6761 * Return value of -1 indicates that error EINVAL has occurred. 6762 */ 6763 static int 6764 audio_track_setinfo_check(audio_format2_t *fmt, const struct audio_prinfo *info, 6765 const audio_format2_t *hwfmt) 6766 { 6767 int changes; 6768 6769 changes = 0; 6770 if (SPECIFIED(info->sample_rate)) { 6771 if (info->sample_rate < AUDIO_MIN_FREQUENCY) 6772 return -1; 6773 if (info->sample_rate > AUDIO_MAX_FREQUENCY) 6774 return -1; 6775 fmt->sample_rate = info->sample_rate; 6776 changes = 1; 6777 } 6778 if (SPECIFIED(info->encoding)) { 6779 fmt->encoding = info->encoding; 6780 changes = 1; 6781 } 6782 if (SPECIFIED(info->precision)) { 6783 fmt->precision = info->precision; 6784 /* we don't have API to specify stride */ 6785 fmt->stride = info->precision; 6786 changes = 1; 6787 } 6788 if (SPECIFIED(info->channels)) { 6789 /* 6790 * We can convert between monaural and stereo each other. 6791 * We can reduce than the number of channels that the hardware 6792 * supports. 6793 */ 6794 if (info->channels > 2 && info->channels > hwfmt->channels) 6795 return -1; 6796 fmt->channels = info->channels; 6797 changes = 1; 6798 } 6799 6800 if (changes) { 6801 if (audio_check_params(fmt) != 0) 6802 return -1; 6803 } 6804 6805 return changes; 6806 } 6807 6808 /* 6809 * Change water marks for playback track if specfied. 6810 */ 6811 static void 6812 audio_track_setinfo_water(audio_track_t *track, const struct audio_info *ai) 6813 { 6814 u_int blks; 6815 u_int maxblks; 6816 u_int blksize; 6817 6818 KASSERT(audio_track_is_playback(track)); 6819 6820 blksize = track->usrbuf_blksize; 6821 maxblks = track->usrbuf.capacity / blksize; 6822 6823 if (SPECIFIED(ai->hiwat)) { 6824 blks = ai->hiwat; 6825 if (blks > maxblks) 6826 blks = maxblks; 6827 if (blks < 2) 6828 blks = 2; 6829 track->usrbuf_usedhigh = blks * blksize; 6830 } 6831 if (SPECIFIED(ai->lowat)) { 6832 blks = ai->lowat; 6833 if (blks > maxblks - 1) 6834 blks = maxblks - 1; 6835 track->usrbuf_usedlow = blks * blksize; 6836 } 6837 if (SPECIFIED(ai->hiwat) || SPECIFIED(ai->lowat)) { 6838 if (track->usrbuf_usedlow > track->usrbuf_usedhigh - blksize) { 6839 track->usrbuf_usedlow = track->usrbuf_usedhigh - 6840 blksize; 6841 } 6842 } 6843 } 6844 6845 /* 6846 * Set hardware part of *newai. 6847 * The parameters handled here are *.port, *.gain, *.balance and monitor_gain. 6848 * If oldai is specified, previous parameters are stored. 6849 * This function itself does not roll back if error occurred. 6850 * Must be called with sc_lock and sc_exlock held. 6851 */ 6852 static int 6853 audio_hw_setinfo(struct audio_softc *sc, const struct audio_info *newai, 6854 struct audio_info *oldai) 6855 { 6856 const struct audio_prinfo *newpi; 6857 const struct audio_prinfo *newri; 6858 struct audio_prinfo *oldpi; 6859 struct audio_prinfo *oldri; 6860 u_int pgain; 6861 u_int rgain; 6862 u_char pbalance; 6863 u_char rbalance; 6864 int error; 6865 6866 KASSERT(mutex_owned(sc->sc_lock)); 6867 KASSERT(sc->sc_exlock); 6868 6869 /* XXX shut up gcc */ 6870 oldpi = NULL; 6871 oldri = NULL; 6872 6873 newpi = &newai->play; 6874 newri = &newai->record; 6875 if (oldai) { 6876 oldpi = &oldai->play; 6877 oldri = &oldai->record; 6878 } 6879 error = 0; 6880 6881 /* 6882 * It looks like unnecessary to halt HW mixers to set HW mixers. 6883 * mixer_ioctl(MIXER_WRITE) also doesn't halt. 6884 */ 6885 6886 if (SPECIFIED(newpi->port)) { 6887 if (oldai) 6888 oldpi->port = au_get_port(sc, &sc->sc_outports); 6889 error = au_set_port(sc, &sc->sc_outports, newpi->port); 6890 if (error) { 6891 device_printf(sc->sc_dev, 6892 "setting play.port=%d failed with %d\n", 6893 newpi->port, error); 6894 goto abort; 6895 } 6896 } 6897 if (SPECIFIED(newri->port)) { 6898 if (oldai) 6899 oldri->port = au_get_port(sc, &sc->sc_inports); 6900 error = au_set_port(sc, &sc->sc_inports, newri->port); 6901 if (error) { 6902 device_printf(sc->sc_dev, 6903 "setting record.port=%d failed with %d\n", 6904 newri->port, error); 6905 goto abort; 6906 } 6907 } 6908 6909 /* Backup play.{gain,balance} */ 6910 if (SPECIFIED(newpi->gain) || SPECIFIED_CH(newpi->balance)) { 6911 au_get_gain(sc, &sc->sc_outports, &pgain, &pbalance); 6912 if (oldai) { 6913 oldpi->gain = pgain; 6914 oldpi->balance = pbalance; 6915 } 6916 } 6917 /* Backup record.{gain,balance} */ 6918 if (SPECIFIED(newri->gain) || SPECIFIED_CH(newri->balance)) { 6919 au_get_gain(sc, &sc->sc_inports, &rgain, &rbalance); 6920 if (oldai) { 6921 oldri->gain = rgain; 6922 oldri->balance = rbalance; 6923 } 6924 } 6925 if (SPECIFIED(newpi->gain)) { 6926 error = au_set_gain(sc, &sc->sc_outports, 6927 newpi->gain, pbalance); 6928 if (error) { 6929 device_printf(sc->sc_dev, 6930 "setting play.gain=%d failed with %d\n", 6931 newpi->gain, error); 6932 goto abort; 6933 } 6934 } 6935 if (SPECIFIED(newri->gain)) { 6936 error = au_set_gain(sc, &sc->sc_inports, 6937 newri->gain, rbalance); 6938 if (error) { 6939 device_printf(sc->sc_dev, 6940 "setting record.gain=%d failed with %d\n", 6941 newri->gain, error); 6942 goto abort; 6943 } 6944 } 6945 if (SPECIFIED_CH(newpi->balance)) { 6946 error = au_set_gain(sc, &sc->sc_outports, 6947 pgain, newpi->balance); 6948 if (error) { 6949 device_printf(sc->sc_dev, 6950 "setting play.balance=%d failed with %d\n", 6951 newpi->balance, error); 6952 goto abort; 6953 } 6954 } 6955 if (SPECIFIED_CH(newri->balance)) { 6956 error = au_set_gain(sc, &sc->sc_inports, 6957 rgain, newri->balance); 6958 if (error) { 6959 device_printf(sc->sc_dev, 6960 "setting record.balance=%d failed with %d\n", 6961 newri->balance, error); 6962 goto abort; 6963 } 6964 } 6965 6966 if (SPECIFIED(newai->monitor_gain) && sc->sc_monitor_port != -1) { 6967 if (oldai) 6968 oldai->monitor_gain = au_get_monitor_gain(sc); 6969 error = au_set_monitor_gain(sc, newai->monitor_gain); 6970 if (error) { 6971 device_printf(sc->sc_dev, 6972 "setting monitor_gain=%d failed with %d\n", 6973 newai->monitor_gain, error); 6974 goto abort; 6975 } 6976 } 6977 6978 /* XXX TODO */ 6979 /* sc->sc_ai = *ai; */ 6980 6981 error = 0; 6982 abort: 6983 return error; 6984 } 6985 6986 /* 6987 * Setup the hardware with mixer format phwfmt, rhwfmt. 6988 * The arguments have following restrictions: 6989 * - setmode is the direction you want to set, AUMODE_PLAY or AUMODE_RECORD, 6990 * or both. 6991 * - phwfmt and rhwfmt must not be NULL regardless of setmode. 6992 * - On non-independent devices, phwfmt and rhwfmt must have the same 6993 * parameters. 6994 * - pfil and rfil must be zero-filled. 6995 * If successful, 6996 * - pfil, rfil will be filled with filter information specified by the 6997 * hardware driver. 6998 * and then returns 0. Otherwise returns errno. 6999 * Must be called with sc_lock held. 7000 */ 7001 static int 7002 audio_hw_set_format(struct audio_softc *sc, int setmode, 7003 const audio_format2_t *phwfmt, const audio_format2_t *rhwfmt, 7004 audio_filter_reg_t *pfil, audio_filter_reg_t *rfil) 7005 { 7006 audio_params_t pp, rp; 7007 int error; 7008 7009 KASSERT(mutex_owned(sc->sc_lock)); 7010 KASSERT(phwfmt != NULL); 7011 KASSERT(rhwfmt != NULL); 7012 7013 pp = format2_to_params(phwfmt); 7014 rp = format2_to_params(rhwfmt); 7015 7016 error = sc->hw_if->set_format(sc->hw_hdl, setmode, 7017 &pp, &rp, pfil, rfil); 7018 if (error) { 7019 device_printf(sc->sc_dev, 7020 "set_format failed with %d\n", error); 7021 return error; 7022 } 7023 7024 if (sc->hw_if->commit_settings) { 7025 error = sc->hw_if->commit_settings(sc->hw_hdl); 7026 if (error) { 7027 device_printf(sc->sc_dev, 7028 "commit_settings failed with %d\n", error); 7029 return error; 7030 } 7031 } 7032 7033 return 0; 7034 } 7035 7036 /* 7037 * Fill audio_info structure. If need_mixerinfo is true, it will also 7038 * fill the hardware mixer information. 7039 * Must be called with sc_lock held. 7040 * Must be called with sc_exlock held, in addition, if need_mixerinfo is 7041 * true. 7042 */ 7043 static int 7044 audiogetinfo(struct audio_softc *sc, struct audio_info *ai, int need_mixerinfo, 7045 audio_file_t *file) 7046 { 7047 struct audio_prinfo *ri, *pi; 7048 audio_track_t *track; 7049 audio_track_t *ptrack; 7050 audio_track_t *rtrack; 7051 int gain; 7052 7053 KASSERT(mutex_owned(sc->sc_lock)); 7054 7055 ri = &ai->record; 7056 pi = &ai->play; 7057 ptrack = file->ptrack; 7058 rtrack = file->rtrack; 7059 7060 memset(ai, 0, sizeof(*ai)); 7061 7062 if (ptrack) { 7063 pi->sample_rate = ptrack->usrbuf.fmt.sample_rate; 7064 pi->channels = ptrack->usrbuf.fmt.channels; 7065 pi->precision = ptrack->usrbuf.fmt.precision; 7066 pi->encoding = ptrack->usrbuf.fmt.encoding; 7067 } else { 7068 /* Set default parameters if the track is not available. */ 7069 if (ISDEVAUDIO(file->dev)) { 7070 pi->sample_rate = audio_default.sample_rate; 7071 pi->channels = audio_default.channels; 7072 pi->precision = audio_default.precision; 7073 pi->encoding = audio_default.encoding; 7074 } else { 7075 pi->sample_rate = sc->sc_sound_pparams.sample_rate; 7076 pi->channels = sc->sc_sound_pparams.channels; 7077 pi->precision = sc->sc_sound_pparams.precision; 7078 pi->encoding = sc->sc_sound_pparams.encoding; 7079 } 7080 } 7081 if (rtrack) { 7082 ri->sample_rate = rtrack->usrbuf.fmt.sample_rate; 7083 ri->channels = rtrack->usrbuf.fmt.channels; 7084 ri->precision = rtrack->usrbuf.fmt.precision; 7085 ri->encoding = rtrack->usrbuf.fmt.encoding; 7086 } else { 7087 /* Set default parameters if the track is not available. */ 7088 if (ISDEVAUDIO(file->dev)) { 7089 ri->sample_rate = audio_default.sample_rate; 7090 ri->channels = audio_default.channels; 7091 ri->precision = audio_default.precision; 7092 ri->encoding = audio_default.encoding; 7093 } else { 7094 ri->sample_rate = sc->sc_sound_rparams.sample_rate; 7095 ri->channels = sc->sc_sound_rparams.channels; 7096 ri->precision = sc->sc_sound_rparams.precision; 7097 ri->encoding = sc->sc_sound_rparams.encoding; 7098 } 7099 } 7100 7101 if (ptrack) { 7102 pi->seek = ptrack->usrbuf.used; 7103 pi->samples = ptrack->usrbuf_stamp; 7104 pi->eof = ptrack->eofcounter; 7105 pi->pause = ptrack->is_pause; 7106 pi->error = (ptrack->dropframes != 0) ? 1 : 0; 7107 pi->waiting = 0; /* open never hangs */ 7108 pi->open = 1; 7109 pi->active = sc->sc_pbusy; 7110 pi->buffer_size = ptrack->usrbuf.capacity; 7111 } 7112 if (rtrack) { 7113 ri->seek = rtrack->usrbuf.used; 7114 ri->samples = rtrack->usrbuf_stamp; 7115 ri->eof = 0; 7116 ri->pause = rtrack->is_pause; 7117 ri->error = (rtrack->dropframes != 0) ? 1 : 0; 7118 ri->waiting = 0; /* open never hangs */ 7119 ri->open = 1; 7120 ri->active = sc->sc_rbusy; 7121 ri->buffer_size = rtrack->usrbuf.capacity; 7122 } 7123 7124 /* 7125 * XXX There may be different number of channels between playback 7126 * and recording, so that blocksize also may be different. 7127 * But struct audio_info has an united blocksize... 7128 * Here, I use play info precedencely if ptrack is available, 7129 * otherwise record info. 7130 * 7131 * XXX hiwat/lowat is a playback-only parameter. What should I 7132 * return for a record-only descriptor? 7133 */ 7134 track = ptrack ? ptrack : rtrack; 7135 if (track) { 7136 ai->blocksize = track->usrbuf_blksize; 7137 ai->hiwat = track->usrbuf_usedhigh / track->usrbuf_blksize; 7138 ai->lowat = track->usrbuf_usedlow / track->usrbuf_blksize; 7139 } 7140 ai->mode = file->mode; 7141 7142 if (need_mixerinfo) { 7143 KASSERT(sc->sc_exlock); 7144 7145 pi->port = au_get_port(sc, &sc->sc_outports); 7146 ri->port = au_get_port(sc, &sc->sc_inports); 7147 7148 pi->avail_ports = sc->sc_outports.allports; 7149 ri->avail_ports = sc->sc_inports.allports; 7150 7151 au_get_gain(sc, &sc->sc_outports, &pi->gain, &pi->balance); 7152 au_get_gain(sc, &sc->sc_inports, &ri->gain, &ri->balance); 7153 7154 if (sc->sc_monitor_port != -1) { 7155 gain = au_get_monitor_gain(sc); 7156 if (gain != -1) 7157 ai->monitor_gain = gain; 7158 } 7159 } 7160 7161 return 0; 7162 } 7163 7164 /* 7165 * Return true if playback is configured. 7166 * This function can be used after audioattach. 7167 */ 7168 static bool 7169 audio_can_playback(struct audio_softc *sc) 7170 { 7171 7172 return (sc->sc_pmixer != NULL); 7173 } 7174 7175 /* 7176 * Return true if recording is configured. 7177 * This function can be used after audioattach. 7178 */ 7179 static bool 7180 audio_can_capture(struct audio_softc *sc) 7181 { 7182 7183 return (sc->sc_rmixer != NULL); 7184 } 7185 7186 /* 7187 * Get the afp->index'th item from the valid one of format[]. 7188 * If found, stores it to afp->fmt and returns 0. Otherwise return EINVAL. 7189 * 7190 * This is common routines for query_format. 7191 * If your hardware driver has struct audio_format[], the simplest case 7192 * you can write your query_format interface as follows: 7193 * 7194 * struct audio_format foo_format[] = { ... }; 7195 * 7196 * int 7197 * foo_query_format(void *hdl, audio_format_query_t *afp) 7198 * { 7199 * return audio_query_format(foo_format, __arraycount(foo_format), afp); 7200 * } 7201 */ 7202 int 7203 audio_query_format(const struct audio_format *format, int nformats, 7204 audio_format_query_t *afp) 7205 { 7206 const struct audio_format *f; 7207 int idx; 7208 int i; 7209 7210 idx = 0; 7211 for (i = 0; i < nformats; i++) { 7212 f = &format[i]; 7213 if (!AUFMT_IS_VALID(f)) 7214 continue; 7215 if (afp->index == idx) { 7216 afp->fmt = *f; 7217 return 0; 7218 } 7219 idx++; 7220 } 7221 return EINVAL; 7222 } 7223 7224 /* 7225 * This function is provided for the hardware driver's set_format() to 7226 * find index matches with 'param' from array of audio_format_t 'formats'. 7227 * 'mode' is either of AUMODE_PLAY or AUMODE_RECORD. 7228 * It returns the matched index and never fails. Because param passed to 7229 * set_format() is selected from query_format(). 7230 * This function will be an alternative to auconv_set_converter() to 7231 * find index. 7232 */ 7233 int 7234 audio_indexof_format(const struct audio_format *formats, int nformats, 7235 int mode, const audio_params_t *param) 7236 { 7237 const struct audio_format *f; 7238 int index; 7239 int j; 7240 7241 for (index = 0; index < nformats; index++) { 7242 f = &formats[index]; 7243 7244 if (!AUFMT_IS_VALID(f)) 7245 continue; 7246 if ((f->mode & mode) == 0) 7247 continue; 7248 if (f->encoding != param->encoding) 7249 continue; 7250 if (f->validbits != param->precision) 7251 continue; 7252 if (f->channels != param->channels) 7253 continue; 7254 7255 if (f->frequency_type == 0) { 7256 if (param->sample_rate < f->frequency[0] || 7257 param->sample_rate > f->frequency[1]) 7258 continue; 7259 } else { 7260 for (j = 0; j < f->frequency_type; j++) { 7261 if (param->sample_rate == f->frequency[j]) 7262 break; 7263 } 7264 if (j == f->frequency_type) 7265 continue; 7266 } 7267 7268 /* Then, matched */ 7269 return index; 7270 } 7271 7272 /* Not matched. This should not be happened. */ 7273 panic("%s: cannot find matched format\n", __func__); 7274 } 7275 7276 /* 7277 * Get or set hardware blocksize in msec. 7278 * XXX It's for debug. 7279 */ 7280 static int 7281 audio_sysctl_blk_ms(SYSCTLFN_ARGS) 7282 { 7283 struct sysctlnode node; 7284 struct audio_softc *sc; 7285 audio_format2_t phwfmt; 7286 audio_format2_t rhwfmt; 7287 audio_filter_reg_t pfil; 7288 audio_filter_reg_t rfil; 7289 int t; 7290 int old_blk_ms; 7291 int mode; 7292 int error; 7293 7294 node = *rnode; 7295 sc = node.sysctl_data; 7296 7297 mutex_enter(sc->sc_lock); 7298 7299 old_blk_ms = sc->sc_blk_ms; 7300 t = old_blk_ms; 7301 node.sysctl_data = &t; 7302 error = sysctl_lookup(SYSCTLFN_CALL(&node)); 7303 if (error || newp == NULL) 7304 goto abort; 7305 7306 if (t < 0) { 7307 error = EINVAL; 7308 goto abort; 7309 } 7310 7311 if (sc->sc_popens + sc->sc_ropens > 0) { 7312 error = EBUSY; 7313 goto abort; 7314 } 7315 sc->sc_blk_ms = t; 7316 mode = 0; 7317 if (sc->sc_pmixer) { 7318 mode |= AUMODE_PLAY; 7319 phwfmt = sc->sc_pmixer->hwbuf.fmt; 7320 } 7321 if (sc->sc_rmixer) { 7322 mode |= AUMODE_RECORD; 7323 rhwfmt = sc->sc_rmixer->hwbuf.fmt; 7324 } 7325 7326 /* re-init hardware */ 7327 memset(&pfil, 0, sizeof(pfil)); 7328 memset(&rfil, 0, sizeof(rfil)); 7329 error = audio_hw_set_format(sc, mode, &phwfmt, &rhwfmt, &pfil, &rfil); 7330 if (error) { 7331 goto abort; 7332 } 7333 7334 /* re-init track mixer */ 7335 error = audio_mixers_init(sc, mode, &phwfmt, &rhwfmt, &pfil, &rfil); 7336 if (error) { 7337 /* Rollback */ 7338 sc->sc_blk_ms = old_blk_ms; 7339 audio_mixers_init(sc, mode, &phwfmt, &rhwfmt, &pfil, &rfil); 7340 goto abort; 7341 } 7342 error = 0; 7343 abort: 7344 mutex_exit(sc->sc_lock); 7345 return error; 7346 } 7347 7348 /* 7349 * Get or set multiuser mode. 7350 */ 7351 static int 7352 audio_sysctl_multiuser(SYSCTLFN_ARGS) 7353 { 7354 struct sysctlnode node; 7355 struct audio_softc *sc; 7356 bool t; 7357 int error; 7358 7359 node = *rnode; 7360 sc = node.sysctl_data; 7361 7362 mutex_enter(sc->sc_lock); 7363 7364 t = sc->sc_multiuser; 7365 node.sysctl_data = &t; 7366 error = sysctl_lookup(SYSCTLFN_CALL(&node)); 7367 if (error || newp == NULL) 7368 goto abort; 7369 7370 sc->sc_multiuser = t; 7371 error = 0; 7372 abort: 7373 mutex_exit(sc->sc_lock); 7374 return error; 7375 } 7376 7377 #if defined(AUDIO_DEBUG) 7378 /* 7379 * Get or set debug verbose level. (0..4) 7380 * XXX It's for debug. 7381 * XXX It is not separated per device. 7382 */ 7383 static int 7384 audio_sysctl_debug(SYSCTLFN_ARGS) 7385 { 7386 struct sysctlnode node; 7387 int t; 7388 int error; 7389 7390 node = *rnode; 7391 t = audiodebug; 7392 node.sysctl_data = &t; 7393 error = sysctl_lookup(SYSCTLFN_CALL(&node)); 7394 if (error || newp == NULL) 7395 return error; 7396 7397 if (t < 0 || t > 4) 7398 return EINVAL; 7399 audiodebug = t; 7400 printf("audio: audiodebug = %d\n", audiodebug); 7401 return 0; 7402 } 7403 #endif /* AUDIO_DEBUG */ 7404 7405 #ifdef AUDIO_PM_IDLE 7406 static void 7407 audio_idle(void *arg) 7408 { 7409 device_t dv = arg; 7410 struct audio_softc *sc = device_private(dv); 7411 7412 #ifdef PNP_DEBUG 7413 extern int pnp_debug_idle; 7414 if (pnp_debug_idle) 7415 printf("%s: idle handler called\n", device_xname(dv)); 7416 #endif 7417 7418 sc->sc_idle = true; 7419 7420 /* XXX joerg Make pmf_device_suspend handle children? */ 7421 if (!pmf_device_suspend(dv, PMF_Q_SELF)) 7422 return; 7423 7424 if (!pmf_device_suspend(sc->hw_dev, PMF_Q_SELF)) 7425 pmf_device_resume(dv, PMF_Q_SELF); 7426 } 7427 7428 static void 7429 audio_activity(device_t dv, devactive_t type) 7430 { 7431 struct audio_softc *sc = device_private(dv); 7432 7433 if (type != DVA_SYSTEM) 7434 return; 7435 7436 callout_schedule(&sc->sc_idle_counter, audio_idle_timeout * hz); 7437 7438 sc->sc_idle = false; 7439 if (!device_is_active(dv)) { 7440 /* XXX joerg How to deal with a failing resume... */ 7441 pmf_device_resume(sc->hw_dev, PMF_Q_SELF); 7442 pmf_device_resume(dv, PMF_Q_SELF); 7443 } 7444 } 7445 #endif 7446 7447 static bool 7448 audio_suspend(device_t dv, const pmf_qual_t *qual) 7449 { 7450 struct audio_softc *sc = device_private(dv); 7451 int error; 7452 7453 error = audio_enter_exclusive(sc); 7454 if (error) 7455 return error; 7456 audio_mixer_capture(sc); 7457 7458 /* Halts mixers but don't clear busy flag for resume */ 7459 if (sc->sc_pbusy) { 7460 audio_pmixer_halt(sc); 7461 sc->sc_pbusy = true; 7462 } 7463 if (sc->sc_rbusy) { 7464 audio_rmixer_halt(sc); 7465 sc->sc_rbusy = true; 7466 } 7467 7468 #ifdef AUDIO_PM_IDLE 7469 callout_halt(&sc->sc_idle_counter, sc->sc_lock); 7470 #endif 7471 audio_exit_exclusive(sc); 7472 7473 return true; 7474 } 7475 7476 static bool 7477 audio_resume(device_t dv, const pmf_qual_t *qual) 7478 { 7479 struct audio_softc *sc = device_private(dv); 7480 struct audio_info ai; 7481 int error; 7482 7483 error = audio_enter_exclusive(sc); 7484 if (error) 7485 return error; 7486 7487 audio_mixer_restore(sc); 7488 /* XXX ? */ 7489 AUDIO_INITINFO(&ai); 7490 audio_hw_setinfo(sc, &ai, NULL); 7491 7492 if (sc->sc_pbusy) 7493 audio_pmixer_start(sc, true); 7494 if (sc->sc_rbusy) 7495 audio_rmixer_start(sc); 7496 7497 audio_exit_exclusive(sc); 7498 7499 return true; 7500 } 7501 7502 #if defined(AUDIO_DEBUG) 7503 static void 7504 audio_format2_tostr(char *buf, size_t bufsize, const audio_format2_t *fmt) 7505 { 7506 int n; 7507 7508 n = 0; 7509 n += snprintf(buf + n, bufsize - n, "%s", 7510 audio_encoding_name(fmt->encoding)); 7511 if (fmt->precision == fmt->stride) { 7512 n += snprintf(buf + n, bufsize - n, " %dbit", fmt->precision); 7513 } else { 7514 n += snprintf(buf + n, bufsize - n, " %d/%dbit", 7515 fmt->precision, fmt->stride); 7516 } 7517 7518 snprintf(buf + n, bufsize - n, " %uch %uHz", 7519 fmt->channels, fmt->sample_rate); 7520 } 7521 #endif 7522 7523 #if defined(AUDIO_DEBUG) 7524 static void 7525 audio_print_format2(const char *s, const audio_format2_t *fmt) 7526 { 7527 char fmtstr[64]; 7528 7529 audio_format2_tostr(fmtstr, sizeof(fmtstr), fmt); 7530 printf("%s %s\n", s, fmtstr); 7531 } 7532 #endif 7533 7534 #ifdef DIAGNOSTIC 7535 void 7536 audio_diagnostic_format2(const char *func, const audio_format2_t *fmt) 7537 { 7538 7539 KASSERTMSG(fmt, "%s: fmt == NULL", func); 7540 7541 /* XXX MSM6258 vs(4) only has 4bit stride format. */ 7542 if (fmt->encoding == AUDIO_ENCODING_ADPCM) { 7543 KASSERTMSG(fmt->stride == 4 || fmt->stride == 8, 7544 "%s: stride(%d) is invalid", func, fmt->stride); 7545 } else { 7546 KASSERTMSG(fmt->stride % NBBY == 0, 7547 "%s: stride(%d) is invalid", func, fmt->stride); 7548 } 7549 KASSERTMSG(fmt->precision <= fmt->stride, 7550 "%s: precision(%d) <= stride(%d)", 7551 func, fmt->precision, fmt->stride); 7552 KASSERTMSG(1 <= fmt->channels && fmt->channels <= AUDIO_MAX_CHANNELS, 7553 "%s: channels(%d) is out of range", 7554 func, fmt->channels); 7555 7556 /* XXX No check for encodings? */ 7557 } 7558 7559 void 7560 audio_diagnostic_filter_arg(const char *func, const audio_filter_arg_t *arg) 7561 { 7562 7563 KASSERT(arg != NULL); 7564 KASSERT(arg->src != NULL); 7565 KASSERT(arg->dst != NULL); 7566 DIAGNOSTIC_format2(arg->srcfmt); 7567 DIAGNOSTIC_format2(arg->dstfmt); 7568 KASSERTMSG(arg->count > 0, 7569 "%s: count(%d) is out of range", func, arg->count); 7570 } 7571 7572 void 7573 audio_diagnostic_ring(const char *func, const audio_ring_t *ring) 7574 { 7575 7576 KASSERTMSG(ring, "%s: ring == NULL", func); 7577 DIAGNOSTIC_format2(&ring->fmt); 7578 KASSERTMSG(0 <= ring->capacity && ring->capacity < INT_MAX / 2, 7579 "%s: capacity(%d) is out of range", func, ring->capacity); 7580 KASSERTMSG(0 <= ring->used && ring->used <= ring->capacity, 7581 "%s: used(%d) is out of range (capacity:%d)", 7582 func, ring->used, ring->capacity); 7583 if (ring->capacity == 0) { 7584 KASSERTMSG(ring->mem == NULL, 7585 "%s: capacity == 0 but mem != NULL", func); 7586 } else { 7587 KASSERTMSG(ring->mem != NULL, 7588 "%s: capacity != 0 but mem == NULL", func); 7589 KASSERTMSG(0 <= ring->head && ring->head < ring->capacity, 7590 "%s: head(%d) is out of range (capacity:%d)", 7591 func, ring->head, ring->capacity); 7592 } 7593 } 7594 #endif /* DIAGNOSTIC */ 7595 7596 7597 /* 7598 * Mixer driver 7599 */ 7600 int 7601 mixer_open(dev_t dev, struct audio_softc *sc, int flags, int ifmt, 7602 struct lwp *l) 7603 { 7604 struct file *fp; 7605 audio_file_t *af; 7606 int error, fd; 7607 7608 KASSERT(mutex_owned(sc->sc_lock)); 7609 7610 TRACE(1, "flags=0x%x", flags); 7611 7612 error = fd_allocfile(&fp, &fd); 7613 if (error) 7614 return error; 7615 7616 af = kmem_zalloc(sizeof(*af), KM_SLEEP); 7617 af->sc = sc; 7618 af->dev = dev; 7619 7620 error = fd_clone(fp, fd, flags, &audio_fileops, af); 7621 KASSERT(error == EMOVEFD); 7622 7623 return error; 7624 } 7625 7626 /* 7627 * Add a process to those to be signalled on mixer activity. 7628 * If the process has already been added, do nothing. 7629 * Must be called with sc_lock held. 7630 */ 7631 static void 7632 mixer_async_add(struct audio_softc *sc, pid_t pid) 7633 { 7634 int i; 7635 7636 KASSERT(mutex_owned(sc->sc_lock)); 7637 7638 /* If already exists, returns without doing anything. */ 7639 for (i = 0; i < sc->sc_am_used; i++) { 7640 if (sc->sc_am[i] == pid) 7641 return; 7642 } 7643 7644 /* Extend array if necessary. */ 7645 if (sc->sc_am_used >= sc->sc_am_capacity) { 7646 sc->sc_am_capacity += AM_CAPACITY; 7647 sc->sc_am = kern_realloc(sc->sc_am, 7648 sc->sc_am_capacity * sizeof(pid_t), M_WAITOK); 7649 TRACE(2, "realloc am_capacity=%d", sc->sc_am_capacity); 7650 } 7651 7652 TRACE(2, "am[%d]=%d", sc->sc_am_used, (int)pid); 7653 sc->sc_am[sc->sc_am_used++] = pid; 7654 } 7655 7656 /* 7657 * Remove a process from those to be signalled on mixer activity. 7658 * If the process has not been added, do nothing. 7659 * Must be called with sc_lock held. 7660 */ 7661 static void 7662 mixer_async_remove(struct audio_softc *sc, pid_t pid) 7663 { 7664 int i; 7665 7666 KASSERT(mutex_owned(sc->sc_lock)); 7667 7668 for (i = 0; i < sc->sc_am_used; i++) { 7669 if (sc->sc_am[i] == pid) { 7670 sc->sc_am[i] = sc->sc_am[--sc->sc_am_used]; 7671 TRACE(2, "am[%d](%d) removed, used=%d", 7672 i, (int)pid, sc->sc_am_used); 7673 7674 /* Empty array if no longer necessary. */ 7675 if (sc->sc_am_used == 0) { 7676 kern_free(sc->sc_am); 7677 sc->sc_am = NULL; 7678 sc->sc_am_capacity = 0; 7679 TRACE(2, "released"); 7680 } 7681 return; 7682 } 7683 } 7684 } 7685 7686 /* 7687 * Signal all processes waiting for the mixer. 7688 * Must be called with sc_lock held. 7689 */ 7690 static void 7691 mixer_signal(struct audio_softc *sc) 7692 { 7693 proc_t *p; 7694 int i; 7695 7696 KASSERT(mutex_owned(sc->sc_lock)); 7697 7698 for (i = 0; i < sc->sc_am_used; i++) { 7699 mutex_enter(proc_lock); 7700 p = proc_find(sc->sc_am[i]); 7701 if (p) 7702 psignal(p, SIGIO); 7703 mutex_exit(proc_lock); 7704 } 7705 } 7706 7707 /* 7708 * Close a mixer device 7709 */ 7710 int 7711 mixer_close(struct audio_softc *sc, audio_file_t *file) 7712 { 7713 7714 mutex_enter(sc->sc_lock); 7715 TRACE(1, ""); 7716 mixer_async_remove(sc, curproc->p_pid); 7717 mutex_exit(sc->sc_lock); 7718 7719 kmem_free(file, sizeof(*file)); 7720 return 0; 7721 } 7722 7723 /* 7724 * Must be called without sc_lock nor sc_exlock held. 7725 */ 7726 int 7727 mixer_ioctl(struct audio_softc *sc, u_long cmd, void *addr, int flag, 7728 struct lwp *l) 7729 { 7730 mixer_devinfo_t *mi; 7731 mixer_ctrl_t *mc; 7732 int error; 7733 7734 TRACE(2, "(%lu,'%c',%lu)", 7735 IOCPARM_LEN(cmd), (char)IOCGROUP(cmd), cmd & 0xff); 7736 error = EINVAL; 7737 7738 /* we can return cached values if we are sleeping */ 7739 if (cmd != AUDIO_MIXER_READ) { 7740 mutex_enter(sc->sc_lock); 7741 device_active(sc->sc_dev, DVA_SYSTEM); 7742 mutex_exit(sc->sc_lock); 7743 } 7744 7745 switch (cmd) { 7746 case FIOASYNC: 7747 mutex_enter(sc->sc_lock); 7748 if (*(int *)addr) { 7749 mixer_async_add(sc, curproc->p_pid); 7750 } else { 7751 mixer_async_remove(sc, curproc->p_pid); 7752 } 7753 mutex_exit(sc->sc_lock); 7754 error = 0; 7755 break; 7756 7757 case AUDIO_GETDEV: 7758 TRACE(2, "AUDIO_GETDEV"); 7759 error = audio_enter_exclusive(sc); 7760 if (error) 7761 break; 7762 error = sc->hw_if->getdev(sc->hw_hdl, (audio_device_t *)addr); 7763 audio_exit_exclusive(sc); 7764 break; 7765 7766 case AUDIO_MIXER_DEVINFO: 7767 TRACE(2, "AUDIO_MIXER_DEVINFO"); 7768 mi = (mixer_devinfo_t *)addr; 7769 7770 mi->un.v.delta = 0; /* default */ 7771 mutex_enter(sc->sc_lock); 7772 error = audio_query_devinfo(sc, mi); 7773 mutex_exit(sc->sc_lock); 7774 break; 7775 7776 case AUDIO_MIXER_READ: 7777 TRACE(2, "AUDIO_MIXER_READ"); 7778 mc = (mixer_ctrl_t *)addr; 7779 7780 error = audio_enter_exclusive(sc); 7781 if (error) 7782 break; 7783 if (device_is_active(sc->hw_dev)) 7784 error = audio_get_port(sc, mc); 7785 else if (mc->dev < 0 || mc->dev >= sc->sc_nmixer_states) 7786 error = ENXIO; 7787 else { 7788 int dev = mc->dev; 7789 memcpy(mc, &sc->sc_mixer_state[dev], 7790 sizeof(mixer_ctrl_t)); 7791 error = 0; 7792 } 7793 audio_exit_exclusive(sc); 7794 break; 7795 7796 case AUDIO_MIXER_WRITE: 7797 TRACE(2, "AUDIO_MIXER_WRITE"); 7798 error = audio_enter_exclusive(sc); 7799 if (error) 7800 break; 7801 error = audio_set_port(sc, (mixer_ctrl_t *)addr); 7802 if (error) { 7803 audio_exit_exclusive(sc); 7804 break; 7805 } 7806 7807 if (sc->hw_if->commit_settings) { 7808 error = sc->hw_if->commit_settings(sc->hw_hdl); 7809 if (error) { 7810 audio_exit_exclusive(sc); 7811 break; 7812 } 7813 } 7814 mixer_signal(sc); 7815 audio_exit_exclusive(sc); 7816 break; 7817 7818 default: 7819 if (sc->hw_if->dev_ioctl) { 7820 error = audio_enter_exclusive(sc); 7821 if (error) 7822 break; 7823 error = sc->hw_if->dev_ioctl(sc->hw_hdl, 7824 cmd, addr, flag, l); 7825 audio_exit_exclusive(sc); 7826 } else 7827 error = EINVAL; 7828 break; 7829 } 7830 TRACE(2, "(%lu,'%c',%lu) result %d", 7831 IOCPARM_LEN(cmd), (char)IOCGROUP(cmd), cmd & 0xff, error); 7832 return error; 7833 } 7834 7835 /* 7836 * Must be called with sc_lock held. 7837 */ 7838 int 7839 au_portof(struct audio_softc *sc, char *name, int class) 7840 { 7841 mixer_devinfo_t mi; 7842 7843 KASSERT(mutex_owned(sc->sc_lock)); 7844 7845 for (mi.index = 0; audio_query_devinfo(sc, &mi) == 0; mi.index++) { 7846 if (mi.mixer_class == class && strcmp(mi.label.name, name) == 0) 7847 return mi.index; 7848 } 7849 return -1; 7850 } 7851 7852 /* 7853 * Must be called with sc_lock held. 7854 */ 7855 void 7856 au_setup_ports(struct audio_softc *sc, struct au_mixer_ports *ports, 7857 mixer_devinfo_t *mi, const struct portname *tbl) 7858 { 7859 int i, j; 7860 7861 KASSERT(mutex_owned(sc->sc_lock)); 7862 7863 ports->index = mi->index; 7864 if (mi->type == AUDIO_MIXER_ENUM) { 7865 ports->isenum = true; 7866 for(i = 0; tbl[i].name; i++) 7867 for(j = 0; j < mi->un.e.num_mem; j++) 7868 if (strcmp(mi->un.e.member[j].label.name, 7869 tbl[i].name) == 0) { 7870 ports->allports |= tbl[i].mask; 7871 ports->aumask[ports->nports] = tbl[i].mask; 7872 ports->misel[ports->nports] = 7873 mi->un.e.member[j].ord; 7874 ports->miport[ports->nports] = 7875 au_portof(sc, mi->un.e.member[j].label.name, 7876 mi->mixer_class); 7877 if (ports->mixerout != -1 && 7878 ports->miport[ports->nports] != -1) 7879 ports->isdual = true; 7880 ++ports->nports; 7881 } 7882 } else if (mi->type == AUDIO_MIXER_SET) { 7883 for(i = 0; tbl[i].name; i++) 7884 for(j = 0; j < mi->un.s.num_mem; j++) 7885 if (strcmp(mi->un.s.member[j].label.name, 7886 tbl[i].name) == 0) { 7887 ports->allports |= tbl[i].mask; 7888 ports->aumask[ports->nports] = tbl[i].mask; 7889 ports->misel[ports->nports] = 7890 mi->un.s.member[j].mask; 7891 ports->miport[ports->nports] = 7892 au_portof(sc, mi->un.s.member[j].label.name, 7893 mi->mixer_class); 7894 ++ports->nports; 7895 } 7896 } 7897 } 7898 7899 /* 7900 * Must be called with sc_lock && sc_exlock held. 7901 */ 7902 int 7903 au_set_lr_value(struct audio_softc *sc, mixer_ctrl_t *ct, int l, int r) 7904 { 7905 7906 KASSERT(mutex_owned(sc->sc_lock)); 7907 KASSERT(sc->sc_exlock); 7908 7909 ct->type = AUDIO_MIXER_VALUE; 7910 ct->un.value.num_channels = 2; 7911 ct->un.value.level[AUDIO_MIXER_LEVEL_LEFT] = l; 7912 ct->un.value.level[AUDIO_MIXER_LEVEL_RIGHT] = r; 7913 if (audio_set_port(sc, ct) == 0) 7914 return 0; 7915 ct->un.value.num_channels = 1; 7916 ct->un.value.level[AUDIO_MIXER_LEVEL_MONO] = (l+r)/2; 7917 return audio_set_port(sc, ct); 7918 } 7919 7920 /* 7921 * Must be called with sc_lock && sc_exlock held. 7922 */ 7923 int 7924 au_get_lr_value(struct audio_softc *sc, mixer_ctrl_t *ct, int *l, int *r) 7925 { 7926 int error; 7927 7928 KASSERT(mutex_owned(sc->sc_lock)); 7929 KASSERT(sc->sc_exlock); 7930 7931 ct->un.value.num_channels = 2; 7932 if (audio_get_port(sc, ct) == 0) { 7933 *l = ct->un.value.level[AUDIO_MIXER_LEVEL_LEFT]; 7934 *r = ct->un.value.level[AUDIO_MIXER_LEVEL_RIGHT]; 7935 } else { 7936 ct->un.value.num_channels = 1; 7937 error = audio_get_port(sc, ct); 7938 if (error) 7939 return error; 7940 *r = *l = ct->un.value.level[AUDIO_MIXER_LEVEL_MONO]; 7941 } 7942 return 0; 7943 } 7944 7945 /* 7946 * Must be called with sc_lock && sc_exlock held. 7947 */ 7948 int 7949 au_set_gain(struct audio_softc *sc, struct au_mixer_ports *ports, 7950 int gain, int balance) 7951 { 7952 mixer_ctrl_t ct; 7953 int i, error; 7954 int l, r; 7955 u_int mask; 7956 int nset; 7957 7958 KASSERT(mutex_owned(sc->sc_lock)); 7959 KASSERT(sc->sc_exlock); 7960 7961 if (balance == AUDIO_MID_BALANCE) { 7962 l = r = gain; 7963 } else if (balance < AUDIO_MID_BALANCE) { 7964 l = gain; 7965 r = (balance * gain) / AUDIO_MID_BALANCE; 7966 } else { 7967 r = gain; 7968 l = ((AUDIO_RIGHT_BALANCE - balance) * gain) 7969 / AUDIO_MID_BALANCE; 7970 } 7971 TRACE(2, "gain=%d balance=%d, l=%d r=%d", gain, balance, l, r); 7972 7973 if (ports->index == -1) { 7974 usemaster: 7975 if (ports->master == -1) 7976 return 0; /* just ignore it silently */ 7977 ct.dev = ports->master; 7978 error = au_set_lr_value(sc, &ct, l, r); 7979 } else { 7980 ct.dev = ports->index; 7981 if (ports->isenum) { 7982 ct.type = AUDIO_MIXER_ENUM; 7983 error = audio_get_port(sc, &ct); 7984 if (error) 7985 return error; 7986 if (ports->isdual) { 7987 if (ports->cur_port == -1) 7988 ct.dev = ports->master; 7989 else 7990 ct.dev = ports->miport[ports->cur_port]; 7991 error = au_set_lr_value(sc, &ct, l, r); 7992 } else { 7993 for(i = 0; i < ports->nports; i++) 7994 if (ports->misel[i] == ct.un.ord) { 7995 ct.dev = ports->miport[i]; 7996 if (ct.dev == -1 || 7997 au_set_lr_value(sc, &ct, l, r)) 7998 goto usemaster; 7999 else 8000 break; 8001 } 8002 } 8003 } else { 8004 ct.type = AUDIO_MIXER_SET; 8005 error = audio_get_port(sc, &ct); 8006 if (error) 8007 return error; 8008 mask = ct.un.mask; 8009 nset = 0; 8010 for(i = 0; i < ports->nports; i++) { 8011 if (ports->misel[i] & mask) { 8012 ct.dev = ports->miport[i]; 8013 if (ct.dev != -1 && 8014 au_set_lr_value(sc, &ct, l, r) == 0) 8015 nset++; 8016 } 8017 } 8018 if (nset == 0) 8019 goto usemaster; 8020 } 8021 } 8022 if (!error) 8023 mixer_signal(sc); 8024 return error; 8025 } 8026 8027 /* 8028 * Must be called with sc_lock && sc_exlock held. 8029 */ 8030 void 8031 au_get_gain(struct audio_softc *sc, struct au_mixer_ports *ports, 8032 u_int *pgain, u_char *pbalance) 8033 { 8034 mixer_ctrl_t ct; 8035 int i, l, r, n; 8036 int lgain, rgain; 8037 8038 KASSERT(mutex_owned(sc->sc_lock)); 8039 KASSERT(sc->sc_exlock); 8040 8041 lgain = AUDIO_MAX_GAIN / 2; 8042 rgain = AUDIO_MAX_GAIN / 2; 8043 if (ports->index == -1) { 8044 usemaster: 8045 if (ports->master == -1) 8046 goto bad; 8047 ct.dev = ports->master; 8048 ct.type = AUDIO_MIXER_VALUE; 8049 if (au_get_lr_value(sc, &ct, &lgain, &rgain)) 8050 goto bad; 8051 } else { 8052 ct.dev = ports->index; 8053 if (ports->isenum) { 8054 ct.type = AUDIO_MIXER_ENUM; 8055 if (audio_get_port(sc, &ct)) 8056 goto bad; 8057 ct.type = AUDIO_MIXER_VALUE; 8058 if (ports->isdual) { 8059 if (ports->cur_port == -1) 8060 ct.dev = ports->master; 8061 else 8062 ct.dev = ports->miport[ports->cur_port]; 8063 au_get_lr_value(sc, &ct, &lgain, &rgain); 8064 } else { 8065 for(i = 0; i < ports->nports; i++) 8066 if (ports->misel[i] == ct.un.ord) { 8067 ct.dev = ports->miport[i]; 8068 if (ct.dev == -1 || 8069 au_get_lr_value(sc, &ct, 8070 &lgain, &rgain)) 8071 goto usemaster; 8072 else 8073 break; 8074 } 8075 } 8076 } else { 8077 ct.type = AUDIO_MIXER_SET; 8078 if (audio_get_port(sc, &ct)) 8079 goto bad; 8080 ct.type = AUDIO_MIXER_VALUE; 8081 lgain = rgain = n = 0; 8082 for(i = 0; i < ports->nports; i++) { 8083 if (ports->misel[i] & ct.un.mask) { 8084 ct.dev = ports->miport[i]; 8085 if (ct.dev == -1 || 8086 au_get_lr_value(sc, &ct, &l, &r)) 8087 goto usemaster; 8088 else { 8089 lgain += l; 8090 rgain += r; 8091 n++; 8092 } 8093 } 8094 } 8095 if (n != 0) { 8096 lgain /= n; 8097 rgain /= n; 8098 } 8099 } 8100 } 8101 bad: 8102 if (lgain == rgain) { /* handles lgain==rgain==0 */ 8103 *pgain = lgain; 8104 *pbalance = AUDIO_MID_BALANCE; 8105 } else if (lgain < rgain) { 8106 *pgain = rgain; 8107 /* balance should be > AUDIO_MID_BALANCE */ 8108 *pbalance = AUDIO_RIGHT_BALANCE - 8109 (AUDIO_MID_BALANCE * lgain) / rgain; 8110 } else /* lgain > rgain */ { 8111 *pgain = lgain; 8112 /* balance should be < AUDIO_MID_BALANCE */ 8113 *pbalance = (AUDIO_MID_BALANCE * rgain) / lgain; 8114 } 8115 } 8116 8117 /* 8118 * Must be called with sc_lock && sc_exlock held. 8119 */ 8120 int 8121 au_set_port(struct audio_softc *sc, struct au_mixer_ports *ports, u_int port) 8122 { 8123 mixer_ctrl_t ct; 8124 int i, error, use_mixerout; 8125 8126 KASSERT(mutex_owned(sc->sc_lock)); 8127 KASSERT(sc->sc_exlock); 8128 8129 use_mixerout = 1; 8130 if (port == 0) { 8131 if (ports->allports == 0) 8132 return 0; /* Allow this special case. */ 8133 else if (ports->isdual) { 8134 if (ports->cur_port == -1) { 8135 return 0; 8136 } else { 8137 port = ports->aumask[ports->cur_port]; 8138 ports->cur_port = -1; 8139 use_mixerout = 0; 8140 } 8141 } 8142 } 8143 if (ports->index == -1) 8144 return EINVAL; 8145 ct.dev = ports->index; 8146 if (ports->isenum) { 8147 if (port & (port-1)) 8148 return EINVAL; /* Only one port allowed */ 8149 ct.type = AUDIO_MIXER_ENUM; 8150 error = EINVAL; 8151 for(i = 0; i < ports->nports; i++) 8152 if (ports->aumask[i] == port) { 8153 if (ports->isdual && use_mixerout) { 8154 ct.un.ord = ports->mixerout; 8155 ports->cur_port = i; 8156 } else { 8157 ct.un.ord = ports->misel[i]; 8158 } 8159 error = audio_set_port(sc, &ct); 8160 break; 8161 } 8162 } else { 8163 ct.type = AUDIO_MIXER_SET; 8164 ct.un.mask = 0; 8165 for(i = 0; i < ports->nports; i++) 8166 if (ports->aumask[i] & port) 8167 ct.un.mask |= ports->misel[i]; 8168 if (port != 0 && ct.un.mask == 0) 8169 error = EINVAL; 8170 else 8171 error = audio_set_port(sc, &ct); 8172 } 8173 if (!error) 8174 mixer_signal(sc); 8175 return error; 8176 } 8177 8178 /* 8179 * Must be called with sc_lock && sc_exlock held. 8180 */ 8181 int 8182 au_get_port(struct audio_softc *sc, struct au_mixer_ports *ports) 8183 { 8184 mixer_ctrl_t ct; 8185 int i, aumask; 8186 8187 KASSERT(mutex_owned(sc->sc_lock)); 8188 KASSERT(sc->sc_exlock); 8189 8190 if (ports->index == -1) 8191 return 0; 8192 ct.dev = ports->index; 8193 ct.type = ports->isenum ? AUDIO_MIXER_ENUM : AUDIO_MIXER_SET; 8194 if (audio_get_port(sc, &ct)) 8195 return 0; 8196 aumask = 0; 8197 if (ports->isenum) { 8198 if (ports->isdual && ports->cur_port != -1) { 8199 if (ports->mixerout == ct.un.ord) 8200 aumask = ports->aumask[ports->cur_port]; 8201 else 8202 ports->cur_port = -1; 8203 } 8204 if (aumask == 0) 8205 for(i = 0; i < ports->nports; i++) 8206 if (ports->misel[i] == ct.un.ord) 8207 aumask = ports->aumask[i]; 8208 } else { 8209 for(i = 0; i < ports->nports; i++) 8210 if (ct.un.mask & ports->misel[i]) 8211 aumask |= ports->aumask[i]; 8212 } 8213 return aumask; 8214 } 8215 8216 /* 8217 * It returns 0 if success, otherwise errno. 8218 * Must be called only if sc->sc_monitor_port != -1. 8219 * Must be called with sc_lock && sc_exlock held. 8220 */ 8221 static int 8222 au_set_monitor_gain(struct audio_softc *sc, int monitor_gain) 8223 { 8224 mixer_ctrl_t ct; 8225 8226 KASSERT(mutex_owned(sc->sc_lock)); 8227 KASSERT(sc->sc_exlock); 8228 8229 ct.dev = sc->sc_monitor_port; 8230 ct.type = AUDIO_MIXER_VALUE; 8231 ct.un.value.num_channels = 1; 8232 ct.un.value.level[AUDIO_MIXER_LEVEL_MONO] = monitor_gain; 8233 return audio_set_port(sc, &ct); 8234 } 8235 8236 /* 8237 * It returns monitor gain if success, otherwise -1. 8238 * Must be called only if sc->sc_monitor_port != -1. 8239 * Must be called with sc_lock && sc_exlock held. 8240 */ 8241 static int 8242 au_get_monitor_gain(struct audio_softc *sc) 8243 { 8244 mixer_ctrl_t ct; 8245 8246 KASSERT(mutex_owned(sc->sc_lock)); 8247 KASSERT(sc->sc_exlock); 8248 8249 ct.dev = sc->sc_monitor_port; 8250 ct.type = AUDIO_MIXER_VALUE; 8251 ct.un.value.num_channels = 1; 8252 if (audio_get_port(sc, &ct)) 8253 return -1; 8254 return ct.un.value.level[AUDIO_MIXER_LEVEL_MONO]; 8255 } 8256 8257 /* 8258 * Must be called with sc_lock && sc_exlock held. 8259 */ 8260 static int 8261 audio_set_port(struct audio_softc *sc, mixer_ctrl_t *mc) 8262 { 8263 8264 KASSERT(mutex_owned(sc->sc_lock)); 8265 KASSERT(sc->sc_exlock); 8266 8267 return sc->hw_if->set_port(sc->hw_hdl, mc); 8268 } 8269 8270 /* 8271 * Must be called with sc_lock && sc_exlock held. 8272 */ 8273 static int 8274 audio_get_port(struct audio_softc *sc, mixer_ctrl_t *mc) 8275 { 8276 8277 KASSERT(mutex_owned(sc->sc_lock)); 8278 KASSERT(sc->sc_exlock); 8279 8280 return sc->hw_if->get_port(sc->hw_hdl, mc); 8281 } 8282 8283 /* 8284 * Must be called with sc_lock && sc_exlock held. 8285 */ 8286 static void 8287 audio_mixer_capture(struct audio_softc *sc) 8288 { 8289 mixer_devinfo_t mi; 8290 mixer_ctrl_t *mc; 8291 8292 KASSERT(mutex_owned(sc->sc_lock)); 8293 KASSERT(sc->sc_exlock); 8294 8295 for (mi.index = 0;; mi.index++) { 8296 if (audio_query_devinfo(sc, &mi) != 0) 8297 break; 8298 KASSERT(mi.index < sc->sc_nmixer_states); 8299 if (mi.type == AUDIO_MIXER_CLASS) 8300 continue; 8301 mc = &sc->sc_mixer_state[mi.index]; 8302 mc->dev = mi.index; 8303 mc->type = mi.type; 8304 mc->un.value.num_channels = mi.un.v.num_channels; 8305 (void)audio_get_port(sc, mc); 8306 } 8307 8308 return; 8309 } 8310 8311 /* 8312 * Must be called with sc_lock && sc_exlock held. 8313 */ 8314 static void 8315 audio_mixer_restore(struct audio_softc *sc) 8316 { 8317 mixer_devinfo_t mi; 8318 mixer_ctrl_t *mc; 8319 8320 KASSERT(mutex_owned(sc->sc_lock)); 8321 KASSERT(sc->sc_exlock); 8322 8323 for (mi.index = 0; ; mi.index++) { 8324 if (audio_query_devinfo(sc, &mi) != 0) 8325 break; 8326 if (mi.type == AUDIO_MIXER_CLASS) 8327 continue; 8328 mc = &sc->sc_mixer_state[mi.index]; 8329 (void)audio_set_port(sc, mc); 8330 } 8331 if (sc->hw_if->commit_settings) 8332 sc->hw_if->commit_settings(sc->hw_hdl); 8333 8334 return; 8335 } 8336 8337 static void 8338 audio_volume_down(device_t dv) 8339 { 8340 struct audio_softc *sc = device_private(dv); 8341 mixer_devinfo_t mi; 8342 int newgain; 8343 u_int gain; 8344 u_char balance; 8345 8346 if (audio_enter_exclusive(sc) != 0) 8347 return; 8348 if (sc->sc_outports.index == -1 && sc->sc_outports.master != -1) { 8349 mi.index = sc->sc_outports.master; 8350 mi.un.v.delta = 0; 8351 if (audio_query_devinfo(sc, &mi) == 0) { 8352 au_get_gain(sc, &sc->sc_outports, &gain, &balance); 8353 newgain = gain - mi.un.v.delta; 8354 if (newgain < AUDIO_MIN_GAIN) 8355 newgain = AUDIO_MIN_GAIN; 8356 au_set_gain(sc, &sc->sc_outports, newgain, balance); 8357 } 8358 } 8359 audio_exit_exclusive(sc); 8360 } 8361 8362 static void 8363 audio_volume_up(device_t dv) 8364 { 8365 struct audio_softc *sc = device_private(dv); 8366 mixer_devinfo_t mi; 8367 u_int gain, newgain; 8368 u_char balance; 8369 8370 if (audio_enter_exclusive(sc) != 0) 8371 return; 8372 if (sc->sc_outports.index == -1 && sc->sc_outports.master != -1) { 8373 mi.index = sc->sc_outports.master; 8374 mi.un.v.delta = 0; 8375 if (audio_query_devinfo(sc, &mi) == 0) { 8376 au_get_gain(sc, &sc->sc_outports, &gain, &balance); 8377 newgain = gain + mi.un.v.delta; 8378 if (newgain > AUDIO_MAX_GAIN) 8379 newgain = AUDIO_MAX_GAIN; 8380 au_set_gain(sc, &sc->sc_outports, newgain, balance); 8381 } 8382 } 8383 audio_exit_exclusive(sc); 8384 } 8385 8386 static void 8387 audio_volume_toggle(device_t dv) 8388 { 8389 struct audio_softc *sc = device_private(dv); 8390 u_int gain, newgain; 8391 u_char balance; 8392 8393 if (audio_enter_exclusive(sc) != 0) 8394 return; 8395 au_get_gain(sc, &sc->sc_outports, &gain, &balance); 8396 if (gain != 0) { 8397 sc->sc_lastgain = gain; 8398 newgain = 0; 8399 } else 8400 newgain = sc->sc_lastgain; 8401 au_set_gain(sc, &sc->sc_outports, newgain, balance); 8402 audio_exit_exclusive(sc); 8403 } 8404 8405 static int 8406 audio_query_devinfo(struct audio_softc *sc, mixer_devinfo_t *di) 8407 { 8408 8409 KASSERT(mutex_owned(sc->sc_lock)); 8410 8411 return sc->hw_if->query_devinfo(sc->hw_hdl, di); 8412 } 8413 8414 #endif /* NAUDIO > 0 */ 8415 8416 #if NAUDIO == 0 && (NMIDI > 0 || NMIDIBUS > 0) 8417 #include <sys/param.h> 8418 #include <sys/systm.h> 8419 #include <sys/device.h> 8420 #include <sys/audioio.h> 8421 #include <dev/audio/audio_if.h> 8422 #endif 8423 8424 #if NAUDIO > 0 || (NMIDI > 0 || NMIDIBUS > 0) 8425 int 8426 audioprint(void *aux, const char *pnp) 8427 { 8428 struct audio_attach_args *arg; 8429 const char *type; 8430 8431 if (pnp != NULL) { 8432 arg = aux; 8433 switch (arg->type) { 8434 case AUDIODEV_TYPE_AUDIO: 8435 type = "audio"; 8436 break; 8437 case AUDIODEV_TYPE_MIDI: 8438 type = "midi"; 8439 break; 8440 case AUDIODEV_TYPE_OPL: 8441 type = "opl"; 8442 break; 8443 case AUDIODEV_TYPE_MPU: 8444 type = "mpu"; 8445 break; 8446 default: 8447 panic("audioprint: unknown type %d", arg->type); 8448 } 8449 aprint_normal("%s at %s", type, pnp); 8450 } 8451 return UNCONF; 8452 } 8453 8454 #endif /* NAUDIO > 0 || (NMIDI > 0 || NMIDIBUS > 0) */ 8455 8456 #ifdef _MODULE 8457 8458 devmajor_t audio_bmajor = -1, audio_cmajor = -1; 8459 8460 #include "ioconf.c" 8461 8462 #endif 8463 8464 MODULE(MODULE_CLASS_DRIVER, audio, NULL); 8465 8466 static int 8467 audio_modcmd(modcmd_t cmd, void *arg) 8468 { 8469 int error = 0; 8470 8471 #ifdef _MODULE 8472 switch (cmd) { 8473 case MODULE_CMD_INIT: 8474 error = devsw_attach(audio_cd.cd_name, NULL, &audio_bmajor, 8475 &audio_cdevsw, &audio_cmajor); 8476 if (error) 8477 break; 8478 8479 error = config_init_component(cfdriver_ioconf_audio, 8480 cfattach_ioconf_audio, cfdata_ioconf_audio); 8481 if (error) { 8482 devsw_detach(NULL, &audio_cdevsw); 8483 } 8484 break; 8485 case MODULE_CMD_FINI: 8486 devsw_detach(NULL, &audio_cdevsw); 8487 error = config_fini_component(cfdriver_ioconf_audio, 8488 cfattach_ioconf_audio, cfdata_ioconf_audio); 8489 if (error) 8490 devsw_attach(audio_cd.cd_name, NULL, &audio_bmajor, 8491 &audio_cdevsw, &audio_cmajor); 8492 break; 8493 default: 8494 error = ENOTTY; 8495 break; 8496 } 8497 #endif 8498 8499 return error; 8500 } 8501
Indexes created Thu Sep 25 16:09:42 GMT 2025