| 1 | /* $NetBSD: audio.c,v 1.268 2016/07/14 10:19:05 msaitoh 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 | * This is a (partially) SunOS-compatible /dev/audio driver for NetBSD. |
| 67 | * |
| 68 | * This code tries to do something half-way sensible with |
| 69 | * half-duplex hardware, such as with the SoundBlaster hardware. With |
| 70 | * half-duplex hardware allowing O_RDWR access doesn't really make |
| 71 | * sense. However, closing and opening the device to "turn around the |
| 72 | * line" is relatively expensive and costs a card reset (which can |
| 73 | * take some time, at least for the SoundBlaster hardware). Instead |
| 74 | * we allow O_RDWR access, and provide an ioctl to set the "mode", |
| 75 | * i.e. playing or recording. |
| 76 | * |
| 77 | * If you write to a half-duplex device in record mode, the data is |
| 78 | * tossed. If you read from the device in play mode, you get silence |
| 79 | * filled buffers at the rate at which samples are naturally |
| 80 | * generated. |
| 81 | * |
| 82 | * If you try to set both play and record mode on a half-duplex |
| 83 | * device, playing takes precedence. |
| 84 | */ |
| 85 | |
| 86 | /* |
| 87 | * Locking: there are three locks. |
| 88 | * |
| 89 | * - sc_lock, provided by the underlying driver. This is an adaptive lock, |
| 90 | * returned in the second parameter to hw_if->get_locks(). It is known |
| 91 | * as the "thread lock". |
| 92 | * |
| 93 | * It serializes access to state in all places except the |
| 94 | * driver's interrupt service routine. This lock is taken from process |
| 95 | * context (example: access to /dev/audio). It is also taken from soft |
| 96 | * interrupt handlers in this module, primarily to serialize delivery of |
| 97 | * wakeups. This lock may be used/provided by modules external to the |
| 98 | * audio subsystem, so take care not to introduce a lock order problem. |
| 99 | * LONG TERM SLEEPS MUST NOT OCCUR WITH THIS LOCK HELD. |
| 100 | * |
| 101 | * - sc_intr_lock, provided by the underlying driver. This may be either a |
| 102 | * spinlock (at IPL_SCHED or IPL_VM) or an adaptive lock (IPL_NONE or |
| 103 | * IPL_SOFT*), returned in the first parameter to hw_if->get_locks(). It |
| 104 | * is known as the "interrupt lock". |
| 105 | * |
| 106 | * It provides atomic access to the device's hardware state, and to audio |
| 107 | * channel data that may be accessed by the hardware driver's ISR. |
| 108 | * In all places outside the ISR, sc_lock must be held before taking |
| 109 | * sc_intr_lock. This is to ensure that groups of hardware operations are |
| 110 | * made atomically. SLEEPS CANNOT OCCUR WITH THIS LOCK HELD. |
| 111 | * |
| 112 | * - sc_dvlock, private to this module. This is a custom reader/writer lock |
| 113 | * built on sc_lock and a condition variable. Some operations release |
| 114 | * sc_lock in order to allocate memory, to wait for in-flight I/O to |
| 115 | * complete, to copy to/from user context, etc. sc_dvlock serializes |
| 116 | * changes to filters and audio device settings while a read/write to the |
| 117 | * hardware is in progress. A write lock is taken only under exceptional |
| 118 | * circumstances, for example when opening /dev/audio or changing audio |
| 119 | * parameters. Long term sleeps and copy to/from user space may be done |
| 120 | * with this lock held. |
| 121 | * |
| 122 | * List of hardware interface methods, and which locks are held when each |
| 123 | * is called by this module: |
| 124 | * |
| 125 | * METHOD INTR THREAD NOTES |
| 126 | * ----------------------- ------- ------- ------------------------- |
| 127 | * open x x |
| 128 | * close x x |
| 129 | * drain x x |
| 130 | * query_encoding - x |
| 131 | * set_params - x |
| 132 | * round_blocksize - x |
| 133 | * commit_settings - x |
| 134 | * init_output x x |
| 135 | * init_input x x |
| 136 | * start_output x x |
| 137 | * start_input x x |
| 138 | * halt_output x x |
| 139 | * halt_input x x |
| 140 | * speaker_ctl x x |
| 141 | * getdev - x |
| 142 | * setfd - x |
| 143 | * set_port - x |
| 144 | * get_port - x |
| 145 | * query_devinfo - x |
| 146 | * allocm - - Called at attach time |
| 147 | * freem - - Called at attach time |
| 148 | * round_buffersize - x |
| 149 | * mappage - - Mem. unchanged after attach |
| 150 | * get_props - x |
| 151 | * trigger_output x x |
| 152 | * trigger_input x x |
| 153 | * dev_ioctl - x |
| 154 | * get_locks - - Called at attach time |
| 155 | */ |
| 156 | |
| 157 | #include <sys/cdefs.h> |
| 158 | __KERNEL_RCSID(0, "$NetBSD: audio.c,v 1.268 2016/07/14 10:19:05 msaitoh Exp $" ); |
| 159 | |
| 160 | #include "audio.h" |
| 161 | #if NAUDIO > 0 |
| 162 | |
| 163 | #include <sys/param.h> |
| 164 | #include <sys/ioctl.h> |
| 165 | #include <sys/fcntl.h> |
| 166 | #include <sys/vnode.h> |
| 167 | #include <sys/select.h> |
| 168 | #include <sys/poll.h> |
| 169 | #include <sys/kmem.h> |
| 170 | #include <sys/malloc.h> |
| 171 | #include <sys/proc.h> |
| 172 | #include <sys/systm.h> |
| 173 | #include <sys/syslog.h> |
| 174 | #include <sys/kernel.h> |
| 175 | #include <sys/signalvar.h> |
| 176 | #include <sys/conf.h> |
| 177 | #include <sys/audioio.h> |
| 178 | #include <sys/device.h> |
| 179 | #include <sys/intr.h> |
| 180 | #include <sys/cpu.h> |
| 181 | |
| 182 | #include <dev/audio_if.h> |
| 183 | #include <dev/audiovar.h> |
| 184 | |
| 185 | #include <machine/endian.h> |
| 186 | |
| 187 | /* #define AUDIO_DEBUG 1 */ |
| 188 | #ifdef AUDIO_DEBUG |
| 189 | #define DPRINTF(x) if (audiodebug) printf x |
| 190 | #define DPRINTFN(n,x) if (audiodebug>(n)) printf x |
| 191 | int audiodebug = AUDIO_DEBUG; |
| 192 | #else |
| 193 | #define DPRINTF(x) |
| 194 | #define DPRINTFN(n,x) |
| 195 | #endif |
| 196 | |
| 197 | #define ROUNDSIZE(x) x &= -16 /* round to nice boundary */ |
| 198 | #define SPECIFIED(x) (x != ~0) |
| 199 | #define SPECIFIED_CH(x) (x != (u_char)~0) |
| 200 | |
| 201 | /* #define AUDIO_PM_IDLE */ |
| 202 | #ifdef AUDIO_PM_IDLE |
| 203 | int audio_idle_timeout = 30; |
| 204 | #endif |
| 205 | |
| 206 | int audio_blk_ms = AUDIO_BLK_MS; |
| 207 | |
| 208 | int audiosetinfo(struct audio_softc *, struct audio_info *); |
| 209 | int audiogetinfo(struct audio_softc *, struct audio_info *, int); |
| 210 | |
| 211 | int audio_open(dev_t, struct audio_softc *, int, int, struct lwp *); |
| 212 | int audio_close(struct audio_softc *, int, int, struct lwp *); |
| 213 | int audio_read(struct audio_softc *, struct uio *, int); |
| 214 | int audio_write(struct audio_softc *, struct uio *, int); |
| 215 | int audio_ioctl(struct audio_softc *, u_long, void *, int, struct lwp *); |
| 216 | int audio_poll(struct audio_softc *, int, struct lwp *); |
| 217 | int audio_kqfilter(struct audio_softc *, struct knote *); |
| 218 | paddr_t audio_mmap(struct audio_softc *, off_t, int); |
| 219 | |
| 220 | int mixer_open(dev_t, struct audio_softc *, int, int, struct lwp *); |
| 221 | int mixer_close(struct audio_softc *, int, int, struct lwp *); |
| 222 | int mixer_ioctl(struct audio_softc *, u_long, void *, int, struct lwp *); |
| 223 | static void mixer_remove(struct audio_softc *); |
| 224 | static void mixer_signal(struct audio_softc *); |
| 225 | |
| 226 | void audio_init_record(struct audio_softc *); |
| 227 | void audio_init_play(struct audio_softc *); |
| 228 | int audiostartr(struct audio_softc *); |
| 229 | int audiostartp(struct audio_softc *); |
| 230 | void audio_rint(void *); |
| 231 | void audio_pint(void *); |
| 232 | int audio_check_params(struct audio_params *); |
| 233 | |
| 234 | void audio_calc_blksize(struct audio_softc *, int); |
| 235 | void audio_fill_silence(struct audio_params *, uint8_t *, int); |
| 236 | int audio_silence_copyout(struct audio_softc *, int, struct uio *); |
| 237 | |
| 238 | void audio_init_ringbuffer(struct audio_softc *, |
| 239 | struct audio_ringbuffer *, int); |
| 240 | int audio_initbufs(struct audio_softc *); |
| 241 | void audio_calcwater(struct audio_softc *); |
| 242 | int audio_drain(struct audio_softc *); |
| 243 | void audio_clear(struct audio_softc *); |
| 244 | void audio_clear_intr_unlocked(struct audio_softc *sc); |
| 245 | static inline void audio_pint_silence |
| 246 | (struct audio_softc *, struct audio_ringbuffer *, uint8_t *, int); |
| 247 | |
| 248 | int audio_alloc_ring |
| 249 | (struct audio_softc *, struct audio_ringbuffer *, int, size_t); |
| 250 | void audio_free_ring(struct audio_softc *, struct audio_ringbuffer *); |
| 251 | static int audio_setup_pfilters(struct audio_softc *, const audio_params_t *, |
| 252 | stream_filter_list_t *); |
| 253 | static int audio_setup_rfilters(struct audio_softc *, const audio_params_t *, |
| 254 | stream_filter_list_t *); |
| 255 | static void audio_stream_dtor(audio_stream_t *); |
| 256 | static int audio_stream_ctor(audio_stream_t *, const audio_params_t *, int); |
| 257 | static void stream_filter_list_append |
| 258 | (stream_filter_list_t *, stream_filter_factory_t, |
| 259 | const audio_params_t *); |
| 260 | static void stream_filter_list_prepend |
| 261 | (stream_filter_list_t *, stream_filter_factory_t, |
| 262 | const audio_params_t *); |
| 263 | static void stream_filter_list_set |
| 264 | (stream_filter_list_t *, int, stream_filter_factory_t, |
| 265 | const audio_params_t *); |
| 266 | int audio_set_defaults(struct audio_softc *, u_int); |
| 267 | |
| 268 | int audioprobe(device_t, cfdata_t, void *); |
| 269 | void audioattach(device_t, device_t, void *); |
| 270 | int audiodetach(device_t, int); |
| 271 | int audioactivate(device_t, enum devact); |
| 272 | |
| 273 | #ifdef AUDIO_PM_IDLE |
| 274 | static void audio_idle(void *); |
| 275 | static void audio_activity(device_t, devactive_t); |
| 276 | #endif |
| 277 | |
| 278 | static bool audio_suspend(device_t dv, const pmf_qual_t *); |
| 279 | static bool audio_resume(device_t dv, const pmf_qual_t *); |
| 280 | static void audio_volume_down(device_t); |
| 281 | static void audio_volume_up(device_t); |
| 282 | static void audio_volume_toggle(device_t); |
| 283 | |
| 284 | static void audio_mixer_capture(struct audio_softc *); |
| 285 | static void audio_mixer_restore(struct audio_softc *); |
| 286 | |
| 287 | static int audio_get_props(struct audio_softc *); |
| 288 | static bool audio_can_playback(struct audio_softc *); |
| 289 | static bool audio_can_capture(struct audio_softc *); |
| 290 | |
| 291 | static void audio_softintr_rd(void *); |
| 292 | static void audio_softintr_wr(void *); |
| 293 | |
| 294 | static int audio_enter(dev_t, krw_t, struct audio_softc **); |
| 295 | static void audio_exit(struct audio_softc *); |
| 296 | static int audio_waitio(struct audio_softc *, kcondvar_t *); |
| 297 | |
| 298 | struct portname { |
| 299 | const char *name; |
| 300 | int mask; |
| 301 | }; |
| 302 | static const struct portname itable[] = { |
| 303 | { AudioNmicrophone, AUDIO_MICROPHONE }, |
| 304 | { AudioNline, AUDIO_LINE_IN }, |
| 305 | { AudioNcd, AUDIO_CD }, |
| 306 | { 0, 0 } |
| 307 | }; |
| 308 | static const struct portname otable[] = { |
| 309 | { AudioNspeaker, AUDIO_SPEAKER }, |
| 310 | { AudioNheadphone, AUDIO_HEADPHONE }, |
| 311 | { AudioNline, AUDIO_LINE_OUT }, |
| 312 | { 0, 0 } |
| 313 | }; |
| 314 | void au_setup_ports(struct audio_softc *, struct au_mixer_ports *, |
| 315 | mixer_devinfo_t *, const struct portname *); |
| 316 | int au_set_gain(struct audio_softc *, struct au_mixer_ports *, |
| 317 | int, int); |
| 318 | void au_get_gain(struct audio_softc *, struct au_mixer_ports *, |
| 319 | u_int *, u_char *); |
| 320 | int au_set_port(struct audio_softc *, struct au_mixer_ports *, |
| 321 | u_int); |
| 322 | int au_get_port(struct audio_softc *, struct au_mixer_ports *); |
| 323 | int au_get_lr_value(struct audio_softc *, mixer_ctrl_t *, int *, int *); |
| 324 | int au_set_lr_value(struct audio_softc *, mixer_ctrl_t *, int, int); |
| 325 | int au_portof(struct audio_softc *, char *, int); |
| 326 | |
| 327 | typedef struct uio_fetcher { |
| 328 | stream_fetcher_t base; |
| 329 | struct uio *uio; |
| 330 | int usedhigh; |
| 331 | int last_used; |
| 332 | } uio_fetcher_t; |
| 333 | |
| 334 | static void uio_fetcher_ctor(uio_fetcher_t *, struct uio *, int); |
| 335 | static int uio_fetcher_fetch_to(struct audio_softc *, stream_fetcher_t *, |
| 336 | audio_stream_t *, int); |
| 337 | static int null_fetcher_fetch_to(struct audio_softc *, stream_fetcher_t *, |
| 338 | audio_stream_t *, int); |
| 339 | |
| 340 | dev_type_open(audioopen); |
| 341 | dev_type_close(audioclose); |
| 342 | dev_type_read(audioread); |
| 343 | dev_type_write(audiowrite); |
| 344 | dev_type_ioctl(audioioctl); |
| 345 | dev_type_poll(audiopoll); |
| 346 | dev_type_mmap(audiommap); |
| 347 | dev_type_kqfilter(audiokqfilter); |
| 348 | |
| 349 | const struct cdevsw audio_cdevsw = { |
| 350 | .d_open = audioopen, |
| 351 | .d_close = audioclose, |
| 352 | .d_read = audioread, |
| 353 | .d_write = audiowrite, |
| 354 | .d_ioctl = audioioctl, |
| 355 | .d_stop = nostop, |
| 356 | .d_tty = notty, |
| 357 | .d_poll = audiopoll, |
| 358 | .d_mmap = audiommap, |
| 359 | .d_kqfilter = audiokqfilter, |
| 360 | .d_discard = nodiscard, |
| 361 | .d_flag = D_OTHER | D_MPSAFE |
| 362 | }; |
| 363 | |
| 364 | /* The default audio mode: 8 kHz mono mu-law */ |
| 365 | const struct audio_params audio_default = { |
| 366 | .sample_rate = 8000, |
| 367 | .encoding = AUDIO_ENCODING_ULAW, |
| 368 | .precision = 8, |
| 369 | .validbits = 8, |
| 370 | .channels = 1, |
| 371 | }; |
| 372 | |
| 373 | CFATTACH_DECL3_NEW(audio, sizeof(struct audio_softc), |
| 374 | audioprobe, audioattach, audiodetach, audioactivate, NULL, NULL, |
| 375 | DVF_DETACH_SHUTDOWN); |
| 376 | |
| 377 | extern struct cfdriver audio_cd; |
| 378 | |
| 379 | int |
| 380 | audioprobe(device_t parent, cfdata_t match, void *aux) |
| 381 | { |
| 382 | struct audio_attach_args *sa; |
| 383 | |
| 384 | sa = aux; |
| 385 | DPRINTF(("audioprobe: type=%d sa=%p hw=%p\n" , |
| 386 | sa->type, sa, sa->hwif)); |
| 387 | return (sa->type == AUDIODEV_TYPE_AUDIO) ? 1 : 0; |
| 388 | } |
| 389 | |
| 390 | void |
| 391 | audioattach(device_t parent, device_t self, void *aux) |
| 392 | { |
| 393 | struct audio_softc *sc; |
| 394 | struct audio_attach_args *sa; |
| 395 | const struct audio_hw_if *hwp; |
| 396 | void *hdlp; |
| 397 | int error; |
| 398 | mixer_devinfo_t mi; |
| 399 | int iclass, mclass, oclass, rclass, props; |
| 400 | int record_master_found, record_source_found; |
| 401 | bool can_capture, can_playback; |
| 402 | |
| 403 | sc = device_private(self); |
| 404 | sc->dev = self; |
| 405 | sa = aux; |
| 406 | hwp = sa->hwif; |
| 407 | hdlp = sa->hdl; |
| 408 | |
| 409 | cv_init(&sc->sc_rchan, "audiord" ); |
| 410 | cv_init(&sc->sc_wchan, "audiowr" ); |
| 411 | cv_init(&sc->sc_lchan, "audiolk" ); |
| 412 | |
| 413 | if (hwp == 0 || hwp->get_locks == 0) { |
| 414 | aprint_error(": missing method\n" ); |
| 415 | panic("audioattach" ); |
| 416 | } |
| 417 | |
| 418 | hwp->get_locks(hdlp, &sc->sc_intr_lock, &sc->sc_lock); |
| 419 | |
| 420 | #ifdef DIAGNOSTIC |
| 421 | if (hwp->query_encoding == 0 || |
| 422 | hwp->set_params == 0 || |
| 423 | (hwp->start_output == 0 && hwp->trigger_output == 0) || |
| 424 | (hwp->start_input == 0 && hwp->trigger_input == 0) || |
| 425 | hwp->halt_output == 0 || |
| 426 | hwp->halt_input == 0 || |
| 427 | hwp->getdev == 0 || |
| 428 | hwp->set_port == 0 || |
| 429 | hwp->get_port == 0 || |
| 430 | hwp->query_devinfo == 0 || |
| 431 | hwp->get_props == 0) { |
| 432 | aprint_error(": missing method\n" ); |
| 433 | sc->hw_if = 0; |
| 434 | return; |
| 435 | } |
| 436 | #endif |
| 437 | |
| 438 | sc->hw_if = hwp; |
| 439 | sc->hw_hdl = hdlp; |
| 440 | sc->sc_dev = parent; |
| 441 | sc->sc_lastinfovalid = false; |
| 442 | |
| 443 | mutex_enter(sc->sc_lock); |
| 444 | props = audio_get_props(sc); |
| 445 | mutex_exit(sc->sc_lock); |
| 446 | |
| 447 | if (props & AUDIO_PROP_FULLDUPLEX) |
| 448 | aprint_normal(": full duplex" ); |
| 449 | else |
| 450 | aprint_normal(": half duplex" ); |
| 451 | |
| 452 | if (props & AUDIO_PROP_PLAYBACK) |
| 453 | aprint_normal(", playback" ); |
| 454 | if (props & AUDIO_PROP_CAPTURE) |
| 455 | aprint_normal(", capture" ); |
| 456 | if (props & AUDIO_PROP_MMAP) |
| 457 | aprint_normal(", mmap" ); |
| 458 | if (props & AUDIO_PROP_INDEPENDENT) |
| 459 | aprint_normal(", independent" ); |
| 460 | |
| 461 | aprint_naive("\n" ); |
| 462 | aprint_normal("\n" ); |
| 463 | |
| 464 | mutex_enter(sc->sc_lock); |
| 465 | can_playback = audio_can_playback(sc); |
| 466 | can_capture = audio_can_capture(sc); |
| 467 | mutex_exit(sc->sc_lock); |
| 468 | |
| 469 | if (can_playback) { |
| 470 | error = audio_alloc_ring(sc, &sc->sc_pr, |
| 471 | AUMODE_PLAY, AU_RING_SIZE); |
| 472 | if (error) { |
| 473 | sc->hw_if = NULL; |
| 474 | aprint_error("audio: could not allocate play buffer\n" ); |
| 475 | return; |
| 476 | } |
| 477 | } |
| 478 | if (can_capture) { |
| 479 | error = audio_alloc_ring(sc, &sc->sc_rr, |
| 480 | AUMODE_RECORD, AU_RING_SIZE); |
| 481 | if (error) { |
| 482 | if (sc->sc_pr.s.start != 0) |
| 483 | audio_free_ring(sc, &sc->sc_pr); |
| 484 | sc->hw_if = NULL; |
| 485 | aprint_error("audio: could not allocate record buffer\n" ); |
| 486 | return; |
| 487 | } |
| 488 | } |
| 489 | |
| 490 | sc->sc_lastgain = 128; |
| 491 | |
| 492 | mutex_enter(sc->sc_lock); |
| 493 | error = audio_set_defaults(sc, 0); |
| 494 | mutex_exit(sc->sc_lock); |
| 495 | if (error != 0) { |
| 496 | aprint_error("audioattach: audio_set_defaults() failed\n" ); |
| 497 | sc->hw_if = NULL; |
| 498 | return; |
| 499 | } |
| 500 | |
| 501 | sc->sc_sih_rd = softint_establish(SOFTINT_SERIAL | SOFTINT_MPSAFE, |
| 502 | audio_softintr_rd, sc); |
| 503 | sc->sc_sih_wr = softint_establish(SOFTINT_SERIAL | SOFTINT_MPSAFE, |
| 504 | audio_softintr_wr, sc); |
| 505 | |
| 506 | iclass = mclass = oclass = rclass = -1; |
| 507 | sc->sc_inports.index = -1; |
| 508 | sc->sc_inports.master = -1; |
| 509 | sc->sc_inports.nports = 0; |
| 510 | sc->sc_inports.isenum = false; |
| 511 | sc->sc_inports.allports = 0; |
| 512 | sc->sc_inports.isdual = false; |
| 513 | sc->sc_inports.mixerout = -1; |
| 514 | sc->sc_inports.cur_port = -1; |
| 515 | sc->sc_outports.index = -1; |
| 516 | sc->sc_outports.master = -1; |
| 517 | sc->sc_outports.nports = 0; |
| 518 | sc->sc_outports.isenum = false; |
| 519 | sc->sc_outports.allports = 0; |
| 520 | sc->sc_outports.isdual = false; |
| 521 | sc->sc_outports.mixerout = -1; |
| 522 | sc->sc_outports.cur_port = -1; |
| 523 | sc->sc_monitor_port = -1; |
| 524 | /* |
| 525 | * Read through the underlying driver's list, picking out the class |
| 526 | * names from the mixer descriptions. We'll need them to decode the |
| 527 | * mixer descriptions on the next pass through the loop. |
| 528 | */ |
| 529 | mutex_enter(sc->sc_lock); |
| 530 | for(mi.index = 0; ; mi.index++) { |
| 531 | if (hwp->query_devinfo(hdlp, &mi) != 0) |
| 532 | break; |
| 533 | /* |
| 534 | * The type of AUDIO_MIXER_CLASS merely introduces a class. |
| 535 | * All the other types describe an actual mixer. |
| 536 | */ |
| 537 | if (mi.type == AUDIO_MIXER_CLASS) { |
| 538 | if (strcmp(mi.label.name, AudioCinputs) == 0) |
| 539 | iclass = mi.mixer_class; |
| 540 | if (strcmp(mi.label.name, AudioCmonitor) == 0) |
| 541 | mclass = mi.mixer_class; |
| 542 | if (strcmp(mi.label.name, AudioCoutputs) == 0) |
| 543 | oclass = mi.mixer_class; |
| 544 | if (strcmp(mi.label.name, AudioCrecord) == 0) |
| 545 | rclass = mi.mixer_class; |
| 546 | } |
| 547 | } |
| 548 | mutex_exit(sc->sc_lock); |
| 549 | |
| 550 | /* Allocate save area. Ensure non-zero allocation. */ |
| 551 | sc->sc_nmixer_states = mi.index; |
| 552 | sc->sc_mixer_state = kmem_alloc(sizeof(mixer_ctrl_t) * |
| 553 | sc->sc_nmixer_states + 1, KM_SLEEP); |
| 554 | |
| 555 | /* |
| 556 | * This is where we assign each control in the "audio" model, to the |
| 557 | * underlying "mixer" control. We walk through the whole list once, |
| 558 | * assigning likely candidates as we come across them. |
| 559 | */ |
| 560 | record_master_found = 0; |
| 561 | record_source_found = 0; |
| 562 | mutex_enter(sc->sc_lock); |
| 563 | for(mi.index = 0; ; mi.index++) { |
| 564 | if (hwp->query_devinfo(hdlp, &mi) != 0) |
| 565 | break; |
| 566 | KASSERT(mi.index < sc->sc_nmixer_states); |
| 567 | if (mi.type == AUDIO_MIXER_CLASS) |
| 568 | continue; |
| 569 | if (mi.mixer_class == iclass) { |
| 570 | /* |
| 571 | * AudioCinputs is only a fallback, when we don't |
| 572 | * find what we're looking for in AudioCrecord, so |
| 573 | * check the flags before accepting one of these. |
| 574 | */ |
| 575 | if (strcmp(mi.label.name, AudioNmaster) == 0 |
| 576 | && record_master_found == 0) |
| 577 | sc->sc_inports.master = mi.index; |
| 578 | if (strcmp(mi.label.name, AudioNsource) == 0 |
| 579 | && record_source_found == 0) { |
| 580 | if (mi.type == AUDIO_MIXER_ENUM) { |
| 581 | int i; |
| 582 | for(i = 0; i < mi.un.e.num_mem; i++) |
| 583 | if (strcmp(mi.un.e.member[i].label.name, |
| 584 | AudioNmixerout) == 0) |
| 585 | sc->sc_inports.mixerout = |
| 586 | mi.un.e.member[i].ord; |
| 587 | } |
| 588 | au_setup_ports(sc, &sc->sc_inports, &mi, |
| 589 | itable); |
| 590 | } |
| 591 | if (strcmp(mi.label.name, AudioNdac) == 0 && |
| 592 | sc->sc_outports.master == -1) |
| 593 | sc->sc_outports.master = mi.index; |
| 594 | } else if (mi.mixer_class == mclass) { |
| 595 | if (strcmp(mi.label.name, AudioNmonitor) == 0) |
| 596 | sc->sc_monitor_port = mi.index; |
| 597 | } else if (mi.mixer_class == oclass) { |
| 598 | if (strcmp(mi.label.name, AudioNmaster) == 0) |
| 599 | sc->sc_outports.master = mi.index; |
| 600 | if (strcmp(mi.label.name, AudioNselect) == 0) |
| 601 | au_setup_ports(sc, &sc->sc_outports, &mi, |
| 602 | otable); |
| 603 | } else if (mi.mixer_class == rclass) { |
| 604 | /* |
| 605 | * These are the preferred mixers for the audio record |
| 606 | * controls, so set the flags here, but don't check. |
| 607 | */ |
| 608 | if (strcmp(mi.label.name, AudioNmaster) == 0) { |
| 609 | sc->sc_inports.master = mi.index; |
| 610 | record_master_found = 1; |
| 611 | } |
| 612 | #if 1 /* Deprecated. Use AudioNmaster. */ |
| 613 | if (strcmp(mi.label.name, AudioNrecord) == 0) { |
| 614 | sc->sc_inports.master = mi.index; |
| 615 | record_master_found = 1; |
| 616 | } |
| 617 | if (strcmp(mi.label.name, AudioNvolume) == 0) { |
| 618 | sc->sc_inports.master = mi.index; |
| 619 | record_master_found = 1; |
| 620 | } |
| 621 | #endif |
| 622 | if (strcmp(mi.label.name, AudioNsource) == 0) { |
| 623 | if (mi.type == AUDIO_MIXER_ENUM) { |
| 624 | int i; |
| 625 | for(i = 0; i < mi.un.e.num_mem; i++) |
| 626 | if (strcmp(mi.un.e.member[i].label.name, |
| 627 | AudioNmixerout) == 0) |
| 628 | sc->sc_inports.mixerout = |
| 629 | mi.un.e.member[i].ord; |
| 630 | } |
| 631 | au_setup_ports(sc, &sc->sc_inports, &mi, |
| 632 | itable); |
| 633 | record_source_found = 1; |
| 634 | } |
| 635 | } |
| 636 | } |
| 637 | mutex_exit(sc->sc_lock); |
| 638 | DPRINTF(("audio_attach: inputs ports=0x%x, input master=%d, " |
| 639 | "output ports=0x%x, output master=%d\n" , |
| 640 | sc->sc_inports.allports, sc->sc_inports.master, |
| 641 | sc->sc_outports.allports, sc->sc_outports.master)); |
| 642 | |
| 643 | selinit(&sc->sc_rsel); |
| 644 | selinit(&sc->sc_wsel); |
| 645 | |
| 646 | #ifdef AUDIO_PM_IDLE |
| 647 | callout_init(&sc->sc_idle_counter, 0); |
| 648 | callout_setfunc(&sc->sc_idle_counter, audio_idle, self); |
| 649 | #endif |
| 650 | |
| 651 | if (!pmf_device_register(self, audio_suspend, audio_resume)) |
| 652 | aprint_error_dev(self, "couldn't establish power handler\n" ); |
| 653 | #ifdef AUDIO_PM_IDLE |
| 654 | if (!device_active_register(self, audio_activity)) |
| 655 | aprint_error_dev(self, "couldn't register activity handler\n" ); |
| 656 | #endif |
| 657 | |
| 658 | if (!pmf_event_register(self, PMFE_AUDIO_VOLUME_DOWN, |
| 659 | audio_volume_down, true)) |
| 660 | aprint_error_dev(self, "couldn't add volume down handler\n" ); |
| 661 | if (!pmf_event_register(self, PMFE_AUDIO_VOLUME_UP, |
| 662 | audio_volume_up, true)) |
| 663 | aprint_error_dev(self, "couldn't add volume up handler\n" ); |
| 664 | if (!pmf_event_register(self, PMFE_AUDIO_VOLUME_TOGGLE, |
| 665 | audio_volume_toggle, true)) |
| 666 | aprint_error_dev(self, "couldn't add volume toggle handler\n" ); |
| 667 | |
| 668 | #ifdef AUDIO_PM_IDLE |
| 669 | callout_schedule(&sc->sc_idle_counter, audio_idle_timeout * hz); |
| 670 | #endif |
| 671 | } |
| 672 | |
| 673 | int |
| 674 | audioactivate(device_t self, enum devact act) |
| 675 | { |
| 676 | struct audio_softc *sc = device_private(self); |
| 677 | |
| 678 | switch (act) { |
| 679 | case DVACT_DEACTIVATE: |
| 680 | mutex_enter(sc->sc_lock); |
| 681 | sc->sc_dying = true; |
| 682 | mutex_exit(sc->sc_lock); |
| 683 | return 0; |
| 684 | default: |
| 685 | return EOPNOTSUPP; |
| 686 | } |
| 687 | } |
| 688 | |
| 689 | int |
| 690 | audiodetach(device_t self, int flags) |
| 691 | { |
| 692 | struct audio_softc *sc; |
| 693 | int maj, mn, i; |
| 694 | |
| 695 | sc = device_private(self); |
| 696 | DPRINTF(("audio_detach: sc=%p flags=%d\n" , sc, flags)); |
| 697 | |
| 698 | /* Start draining existing accessors of the device. */ |
| 699 | mutex_enter(sc->sc_lock); |
| 700 | sc->sc_dying = true; |
| 701 | cv_broadcast(&sc->sc_wchan); |
| 702 | cv_broadcast(&sc->sc_rchan); |
| 703 | mutex_exit(sc->sc_lock); |
| 704 | |
| 705 | /* locate the major number */ |
| 706 | maj = cdevsw_lookup_major(&audio_cdevsw); |
| 707 | |
| 708 | /* |
| 709 | * Nuke the vnodes for any open instances (calls close). |
| 710 | * Will wait until any activity on the device nodes has ceased. |
| 711 | * |
| 712 | * XXXAD NOT YET. |
| 713 | * |
| 714 | * XXXAD NEED TO PREVENT NEW REFERENCES THROUGH AUDIO_ENTER(). |
| 715 | */ |
| 716 | mn = device_unit(self); |
| 717 | vdevgone(maj, mn | SOUND_DEVICE, mn | SOUND_DEVICE, VCHR); |
| 718 | vdevgone(maj, mn | AUDIO_DEVICE, mn | AUDIO_DEVICE, VCHR); |
| 719 | vdevgone(maj, mn | AUDIOCTL_DEVICE, mn | AUDIOCTL_DEVICE, VCHR); |
| 720 | vdevgone(maj, mn | MIXER_DEVICE, mn | MIXER_DEVICE, VCHR); |
| 721 | |
| 722 | pmf_event_deregister(self, PMFE_AUDIO_VOLUME_DOWN, |
| 723 | audio_volume_down, true); |
| 724 | pmf_event_deregister(self, PMFE_AUDIO_VOLUME_UP, |
| 725 | audio_volume_up, true); |
| 726 | pmf_event_deregister(self, PMFE_AUDIO_VOLUME_TOGGLE, |
| 727 | audio_volume_toggle, true); |
| 728 | |
| 729 | #ifdef AUDIO_PM_IDLE |
| 730 | callout_halt(&sc->sc_idle_counter, sc->sc_lock); |
| 731 | |
| 732 | device_active_deregister(self, audio_activity); |
| 733 | #endif |
| 734 | |
| 735 | pmf_device_deregister(self); |
| 736 | |
| 737 | /* free resources */ |
| 738 | audio_free_ring(sc, &sc->sc_pr); |
| 739 | audio_free_ring(sc, &sc->sc_rr); |
| 740 | for (i = 0; i < sc->sc_nrfilters; i++) { |
| 741 | sc->sc_rfilters[i]->dtor(sc->sc_rfilters[i]); |
| 742 | sc->sc_rfilters[i] = NULL; |
| 743 | audio_stream_dtor(&sc->sc_rstreams[i]); |
| 744 | } |
| 745 | sc->sc_nrfilters = 0; |
| 746 | for (i = 0; i < sc->sc_npfilters; i++) { |
| 747 | sc->sc_pfilters[i]->dtor(sc->sc_pfilters[i]); |
| 748 | sc->sc_pfilters[i] = NULL; |
| 749 | audio_stream_dtor(&sc->sc_pstreams[i]); |
| 750 | } |
| 751 | sc->sc_npfilters = 0; |
| 752 | |
| 753 | if (sc->sc_sih_rd) { |
| 754 | softint_disestablish(sc->sc_sih_rd); |
| 755 | sc->sc_sih_rd = NULL; |
| 756 | } |
| 757 | if (sc->sc_sih_wr) { |
| 758 | softint_disestablish(sc->sc_sih_wr); |
| 759 | sc->sc_sih_wr = NULL; |
| 760 | } |
| 761 | |
| 762 | #ifdef AUDIO_PM_IDLE |
| 763 | callout_destroy(&sc->sc_idle_counter); |
| 764 | #endif |
| 765 | seldestroy(&sc->sc_rsel); |
| 766 | seldestroy(&sc->sc_wsel); |
| 767 | |
| 768 | cv_destroy(&sc->sc_rchan); |
| 769 | cv_destroy(&sc->sc_wchan); |
| 770 | cv_destroy(&sc->sc_lchan); |
| 771 | |
| 772 | return 0; |
| 773 | } |
| 774 | |
| 775 | int |
| 776 | au_portof(struct audio_softc *sc, char *name, int class) |
| 777 | { |
| 778 | mixer_devinfo_t mi; |
| 779 | |
| 780 | for(mi.index = 0; |
| 781 | sc->hw_if->query_devinfo(sc->hw_hdl, &mi) == 0; |
| 782 | mi.index++) |
| 783 | if (mi.mixer_class == class && strcmp(mi.label.name, name) == 0) |
| 784 | return mi.index; |
| 785 | return -1; |
| 786 | } |
| 787 | |
| 788 | void |
| 789 | au_setup_ports(struct audio_softc *sc, struct au_mixer_ports *ports, |
| 790 | mixer_devinfo_t *mi, const struct portname *tbl) |
| 791 | { |
| 792 | int i, j; |
| 793 | |
| 794 | ports->index = mi->index; |
| 795 | if (mi->type == AUDIO_MIXER_ENUM) { |
| 796 | ports->isenum = true; |
| 797 | for(i = 0; tbl[i].name; i++) |
| 798 | for(j = 0; j < mi->un.e.num_mem; j++) |
| 799 | if (strcmp(mi->un.e.member[j].label.name, |
| 800 | tbl[i].name) == 0) { |
| 801 | ports->allports |= tbl[i].mask; |
| 802 | ports->aumask[ports->nports] = tbl[i].mask; |
| 803 | ports->misel[ports->nports] = |
| 804 | mi->un.e.member[j].ord; |
| 805 | ports->miport[ports->nports] = |
| 806 | au_portof(sc, mi->un.e.member[j].label.name, |
| 807 | mi->mixer_class); |
| 808 | if (ports->mixerout != -1 && |
| 809 | ports->miport[ports->nports] != -1) |
| 810 | ports->isdual = true; |
| 811 | ++ports->nports; |
| 812 | } |
| 813 | } else if (mi->type == AUDIO_MIXER_SET) { |
| 814 | for(i = 0; tbl[i].name; i++) |
| 815 | for(j = 0; j < mi->un.s.num_mem; j++) |
| 816 | if (strcmp(mi->un.s.member[j].label.name, |
| 817 | tbl[i].name) == 0) { |
| 818 | ports->allports |= tbl[i].mask; |
| 819 | ports->aumask[ports->nports] = tbl[i].mask; |
| 820 | ports->misel[ports->nports] = |
| 821 | mi->un.s.member[j].mask; |
| 822 | ports->miport[ports->nports] = |
| 823 | au_portof(sc, mi->un.s.member[j].label.name, |
| 824 | mi->mixer_class); |
| 825 | ++ports->nports; |
| 826 | } |
| 827 | } |
| 828 | } |
| 829 | |
| 830 | /* |
| 831 | * Called from hardware driver. This is where the MI audio driver gets |
| 832 | * probed/attached to the hardware driver. |
| 833 | */ |
| 834 | device_t |
| 835 | audio_attach_mi(const struct audio_hw_if *ahwp, void *hdlp, device_t dev) |
| 836 | { |
| 837 | struct audio_attach_args arg; |
| 838 | |
| 839 | #ifdef DIAGNOSTIC |
| 840 | if (ahwp == NULL) { |
| 841 | aprint_error("audio_attach_mi: NULL\n" ); |
| 842 | return 0; |
| 843 | } |
| 844 | #endif |
| 845 | arg.type = AUDIODEV_TYPE_AUDIO; |
| 846 | arg.hwif = ahwp; |
| 847 | arg.hdl = hdlp; |
| 848 | return config_found(dev, &arg, audioprint); |
| 849 | } |
| 850 | |
| 851 | #ifdef AUDIO_DEBUG |
| 852 | void audio_printsc(struct audio_softc *); |
| 853 | void audio_print_params(const char *, struct audio_params *); |
| 854 | |
| 855 | void |
| 856 | audio_printsc(struct audio_softc *sc) |
| 857 | { |
| 858 | printf("hwhandle %p hw_if %p " , sc->hw_hdl, sc->hw_if); |
| 859 | printf("open 0x%x mode 0x%x\n" , sc->sc_open, sc->sc_mode); |
| 860 | printf("rchan 0x%x wchan 0x%x " , cv_has_waiters(&sc->sc_rchan), |
| 861 | cv_has_waiters(&sc->sc_wchan)); |
| 862 | printf("rring used 0x%x pring used=%d\n" , |
| 863 | audio_stream_get_used(&sc->sc_rr.s), |
| 864 | audio_stream_get_used(&sc->sc_pr.s)); |
| 865 | printf("rbus 0x%x pbus 0x%x " , sc->sc_rbus, sc->sc_pbus); |
| 866 | printf("blksize %d" , sc->sc_pr.blksize); |
| 867 | printf("hiwat %d lowat %d\n" , sc->sc_pr.usedhigh, sc->sc_pr.usedlow); |
| 868 | } |
| 869 | |
| 870 | void |
| 871 | audio_print_params(const char *s, struct audio_params *p) |
| 872 | { |
| 873 | printf("%s enc=%u %uch %u/%ubit %uHz\n" , s, p->encoding, p->channels, |
| 874 | p->validbits, p->precision, p->sample_rate); |
| 875 | } |
| 876 | #endif |
| 877 | |
| 878 | int |
| 879 | audio_alloc_ring(struct audio_softc *sc, struct audio_ringbuffer *r, |
| 880 | int direction, size_t bufsize) |
| 881 | { |
| 882 | const struct audio_hw_if *hw; |
| 883 | void *hdl; |
| 884 | |
| 885 | hw = sc->hw_if; |
| 886 | hdl = sc->hw_hdl; |
| 887 | /* |
| 888 | * Alloc DMA play and record buffers |
| 889 | */ |
| 890 | if (bufsize < AUMINBUF) |
| 891 | bufsize = AUMINBUF; |
| 892 | ROUNDSIZE(bufsize); |
| 893 | if (hw->round_buffersize) { |
| 894 | mutex_enter(sc->sc_lock); |
| 895 | bufsize = hw->round_buffersize(hdl, direction, bufsize); |
| 896 | mutex_exit(sc->sc_lock); |
| 897 | } |
| 898 | if (hw->allocm) |
| 899 | r->s.start = hw->allocm(hdl, direction, bufsize); |
| 900 | else |
| 901 | r->s.start = kmem_alloc(bufsize, KM_SLEEP); |
| 902 | if (r->s.start == 0) |
| 903 | return ENOMEM; |
| 904 | r->s.bufsize = bufsize; |
| 905 | return 0; |
| 906 | } |
| 907 | |
| 908 | void |
| 909 | audio_free_ring(struct audio_softc *sc, struct audio_ringbuffer *r) |
| 910 | { |
| 911 | if (r->s.start == 0) |
| 912 | return; |
| 913 | |
| 914 | if (sc->hw_if->freem) |
| 915 | sc->hw_if->freem(sc->hw_hdl, r->s.start, r->s.bufsize); |
| 916 | else |
| 917 | kmem_free(r->s.start, r->s.bufsize); |
| 918 | r->s.start = 0; |
| 919 | } |
| 920 | |
| 921 | static int |
| 922 | audio_setup_pfilters(struct audio_softc *sc, const audio_params_t *pp, |
| 923 | stream_filter_list_t *pfilters) |
| 924 | { |
| 925 | stream_filter_t *pf[AUDIO_MAX_FILTERS], *of[AUDIO_MAX_FILTERS]; |
| 926 | audio_stream_t ps[AUDIO_MAX_FILTERS], os[AUDIO_MAX_FILTERS]; |
| 927 | const audio_params_t *from_param; |
| 928 | audio_params_t *to_param; |
| 929 | int i, n, onfilters; |
| 930 | |
| 931 | KASSERT(mutex_owned(sc->sc_lock)); |
| 932 | |
| 933 | /* Construct new filters. */ |
| 934 | mutex_exit(sc->sc_lock); |
| 935 | memset(pf, 0, sizeof(pf)); |
| 936 | memset(ps, 0, sizeof(ps)); |
| 937 | from_param = pp; |
| 938 | for (i = 0; i < pfilters->req_size; i++) { |
| 939 | n = pfilters->req_size - i - 1; |
| 940 | to_param = &pfilters->filters[n].param; |
| 941 | audio_check_params(to_param); |
| 942 | pf[i] = pfilters->filters[n].factory(sc, from_param, to_param); |
| 943 | if (pf[i] == NULL) |
| 944 | break; |
| 945 | if (audio_stream_ctor(&ps[i], from_param, AU_RING_SIZE)) |
| 946 | break; |
| 947 | if (i > 0) |
| 948 | pf[i]->set_fetcher(pf[i], &pf[i - 1]->base); |
| 949 | from_param = to_param; |
| 950 | } |
| 951 | if (i < pfilters->req_size) { /* failure */ |
| 952 | DPRINTF(("%s: pfilters failure\n" , __func__)); |
| 953 | for (; i >= 0; i--) { |
| 954 | if (pf[i] != NULL) |
| 955 | pf[i]->dtor(pf[i]); |
| 956 | audio_stream_dtor(&ps[i]); |
| 957 | } |
| 958 | mutex_enter(sc->sc_lock); |
| 959 | return EINVAL; |
| 960 | } |
| 961 | mutex_enter(sc->sc_lock); |
| 962 | |
| 963 | /* Swap in new filters. */ |
| 964 | mutex_enter(sc->sc_intr_lock); |
| 965 | memcpy(of, sc->sc_pfilters, sizeof(of)); |
| 966 | memcpy(os, sc->sc_pstreams, sizeof(os)); |
| 967 | onfilters = sc->sc_npfilters; |
| 968 | memcpy(sc->sc_pfilters, pf, sizeof(pf)); |
| 969 | memcpy(sc->sc_pstreams, ps, sizeof(ps)); |
| 970 | sc->sc_npfilters = pfilters->req_size; |
| 971 | for (i = 0; i < pfilters->req_size; i++) { |
| 972 | pf[i]->set_inputbuffer(pf[i], &sc->sc_pstreams[i]); |
| 973 | } |
| 974 | /* hardware format and the buffer near to userland */ |
| 975 | if (pfilters->req_size <= 0) { |
| 976 | sc->sc_pr.s.param = *pp; |
| 977 | sc->sc_pustream = &sc->sc_pr.s; |
| 978 | } else { |
| 979 | sc->sc_pr.s.param = pfilters->filters[0].param; |
| 980 | sc->sc_pustream = &sc->sc_pstreams[0]; |
| 981 | } |
| 982 | mutex_exit(sc->sc_intr_lock); |
| 983 | |
| 984 | /* Destroy old filters. */ |
| 985 | mutex_exit(sc->sc_lock); |
| 986 | for (i = 0; i < onfilters; i++) { |
| 987 | of[i]->dtor(of[i]); |
| 988 | audio_stream_dtor(&os[i]); |
| 989 | } |
| 990 | mutex_enter(sc->sc_lock); |
| 991 | |
| 992 | #ifdef AUDIO_DEBUG |
| 993 | printf("%s: HW-buffer=%p pustream=%p\n" , |
| 994 | __func__, &sc->sc_pr.s, sc->sc_pustream); |
| 995 | for (i = 0; i < pfilters->req_size; i++) { |
| 996 | char num[100]; |
| 997 | snprintf(num, 100, "[%d]" , i); |
| 998 | audio_print_params(num, &sc->sc_pstreams[i].param); |
| 999 | } |
| 1000 | audio_print_params("[HW]" , &sc->sc_pr.s.param); |
| 1001 | #endif /* AUDIO_DEBUG */ |
| 1002 | |
| 1003 | return 0; |
| 1004 | } |
| 1005 | |
| 1006 | static int |
| 1007 | audio_setup_rfilters(struct audio_softc *sc, const audio_params_t *rp, |
| 1008 | stream_filter_list_t *rfilters) |
| 1009 | { |
| 1010 | stream_filter_t *rf[AUDIO_MAX_FILTERS], *of[AUDIO_MAX_FILTERS]; |
| 1011 | audio_stream_t rs[AUDIO_MAX_FILTERS], os[AUDIO_MAX_FILTERS]; |
| 1012 | const audio_params_t *to_param; |
| 1013 | audio_params_t *from_param; |
| 1014 | int i, onfilters; |
| 1015 | |
| 1016 | KASSERT(mutex_owned(sc->sc_lock)); |
| 1017 | |
| 1018 | /* Construct new filters. */ |
| 1019 | mutex_exit(sc->sc_lock); |
| 1020 | memset(rf, 0, sizeof(rf)); |
| 1021 | memset(rs, 0, sizeof(rs)); |
| 1022 | for (i = 0; i < rfilters->req_size; i++) { |
| 1023 | from_param = &rfilters->filters[i].param; |
| 1024 | audio_check_params(from_param); |
| 1025 | to_param = i + 1 < rfilters->req_size |
| 1026 | ? &rfilters->filters[i + 1].param : rp; |
| 1027 | rf[i] = rfilters->filters[i].factory(sc, from_param, to_param); |
| 1028 | if (rf[i] == NULL) |
| 1029 | break; |
| 1030 | if (audio_stream_ctor(&rs[i], to_param, AU_RING_SIZE)) |
| 1031 | break; |
| 1032 | if (i > 0) { |
| 1033 | rf[i]->set_fetcher(rf[i], &rf[i - 1]->base); |
| 1034 | } else { |
| 1035 | /* rf[0] has no previous fetcher because |
| 1036 | * the audio hardware fills data to the |
| 1037 | * input buffer. */ |
| 1038 | rf[0]->set_inputbuffer(rf[0], &sc->sc_rr.s); |
| 1039 | } |
| 1040 | } |
| 1041 | if (i < rfilters->req_size) { /* failure */ |
| 1042 | DPRINTF(("%s: rfilters failure\n" , __func__)); |
| 1043 | for (; i >= 0; i--) { |
| 1044 | if (rf[i] != NULL) |
| 1045 | rf[i]->dtor(rf[i]); |
| 1046 | audio_stream_dtor(&rs[i]); |
| 1047 | } |
| 1048 | mutex_enter(sc->sc_lock); |
| 1049 | return EINVAL; |
| 1050 | } |
| 1051 | mutex_enter(sc->sc_lock); |
| 1052 | |
| 1053 | /* Swap in new filters. */ |
| 1054 | mutex_enter(sc->sc_intr_lock); |
| 1055 | memcpy(of, sc->sc_rfilters, sizeof(of)); |
| 1056 | memcpy(os, sc->sc_rstreams, sizeof(os)); |
| 1057 | onfilters = sc->sc_nrfilters; |
| 1058 | memcpy(sc->sc_rfilters, rf, sizeof(rf)); |
| 1059 | memcpy(sc->sc_rstreams, rs, sizeof(rs)); |
| 1060 | sc->sc_nrfilters = rfilters->req_size; |
| 1061 | for (i = 1; i < rfilters->req_size; i++) { |
| 1062 | rf[i]->set_inputbuffer(rf[i], &sc->sc_rstreams[i - 1]); |
| 1063 | } |
| 1064 | /* hardware format and the buffer near to userland */ |
| 1065 | if (rfilters->req_size <= 0) { |
| 1066 | sc->sc_rr.s.param = *rp; |
| 1067 | sc->sc_rustream = &sc->sc_rr.s; |
| 1068 | } else { |
| 1069 | sc->sc_rr.s.param = rfilters->filters[0].param; |
| 1070 | sc->sc_rustream = &sc->sc_rstreams[rfilters->req_size - 1]; |
| 1071 | } |
| 1072 | mutex_exit(sc->sc_intr_lock); |
| 1073 | |
| 1074 | #ifdef AUDIO_DEBUG |
| 1075 | printf("%s: HW-buffer=%p pustream=%p\n" , |
| 1076 | __func__, &sc->sc_rr.s, sc->sc_rustream); |
| 1077 | audio_print_params("[HW]" , &sc->sc_rr.s.param); |
| 1078 | for (i = 0; i < rfilters->req_size; i++) { |
| 1079 | char num[100]; |
| 1080 | snprintf(num, 100, "[%d]" , i); |
| 1081 | audio_print_params(num, &sc->sc_rstreams[i].param); |
| 1082 | } |
| 1083 | #endif /* AUDIO_DEBUG */ |
| 1084 | |
| 1085 | /* Destroy old filters. */ |
| 1086 | mutex_exit(sc->sc_lock); |
| 1087 | for (i = 0; i < onfilters; i++) { |
| 1088 | of[i]->dtor(of[i]); |
| 1089 | audio_stream_dtor(&os[i]); |
| 1090 | } |
| 1091 | mutex_enter(sc->sc_lock); |
| 1092 | |
| 1093 | return 0; |
| 1094 | } |
| 1095 | |
| 1096 | static void |
| 1097 | audio_stream_dtor(audio_stream_t *stream) |
| 1098 | { |
| 1099 | |
| 1100 | if (stream->start != NULL) |
| 1101 | kmem_free(stream->start, stream->bufsize); |
| 1102 | memset(stream, 0, sizeof(audio_stream_t)); |
| 1103 | } |
| 1104 | |
| 1105 | static int |
| 1106 | audio_stream_ctor(audio_stream_t *stream, const audio_params_t *param, int size) |
| 1107 | { |
| 1108 | int frame_size; |
| 1109 | |
| 1110 | size = min(size, AU_RING_SIZE); |
| 1111 | stream->bufsize = size; |
| 1112 | stream->start = kmem_alloc(size, KM_SLEEP); |
| 1113 | if (stream->start == NULL) |
| 1114 | return ENOMEM; |
| 1115 | frame_size = (param->precision + 7) / 8 * param->channels; |
| 1116 | size = (size / frame_size) * frame_size; |
| 1117 | stream->end = stream->start + size; |
| 1118 | stream->inp = stream->start; |
| 1119 | stream->outp = stream->start; |
| 1120 | stream->used = 0; |
| 1121 | stream->param = *param; |
| 1122 | stream->loop = false; |
| 1123 | return 0; |
| 1124 | } |
| 1125 | |
| 1126 | static void |
| 1127 | stream_filter_list_append(stream_filter_list_t *list, |
| 1128 | stream_filter_factory_t factory, |
| 1129 | const audio_params_t *param) |
| 1130 | { |
| 1131 | |
| 1132 | if (list->req_size >= AUDIO_MAX_FILTERS) { |
| 1133 | printf("%s: increase AUDIO_MAX_FILTERS in sys/dev/audio_if.h\n" , |
| 1134 | __func__); |
| 1135 | return; |
| 1136 | } |
| 1137 | list->filters[list->req_size].factory = factory; |
| 1138 | list->filters[list->req_size].param = *param; |
| 1139 | list->req_size++; |
| 1140 | } |
| 1141 | |
| 1142 | static void |
| 1143 | stream_filter_list_set(stream_filter_list_t *list, int i, |
| 1144 | stream_filter_factory_t factory, |
| 1145 | const audio_params_t *param) |
| 1146 | { |
| 1147 | |
| 1148 | if (i < 0 || i >= AUDIO_MAX_FILTERS) { |
| 1149 | printf("%s: invalid index: %d\n" , __func__, i); |
| 1150 | return; |
| 1151 | } |
| 1152 | |
| 1153 | list->filters[i].factory = factory; |
| 1154 | list->filters[i].param = *param; |
| 1155 | if (list->req_size <= i) |
| 1156 | list->req_size = i + 1; |
| 1157 | } |
| 1158 | |
| 1159 | static void |
| 1160 | stream_filter_list_prepend(stream_filter_list_t *list, |
| 1161 | stream_filter_factory_t factory, |
| 1162 | const audio_params_t *param) |
| 1163 | { |
| 1164 | |
| 1165 | if (list->req_size >= AUDIO_MAX_FILTERS) { |
| 1166 | printf("%s: increase AUDIO_MAX_FILTERS in sys/dev/audio_if.h\n" , |
| 1167 | __func__); |
| 1168 | return; |
| 1169 | } |
| 1170 | memmove(&list->filters[1], &list->filters[0], |
| 1171 | sizeof(struct stream_filter_req) * list->req_size); |
| 1172 | list->filters[0].factory = factory; |
| 1173 | list->filters[0].param = *param; |
| 1174 | list->req_size++; |
| 1175 | } |
| 1176 | |
| 1177 | /* |
| 1178 | * Look up audio device and acquire locks for device access. |
| 1179 | */ |
| 1180 | static int |
| 1181 | audio_enter(dev_t dev, krw_t rw, struct audio_softc **scp) |
| 1182 | { |
| 1183 | struct audio_softc *sc; |
| 1184 | |
| 1185 | /* First, find the device and take sc_lock. */ |
| 1186 | sc = device_lookup_private(&audio_cd, AUDIOUNIT(dev)); |
| 1187 | if (sc == NULL) |
| 1188 | return ENXIO; |
| 1189 | mutex_enter(sc->sc_lock); |
| 1190 | if (sc->sc_dying) { |
| 1191 | mutex_exit(sc->sc_lock); |
| 1192 | return EIO; |
| 1193 | } |
| 1194 | |
| 1195 | /* Acquire device access lock. */ |
| 1196 | switch (rw) { |
| 1197 | case RW_WRITER: |
| 1198 | while (__predict_false(sc->sc_dvlock != 0)) { |
| 1199 | cv_wait(&sc->sc_lchan, sc->sc_lock); |
| 1200 | } |
| 1201 | sc->sc_dvlock = -1; |
| 1202 | break; |
| 1203 | case RW_READER: |
| 1204 | while (__predict_false(sc->sc_dvlock < 0)) { |
| 1205 | cv_wait(&sc->sc_lchan, sc->sc_lock); |
| 1206 | } |
| 1207 | sc->sc_dvlock++; |
| 1208 | break; |
| 1209 | default: |
| 1210 | panic("audio_enter" ); |
| 1211 | } |
| 1212 | |
| 1213 | *scp = sc; |
| 1214 | return 0; |
| 1215 | } |
| 1216 | |
| 1217 | /* |
| 1218 | * Release reference to device acquired with audio_enter(). |
| 1219 | */ |
| 1220 | static void |
| 1221 | audio_exit(struct audio_softc *sc) |
| 1222 | { |
| 1223 | |
| 1224 | KASSERT(mutex_owned(sc->sc_lock)); |
| 1225 | KASSERT(sc->sc_dvlock != 0); |
| 1226 | |
| 1227 | /* Release device level lock. */ |
| 1228 | if (__predict_false(sc->sc_dvlock < 0)) { |
| 1229 | sc->sc_dvlock = 0; |
| 1230 | } else { |
| 1231 | sc->sc_dvlock--; |
| 1232 | } |
| 1233 | cv_broadcast(&sc->sc_lchan); |
| 1234 | mutex_exit(sc->sc_lock); |
| 1235 | } |
| 1236 | |
| 1237 | /* |
| 1238 | * Wait for I/O to complete, releasing device lock. |
| 1239 | */ |
| 1240 | static int |
| 1241 | audio_waitio(struct audio_softc *sc, kcondvar_t *chan) |
| 1242 | { |
| 1243 | int error; |
| 1244 | krw_t rw; |
| 1245 | |
| 1246 | KASSERT(mutex_owned(sc->sc_lock)); |
| 1247 | |
| 1248 | /* Release device level lock while sleeping. */ |
| 1249 | if (__predict_false(sc->sc_dvlock < 0)) { |
| 1250 | sc->sc_dvlock = 0; |
| 1251 | rw = RW_WRITER; |
| 1252 | } else { |
| 1253 | KASSERT(sc->sc_dvlock > 0); |
| 1254 | sc->sc_dvlock--; |
| 1255 | rw = RW_READER; |
| 1256 | } |
| 1257 | cv_broadcast(&sc->sc_lchan); |
| 1258 | |
| 1259 | /* Wait for pending I/O to complete. */ |
| 1260 | error = cv_wait_sig(chan, sc->sc_lock); |
| 1261 | |
| 1262 | /* Re-acquire device level lock. */ |
| 1263 | if (__predict_false(rw == RW_WRITER)) { |
| 1264 | while (__predict_false(sc->sc_dvlock != 0)) { |
| 1265 | cv_wait(&sc->sc_lchan, sc->sc_lock); |
| 1266 | } |
| 1267 | sc->sc_dvlock = -1; |
| 1268 | } else { |
| 1269 | while (__predict_false(sc->sc_dvlock < 0)) { |
| 1270 | cv_wait(&sc->sc_lchan, sc->sc_lock); |
| 1271 | } |
| 1272 | sc->sc_dvlock++; |
| 1273 | } |
| 1274 | |
| 1275 | return error; |
| 1276 | } |
| 1277 | |
| 1278 | int |
| 1279 | audioopen(dev_t dev, int flags, int ifmt, struct lwp *l) |
| 1280 | { |
| 1281 | struct audio_softc *sc; |
| 1282 | int error; |
| 1283 | |
| 1284 | if ((error = audio_enter(dev, RW_WRITER, &sc)) != 0) |
| 1285 | return error; |
| 1286 | device_active(sc->dev, DVA_SYSTEM); |
| 1287 | switch (AUDIODEV(dev)) { |
| 1288 | case SOUND_DEVICE: |
| 1289 | case AUDIO_DEVICE: |
| 1290 | error = audio_open(dev, sc, flags, ifmt, l); |
| 1291 | break; |
| 1292 | case AUDIOCTL_DEVICE: |
| 1293 | error = 0; |
| 1294 | break; |
| 1295 | case MIXER_DEVICE: |
| 1296 | error = mixer_open(dev, sc, flags, ifmt, l); |
| 1297 | break; |
| 1298 | default: |
| 1299 | error = ENXIO; |
| 1300 | break; |
| 1301 | } |
| 1302 | audio_exit(sc); |
| 1303 | |
| 1304 | return error; |
| 1305 | } |
| 1306 | |
| 1307 | int |
| 1308 | audioclose(dev_t dev, int flags, int ifmt, struct lwp *l) |
| 1309 | { |
| 1310 | struct audio_softc *sc; |
| 1311 | int error; |
| 1312 | |
| 1313 | if ((error = audio_enter(dev, RW_WRITER, &sc)) != 0) |
| 1314 | return error; |
| 1315 | device_active(sc->dev, DVA_SYSTEM); |
| 1316 | switch (AUDIODEV(dev)) { |
| 1317 | case SOUND_DEVICE: |
| 1318 | case AUDIO_DEVICE: |
| 1319 | error = audio_close(sc, flags, ifmt, l); |
| 1320 | break; |
| 1321 | case MIXER_DEVICE: |
| 1322 | error = mixer_close(sc, flags, ifmt, l); |
| 1323 | break; |
| 1324 | case AUDIOCTL_DEVICE: |
| 1325 | error = 0; |
| 1326 | break; |
| 1327 | default: |
| 1328 | error = ENXIO; |
| 1329 | break; |
| 1330 | } |
| 1331 | audio_exit(sc); |
| 1332 | |
| 1333 | return error; |
| 1334 | } |
| 1335 | |
| 1336 | int |
| 1337 | audioread(dev_t dev, struct uio *uio, int ioflag) |
| 1338 | { |
| 1339 | struct audio_softc *sc; |
| 1340 | int error; |
| 1341 | |
| 1342 | if ((error = audio_enter(dev, RW_READER, &sc)) != 0) |
| 1343 | return error; |
| 1344 | switch (AUDIODEV(dev)) { |
| 1345 | case SOUND_DEVICE: |
| 1346 | case AUDIO_DEVICE: |
| 1347 | error = audio_read(sc, uio, ioflag); |
| 1348 | break; |
| 1349 | case AUDIOCTL_DEVICE: |
| 1350 | case MIXER_DEVICE: |
| 1351 | error = ENODEV; |
| 1352 | break; |
| 1353 | default: |
| 1354 | error = ENXIO; |
| 1355 | break; |
| 1356 | } |
| 1357 | audio_exit(sc); |
| 1358 | |
| 1359 | return error; |
| 1360 | } |
| 1361 | |
| 1362 | int |
| 1363 | audiowrite(dev_t dev, struct uio *uio, int ioflag) |
| 1364 | { |
| 1365 | struct audio_softc *sc; |
| 1366 | int error; |
| 1367 | |
| 1368 | if ((error = audio_enter(dev, RW_READER, &sc)) != 0) |
| 1369 | return error; |
| 1370 | switch (AUDIODEV(dev)) { |
| 1371 | case SOUND_DEVICE: |
| 1372 | case AUDIO_DEVICE: |
| 1373 | error = audio_write(sc, uio, ioflag); |
| 1374 | break; |
| 1375 | case AUDIOCTL_DEVICE: |
| 1376 | case MIXER_DEVICE: |
| 1377 | error = ENODEV; |
| 1378 | break; |
| 1379 | default: |
| 1380 | error = ENXIO; |
| 1381 | break; |
| 1382 | } |
| 1383 | audio_exit(sc); |
| 1384 | |
| 1385 | return error; |
| 1386 | } |
| 1387 | |
| 1388 | int |
| 1389 | audioioctl(dev_t dev, u_long cmd, void *addr, int flag, struct lwp *l) |
| 1390 | { |
| 1391 | struct audio_softc *sc; |
| 1392 | int error; |
| 1393 | krw_t rw; |
| 1394 | |
| 1395 | /* Figure out which lock type we need. */ |
| 1396 | switch (cmd) { |
| 1397 | case AUDIO_FLUSH: |
| 1398 | case AUDIO_SETINFO: |
| 1399 | case AUDIO_DRAIN: |
| 1400 | case AUDIO_SETFD: |
| 1401 | rw = RW_WRITER; |
| 1402 | break; |
| 1403 | default: |
| 1404 | rw = RW_READER; |
| 1405 | break; |
| 1406 | } |
| 1407 | |
| 1408 | if ((error = audio_enter(dev, rw, &sc)) != 0) |
| 1409 | return error; |
| 1410 | switch (AUDIODEV(dev)) { |
| 1411 | case SOUND_DEVICE: |
| 1412 | case AUDIO_DEVICE: |
| 1413 | case AUDIOCTL_DEVICE: |
| 1414 | device_active(sc->dev, DVA_SYSTEM); |
| 1415 | if (IOCGROUP(cmd) == IOCGROUP(AUDIO_MIXER_READ)) |
| 1416 | error = mixer_ioctl(sc, cmd, addr, flag, l); |
| 1417 | else |
| 1418 | error = audio_ioctl(sc, cmd, addr, flag, l); |
| 1419 | break; |
| 1420 | case MIXER_DEVICE: |
| 1421 | error = mixer_ioctl(sc, cmd, addr, flag, l); |
| 1422 | break; |
| 1423 | default: |
| 1424 | error = ENXIO; |
| 1425 | break; |
| 1426 | } |
| 1427 | audio_exit(sc); |
| 1428 | |
| 1429 | return error; |
| 1430 | } |
| 1431 | |
| 1432 | int |
| 1433 | audiopoll(dev_t dev, int events, struct lwp *l) |
| 1434 | { |
| 1435 | struct audio_softc *sc; |
| 1436 | int revents; |
| 1437 | |
| 1438 | /* Don't bother with device level lock here. */ |
| 1439 | sc = device_lookup_private(&audio_cd, AUDIOUNIT(dev)); |
| 1440 | if (sc == NULL) |
| 1441 | return ENXIO; |
| 1442 | mutex_enter(sc->sc_lock); |
| 1443 | if (sc->sc_dying) { |
| 1444 | mutex_exit(sc->sc_lock); |
| 1445 | return EIO; |
| 1446 | } |
| 1447 | switch (AUDIODEV(dev)) { |
| 1448 | case SOUND_DEVICE: |
| 1449 | case AUDIO_DEVICE: |
| 1450 | revents = audio_poll(sc, events, l); |
| 1451 | break; |
| 1452 | case AUDIOCTL_DEVICE: |
| 1453 | case MIXER_DEVICE: |
| 1454 | revents = 0; |
| 1455 | break; |
| 1456 | default: |
| 1457 | revents = POLLERR; |
| 1458 | break; |
| 1459 | } |
| 1460 | mutex_exit(sc->sc_lock); |
| 1461 | |
| 1462 | return revents; |
| 1463 | } |
| 1464 | |
| 1465 | int |
| 1466 | audiokqfilter(dev_t dev, struct knote *kn) |
| 1467 | { |
| 1468 | struct audio_softc *sc; |
| 1469 | int rv; |
| 1470 | |
| 1471 | /* Don't bother with device level lock here. */ |
| 1472 | sc = device_lookup_private(&audio_cd, AUDIOUNIT(dev)); |
| 1473 | if (sc == NULL) |
| 1474 | return ENXIO; |
| 1475 | mutex_enter(sc->sc_lock); |
| 1476 | if (sc->sc_dying) { |
| 1477 | mutex_exit(sc->sc_lock); |
| 1478 | return EIO; |
| 1479 | } |
| 1480 | switch (AUDIODEV(dev)) { |
| 1481 | case SOUND_DEVICE: |
| 1482 | case AUDIO_DEVICE: |
| 1483 | rv = audio_kqfilter(sc, kn); |
| 1484 | break; |
| 1485 | case AUDIOCTL_DEVICE: |
| 1486 | case MIXER_DEVICE: |
| 1487 | rv = 1; |
| 1488 | break; |
| 1489 | default: |
| 1490 | rv = 1; |
| 1491 | } |
| 1492 | mutex_exit(sc->sc_lock); |
| 1493 | |
| 1494 | return rv; |
| 1495 | } |
| 1496 | |
| 1497 | paddr_t |
| 1498 | audiommap(dev_t dev, off_t off, int prot) |
| 1499 | { |
| 1500 | struct audio_softc *sc; |
| 1501 | paddr_t error; |
| 1502 | |
| 1503 | /* |
| 1504 | * Acquire a reader lock. audio_mmap() will drop sc_lock |
| 1505 | * in order to allow the device's mmap routine to sleep. |
| 1506 | * Although not yet possible, we want to prevent memory |
| 1507 | * from being allocated or freed out from under us. |
| 1508 | */ |
| 1509 | if ((error = audio_enter(dev, RW_READER, &sc)) != 0) |
| 1510 | return 1; |
| 1511 | device_active(sc->dev, DVA_SYSTEM); /* XXXJDM */ |
| 1512 | switch (AUDIODEV(dev)) { |
| 1513 | case SOUND_DEVICE: |
| 1514 | case AUDIO_DEVICE: |
| 1515 | error = audio_mmap(sc, off, prot); |
| 1516 | break; |
| 1517 | case AUDIOCTL_DEVICE: |
| 1518 | case MIXER_DEVICE: |
| 1519 | error = -1; |
| 1520 | break; |
| 1521 | default: |
| 1522 | error = -1; |
| 1523 | break; |
| 1524 | } |
| 1525 | audio_exit(sc); |
| 1526 | return error; |
| 1527 | } |
| 1528 | |
| 1529 | /* |
| 1530 | * Audio driver |
| 1531 | */ |
| 1532 | void |
| 1533 | audio_init_ringbuffer(struct audio_softc *sc, struct audio_ringbuffer *rp, |
| 1534 | int mode) |
| 1535 | { |
| 1536 | int nblks; |
| 1537 | int blksize; |
| 1538 | |
| 1539 | blksize = rp->blksize; |
| 1540 | if (blksize < AUMINBLK) |
| 1541 | blksize = AUMINBLK; |
| 1542 | if (blksize > rp->s.bufsize / AUMINNOBLK) |
| 1543 | blksize = rp->s.bufsize / AUMINNOBLK; |
| 1544 | ROUNDSIZE(blksize); |
| 1545 | DPRINTF(("audio_init_ringbuffer: MI blksize=%d\n" , blksize)); |
| 1546 | if (sc->hw_if->round_blocksize) |
| 1547 | blksize = sc->hw_if->round_blocksize(sc->hw_hdl, blksize, |
| 1548 | mode, &rp->s.param); |
| 1549 | if (blksize <= 0) |
| 1550 | panic("audio_init_ringbuffer: blksize" ); |
| 1551 | nblks = rp->s.bufsize / blksize; |
| 1552 | |
| 1553 | DPRINTF(("audio_init_ringbuffer: final blksize=%d\n" , blksize)); |
| 1554 | rp->blksize = blksize; |
| 1555 | rp->maxblks = nblks; |
| 1556 | rp->s.end = rp->s.start + nblks * blksize; |
| 1557 | rp->s.outp = rp->s.inp = rp->s.start; |
| 1558 | rp->s.used = 0; |
| 1559 | rp->stamp = 0; |
| 1560 | rp->stamp_last = 0; |
| 1561 | rp->fstamp = 0; |
| 1562 | rp->drops = 0; |
| 1563 | rp->copying = false; |
| 1564 | rp->needfill = false; |
| 1565 | rp->mmapped = false; |
| 1566 | } |
| 1567 | |
| 1568 | int |
| 1569 | audio_initbufs(struct audio_softc *sc) |
| 1570 | { |
| 1571 | const struct audio_hw_if *hw; |
| 1572 | int error; |
| 1573 | |
| 1574 | DPRINTF(("audio_initbufs: mode=0x%x\n" , sc->sc_mode)); |
| 1575 | hw = sc->hw_if; |
| 1576 | if (audio_can_capture(sc)) { |
| 1577 | audio_init_ringbuffer(sc, &sc->sc_rr, AUMODE_RECORD); |
| 1578 | if (hw->init_input && (sc->sc_mode & AUMODE_RECORD)) { |
| 1579 | error = hw->init_input(sc->hw_hdl, sc->sc_rr.s.start, |
| 1580 | sc->sc_rr.s.end - sc->sc_rr.s.start); |
| 1581 | if (error) |
| 1582 | return error; |
| 1583 | } |
| 1584 | } |
| 1585 | |
| 1586 | if (audio_can_playback(sc)) { |
| 1587 | audio_init_ringbuffer(sc, &sc->sc_pr, AUMODE_PLAY); |
| 1588 | sc->sc_sil_count = 0; |
| 1589 | if (hw->init_output && (sc->sc_mode & AUMODE_PLAY)) { |
| 1590 | error = hw->init_output(sc->hw_hdl, sc->sc_pr.s.start, |
| 1591 | sc->sc_pr.s.end - sc->sc_pr.s.start); |
| 1592 | if (error) |
| 1593 | return error; |
| 1594 | } |
| 1595 | } |
| 1596 | |
| 1597 | #ifdef AUDIO_INTR_TIME |
| 1598 | #define double u_long |
| 1599 | if (audio_can_playback(sc)) { |
| 1600 | sc->sc_pnintr = 0; |
| 1601 | sc->sc_pblktime = (u_long)( |
| 1602 | (double)sc->sc_pr.blksize * 100000 / |
| 1603 | (double)(sc->sc_pparams.precision / NBBY * |
| 1604 | sc->sc_pparams.channels * |
| 1605 | sc->sc_pparams.sample_rate)) * 10; |
| 1606 | DPRINTF(("audio: play blktime = %lu for %d\n" , |
| 1607 | sc->sc_pblktime, sc->sc_pr.blksize)); |
| 1608 | } |
| 1609 | if (audio_can_capture(sc)) { |
| 1610 | sc->sc_rnintr = 0; |
| 1611 | sc->sc_rblktime = (u_long)( |
| 1612 | (double)sc->sc_rr.blksize * 100000 / |
| 1613 | (double)(sc->sc_rparams.precision / NBBY * |
| 1614 | sc->sc_rparams.channels * |
| 1615 | sc->sc_rparams.sample_rate)) * 10; |
| 1616 | DPRINTF(("audio: record blktime = %lu for %d\n" , |
| 1617 | sc->sc_rblktime, sc->sc_rr.blksize)); |
| 1618 | } |
| 1619 | #undef double |
| 1620 | #endif |
| 1621 | |
| 1622 | return 0; |
| 1623 | } |
| 1624 | |
| 1625 | void |
| 1626 | audio_calcwater(struct audio_softc *sc) |
| 1627 | { |
| 1628 | |
| 1629 | /* set high at 100% */ |
| 1630 | if (audio_can_playback(sc)) { |
| 1631 | sc->sc_pr.usedhigh = |
| 1632 | sc->sc_pustream->end - sc->sc_pustream->start; |
| 1633 | /* set low at 75% of usedhigh */ |
| 1634 | sc->sc_pr.usedlow = sc->sc_pr.usedhigh * 3 / 4; |
| 1635 | if (sc->sc_pr.usedlow == sc->sc_pr.usedhigh) |
| 1636 | sc->sc_pr.usedlow -= sc->sc_pr.blksize; |
| 1637 | } |
| 1638 | |
| 1639 | if (audio_can_capture(sc)) { |
| 1640 | sc->sc_rr.usedhigh = |
| 1641 | sc->sc_rustream->end - sc->sc_rustream->start - |
| 1642 | sc->sc_rr.blksize; |
| 1643 | sc->sc_rr.usedlow = 0; |
| 1644 | DPRINTF(("%s: plow=%d phigh=%d rlow=%d rhigh=%d\n" , __func__, |
| 1645 | sc->sc_pr.usedlow, sc->sc_pr.usedhigh, |
| 1646 | sc->sc_rr.usedlow, sc->sc_rr.usedhigh)); |
| 1647 | } |
| 1648 | } |
| 1649 | |
| 1650 | int |
| 1651 | audio_open(dev_t dev, struct audio_softc *sc, int flags, int ifmt, |
| 1652 | struct lwp *l) |
| 1653 | { |
| 1654 | int error; |
| 1655 | u_int mode; |
| 1656 | const struct audio_hw_if *hw; |
| 1657 | |
| 1658 | KASSERT(mutex_owned(sc->sc_lock)); |
| 1659 | |
| 1660 | hw = sc->hw_if; |
| 1661 | if (hw == NULL) |
| 1662 | return ENXIO; |
| 1663 | |
| 1664 | DPRINTF(("audio_open: flags=0x%x sc=%p hdl=%p\n" , |
| 1665 | flags, sc, sc->hw_hdl)); |
| 1666 | |
| 1667 | if (((flags & FREAD) && (sc->sc_open & AUOPEN_READ)) || |
| 1668 | ((flags & FWRITE) && (sc->sc_open & AUOPEN_WRITE))) |
| 1669 | return EBUSY; |
| 1670 | |
| 1671 | if (hw->open != NULL) { |
| 1672 | mutex_enter(sc->sc_intr_lock); |
| 1673 | error = hw->open(sc->hw_hdl, flags); |
| 1674 | mutex_exit(sc->sc_intr_lock); |
| 1675 | if (error) |
| 1676 | return error; |
| 1677 | } |
| 1678 | |
| 1679 | sc->sc_async_audio = 0; |
| 1680 | sc->sc_sil_count = 0; |
| 1681 | sc->sc_rbus = false; |
| 1682 | sc->sc_pbus = false; |
| 1683 | sc->sc_eof = 0; |
| 1684 | sc->sc_playdrop = 0; |
| 1685 | |
| 1686 | mutex_enter(sc->sc_intr_lock); |
| 1687 | sc->sc_full_duplex = |
| 1688 | (flags & (FWRITE|FREAD)) == (FWRITE|FREAD) && |
| 1689 | (audio_get_props(sc) & AUDIO_PROP_FULLDUPLEX); |
| 1690 | mutex_exit(sc->sc_intr_lock); |
| 1691 | |
| 1692 | mode = 0; |
| 1693 | if (flags & FREAD) { |
| 1694 | sc->sc_open |= AUOPEN_READ; |
| 1695 | mode |= AUMODE_RECORD; |
| 1696 | } |
| 1697 | if (flags & FWRITE) { |
| 1698 | sc->sc_open |= AUOPEN_WRITE; |
| 1699 | mode |= AUMODE_PLAY | AUMODE_PLAY_ALL; |
| 1700 | } |
| 1701 | |
| 1702 | /* |
| 1703 | * Multiplex device: /dev/audio (MU-Law) and /dev/sound (linear) |
| 1704 | * The /dev/audio is always (re)set to 8-bit MU-Law mono |
| 1705 | * For the other devices, you get what they were last set to. |
| 1706 | */ |
| 1707 | if (ISDEVAUDIO(dev)) { |
| 1708 | error = audio_set_defaults(sc, mode); |
| 1709 | } else { |
| 1710 | struct audio_info ai; |
| 1711 | |
| 1712 | AUDIO_INITINFO(&ai); |
| 1713 | ai.mode = mode; |
| 1714 | error = audiosetinfo(sc, &ai); |
| 1715 | } |
| 1716 | if (error) |
| 1717 | goto bad; |
| 1718 | |
| 1719 | #ifdef DIAGNOSTIC |
| 1720 | /* |
| 1721 | * Sample rate and precision are supposed to be set to proper |
| 1722 | * default values by the hardware driver, so that it may give |
| 1723 | * us these values. |
| 1724 | */ |
| 1725 | if (sc->sc_rparams.precision == 0 || sc->sc_pparams.precision == 0) { |
| 1726 | printf("audio_open: 0 precision\n" ); |
| 1727 | return EINVAL; |
| 1728 | } |
| 1729 | #endif |
| 1730 | |
| 1731 | /* audio_close() decreases sc_pr.usedlow, recalculate here */ |
| 1732 | audio_calcwater(sc); |
| 1733 | |
| 1734 | DPRINTF(("audio_open: done sc_mode = 0x%x\n" , sc->sc_mode)); |
| 1735 | |
| 1736 | return 0; |
| 1737 | |
| 1738 | bad: |
| 1739 | mutex_enter(sc->sc_intr_lock); |
| 1740 | if (hw->close != NULL) |
| 1741 | hw->close(sc->hw_hdl); |
| 1742 | sc->sc_open = 0; |
| 1743 | sc->sc_mode = 0; |
| 1744 | mutex_exit(sc->sc_intr_lock); |
| 1745 | sc->sc_full_duplex = 0; |
| 1746 | return error; |
| 1747 | } |
| 1748 | |
| 1749 | /* |
| 1750 | * Must be called from task context. |
| 1751 | */ |
| 1752 | void |
| 1753 | audio_init_record(struct audio_softc *sc) |
| 1754 | { |
| 1755 | |
| 1756 | KASSERT(mutex_owned(sc->sc_lock)); |
| 1757 | |
| 1758 | mutex_enter(sc->sc_intr_lock); |
| 1759 | if (sc->hw_if->speaker_ctl && |
| 1760 | (!sc->sc_full_duplex || (sc->sc_mode & AUMODE_PLAY) == 0)) |
| 1761 | sc->hw_if->speaker_ctl(sc->hw_hdl, SPKR_OFF); |
| 1762 | mutex_exit(sc->sc_intr_lock); |
| 1763 | } |
| 1764 | |
| 1765 | /* |
| 1766 | * Must be called from task context. |
| 1767 | */ |
| 1768 | void |
| 1769 | audio_init_play(struct audio_softc *sc) |
| 1770 | { |
| 1771 | |
| 1772 | KASSERT(mutex_owned(sc->sc_lock)); |
| 1773 | |
| 1774 | mutex_enter(sc->sc_intr_lock); |
| 1775 | sc->sc_wstamp = sc->sc_pr.stamp; |
| 1776 | if (sc->hw_if->speaker_ctl) |
| 1777 | sc->hw_if->speaker_ctl(sc->hw_hdl, SPKR_ON); |
| 1778 | mutex_exit(sc->sc_intr_lock); |
| 1779 | } |
| 1780 | |
| 1781 | int |
| 1782 | audio_drain(struct audio_softc *sc) |
| 1783 | { |
| 1784 | struct audio_ringbuffer *cb; |
| 1785 | int error, drops; |
| 1786 | int i, used; |
| 1787 | |
| 1788 | KASSERT(mutex_owned(sc->sc_lock)); |
| 1789 | KASSERT(mutex_owned(sc->sc_intr_lock)); |
| 1790 | |
| 1791 | DPRINTF(("audio_drain: enter busy=%d\n" , sc->sc_pbus)); |
| 1792 | cb = &sc->sc_pr; |
| 1793 | if (cb->mmapped) |
| 1794 | return 0; |
| 1795 | |
| 1796 | used = audio_stream_get_used(&sc->sc_pr.s); |
| 1797 | for (i = 0; i < sc->sc_npfilters; i++) |
| 1798 | used += audio_stream_get_used(&sc->sc_pstreams[i]); |
| 1799 | if (used <= 0) |
| 1800 | return 0; |
| 1801 | |
| 1802 | if (!sc->sc_pbus) { |
| 1803 | /* We've never started playing, probably because the |
| 1804 | * block was too short. Pad it and start now. |
| 1805 | */ |
| 1806 | int cc; |
| 1807 | uint8_t *inp = cb->s.inp; |
| 1808 | |
| 1809 | cc = cb->blksize - (inp - cb->s.start) % cb->blksize; |
| 1810 | audio_fill_silence(&cb->s.param, inp, cc); |
| 1811 | cb->s.inp = audio_stream_add_inp(&cb->s, inp, cc); |
| 1812 | error = audiostartp(sc); |
| 1813 | if (error) |
| 1814 | return error; |
| 1815 | } |
| 1816 | /* |
| 1817 | * Play until a silence block has been played, then we |
| 1818 | * know all has been drained. |
| 1819 | * XXX This should be done some other way to avoid |
| 1820 | * playing silence. |
| 1821 | */ |
| 1822 | #ifdef DIAGNOSTIC |
| 1823 | if (cb->copying) { |
| 1824 | printf("audio_drain: copying in progress!?!\n" ); |
| 1825 | cb->copying = false; |
| 1826 | } |
| 1827 | #endif |
| 1828 | drops = cb->drops; |
| 1829 | error = 0; |
| 1830 | while (cb->drops == drops && !error) { |
| 1831 | DPRINTF(("audio_drain: used=%d, drops=%ld\n" , |
| 1832 | audio_stream_get_used(&sc->sc_pr.s), cb->drops)); |
| 1833 | mutex_exit(sc->sc_intr_lock); |
| 1834 | error = audio_waitio(sc, &sc->sc_wchan); |
| 1835 | mutex_enter(sc->sc_intr_lock); |
| 1836 | if (sc->sc_dying) |
| 1837 | error = EIO; |
| 1838 | } |
| 1839 | return error; |
| 1840 | } |
| 1841 | |
| 1842 | /* |
| 1843 | * Close an audio chip. |
| 1844 | */ |
| 1845 | /* ARGSUSED */ |
| 1846 | int |
| 1847 | audio_close(struct audio_softc *sc, int flags, int ifmt, struct lwp *l) |
| 1848 | { |
| 1849 | const struct audio_hw_if *hw; |
| 1850 | |
| 1851 | KASSERT(mutex_owned(sc->sc_lock)); |
| 1852 | |
| 1853 | DPRINTF(("audio_close: sc=%p\n" , sc)); |
| 1854 | hw = sc->hw_if; |
| 1855 | mutex_enter(sc->sc_intr_lock); |
| 1856 | /* Stop recording. */ |
| 1857 | if ((flags & FREAD) && sc->sc_rbus) { |
| 1858 | /* |
| 1859 | * XXX Some drivers (e.g. SB) use the same routine |
| 1860 | * to halt input and output so don't halt input if |
| 1861 | * in full duplex mode. These drivers should be fixed. |
| 1862 | */ |
| 1863 | if (!sc->sc_full_duplex || hw->halt_input != hw->halt_output) |
| 1864 | hw->halt_input(sc->hw_hdl); |
| 1865 | sc->sc_rbus = false; |
| 1866 | } |
| 1867 | /* |
| 1868 | * Block until output drains, but allow ^C interrupt. |
| 1869 | */ |
| 1870 | sc->sc_pr.usedlow = sc->sc_pr.blksize; /* avoid excessive wakeups */ |
| 1871 | /* |
| 1872 | * If there is pending output, let it drain (unless |
| 1873 | * the output is paused). |
| 1874 | */ |
| 1875 | if ((flags & FWRITE) && sc->sc_pbus) { |
| 1876 | if (!sc->sc_pr.pause && !audio_drain(sc) && hw->drain) |
| 1877 | (void)hw->drain(sc->hw_hdl); |
| 1878 | hw->halt_output(sc->hw_hdl); |
| 1879 | sc->sc_pbus = false; |
| 1880 | } |
| 1881 | if (hw->close != NULL) |
| 1882 | hw->close(sc->hw_hdl); |
| 1883 | sc->sc_open = 0; |
| 1884 | sc->sc_mode = 0; |
| 1885 | sc->sc_full_duplex = 0; |
| 1886 | mutex_exit(sc->sc_intr_lock); |
| 1887 | sc->sc_async_audio = 0; |
| 1888 | |
| 1889 | return 0; |
| 1890 | } |
| 1891 | |
| 1892 | int |
| 1893 | audio_read(struct audio_softc *sc, struct uio *uio, int ioflag) |
| 1894 | { |
| 1895 | struct audio_ringbuffer *cb; |
| 1896 | const uint8_t *outp; |
| 1897 | uint8_t *inp; |
| 1898 | int error, used, cc, n; |
| 1899 | |
| 1900 | KASSERT(mutex_owned(sc->sc_lock)); |
| 1901 | |
| 1902 | cb = &sc->sc_rr; |
| 1903 | if (cb->mmapped) |
| 1904 | return EINVAL; |
| 1905 | |
| 1906 | DPRINTFN(1,("audio_read: cc=%zu mode=%d\n" , |
| 1907 | uio->uio_resid, sc->sc_mode)); |
| 1908 | |
| 1909 | #ifdef AUDIO_PM_IDLE |
| 1910 | if (device_is_active(&sc->dev) || sc->sc_idle) |
| 1911 | device_active(&sc->dev, DVA_SYSTEM); |
| 1912 | #endif |
| 1913 | |
| 1914 | error = 0; |
| 1915 | /* |
| 1916 | * If hardware is half-duplex and currently playing, return |
| 1917 | * silence blocks based on the number of blocks we have output. |
| 1918 | */ |
| 1919 | if (!sc->sc_full_duplex && (sc->sc_mode & AUMODE_PLAY)) { |
| 1920 | while (uio->uio_resid > 0 && !error) { |
| 1921 | for(;;) { |
| 1922 | /* |
| 1923 | * No need to lock, as any wakeup will be |
| 1924 | * held for us while holding sc_lock. |
| 1925 | */ |
| 1926 | cc = sc->sc_pr.stamp - sc->sc_wstamp; |
| 1927 | if (cc > 0) |
| 1928 | break; |
| 1929 | DPRINTF(("audio_read: stamp=%lu, wstamp=%lu\n" , |
| 1930 | sc->sc_pr.stamp, sc->sc_wstamp)); |
| 1931 | if (ioflag & IO_NDELAY) |
| 1932 | return EWOULDBLOCK; |
| 1933 | error = audio_waitio(sc, &sc->sc_rchan); |
| 1934 | if (sc->sc_dying) |
| 1935 | error = EIO; |
| 1936 | if (error) |
| 1937 | return error; |
| 1938 | } |
| 1939 | |
| 1940 | if (uio->uio_resid < cc) |
| 1941 | cc = uio->uio_resid; |
| 1942 | DPRINTFN(1,("audio_read: reading in write mode, " |
| 1943 | "cc=%d\n" , cc)); |
| 1944 | error = audio_silence_copyout(sc, cc, uio); |
| 1945 | sc->sc_wstamp += cc; |
| 1946 | } |
| 1947 | return error; |
| 1948 | } |
| 1949 | |
| 1950 | mutex_enter(sc->sc_intr_lock); |
| 1951 | while (uio->uio_resid > 0 && !error) { |
| 1952 | while ((used = audio_stream_get_used(sc->sc_rustream)) <= 0) { |
| 1953 | if (!sc->sc_rbus && !sc->sc_rr.pause) |
| 1954 | error = audiostartr(sc); |
| 1955 | mutex_exit(sc->sc_intr_lock); |
| 1956 | if (error) |
| 1957 | return error; |
| 1958 | if (ioflag & IO_NDELAY) |
| 1959 | return EWOULDBLOCK; |
| 1960 | DPRINTFN(2, ("audio_read: sleep used=%d\n" , used)); |
| 1961 | error = audio_waitio(sc, &sc->sc_rchan); |
| 1962 | if (sc->sc_dying) |
| 1963 | error = EIO; |
| 1964 | if (error) |
| 1965 | return error; |
| 1966 | mutex_enter(sc->sc_intr_lock); |
| 1967 | } |
| 1968 | |
| 1969 | outp = sc->sc_rustream->outp; |
| 1970 | inp = sc->sc_rustream->inp; |
| 1971 | cb->copying = true; |
| 1972 | |
| 1973 | /* |
| 1974 | * cc is the amount of data in the sc_rustream excluding |
| 1975 | * wrapped data. Note the tricky case of inp == outp, which |
| 1976 | * must mean the buffer is full, not empty, because used > 0. |
| 1977 | */ |
| 1978 | cc = outp < inp ? inp - outp :sc->sc_rustream->end - outp; |
| 1979 | DPRINTFN(1,("audio_read: outp=%p, cc=%d\n" , outp, cc)); |
| 1980 | |
| 1981 | n = uio->uio_resid; |
| 1982 | mutex_exit(sc->sc_intr_lock); |
| 1983 | mutex_exit(sc->sc_lock); |
| 1984 | error = uiomove(__UNCONST(outp), cc, uio); |
| 1985 | mutex_enter(sc->sc_lock); |
| 1986 | mutex_enter(sc->sc_intr_lock); |
| 1987 | n -= uio->uio_resid; /* number of bytes actually moved */ |
| 1988 | |
| 1989 | sc->sc_rustream->outp = audio_stream_add_outp |
| 1990 | (sc->sc_rustream, outp, n); |
| 1991 | cb->copying = false; |
| 1992 | } |
| 1993 | mutex_exit(sc->sc_intr_lock); |
| 1994 | return error; |
| 1995 | } |
| 1996 | |
| 1997 | void |
| 1998 | audio_clear(struct audio_softc *sc) |
| 1999 | { |
| 2000 | |
| 2001 | KASSERT(mutex_owned(sc->sc_intr_lock)); |
| 2002 | |
| 2003 | if (sc->sc_rbus) { |
| 2004 | cv_broadcast(&sc->sc_rchan); |
| 2005 | sc->hw_if->halt_input(sc->hw_hdl); |
| 2006 | sc->sc_rbus = false; |
| 2007 | sc->sc_rr.pause = false; |
| 2008 | } |
| 2009 | if (sc->sc_pbus) { |
| 2010 | cv_broadcast(&sc->sc_wchan); |
| 2011 | sc->hw_if->halt_output(sc->hw_hdl); |
| 2012 | sc->sc_pbus = false; |
| 2013 | sc->sc_pr.pause = false; |
| 2014 | } |
| 2015 | } |
| 2016 | |
| 2017 | void |
| 2018 | audio_clear_intr_unlocked(struct audio_softc *sc) |
| 2019 | { |
| 2020 | |
| 2021 | mutex_enter(sc->sc_intr_lock); |
| 2022 | audio_clear(sc); |
| 2023 | mutex_exit(sc->sc_intr_lock); |
| 2024 | } |
| 2025 | |
| 2026 | void |
| 2027 | audio_calc_blksize(struct audio_softc *sc, int mode) |
| 2028 | { |
| 2029 | const audio_params_t *parm; |
| 2030 | struct audio_ringbuffer *rb; |
| 2031 | |
| 2032 | if (sc->sc_blkset) |
| 2033 | return; |
| 2034 | |
| 2035 | if (mode == AUMODE_PLAY) { |
| 2036 | rb = &sc->sc_pr; |
| 2037 | parm = &rb->s.param; |
| 2038 | } else { |
| 2039 | rb = &sc->sc_rr; |
| 2040 | parm = &rb->s.param; |
| 2041 | } |
| 2042 | |
| 2043 | rb->blksize = parm->sample_rate * audio_blk_ms / 1000 * |
| 2044 | parm->channels * parm->precision / NBBY; |
| 2045 | |
| 2046 | DPRINTF(("audio_calc_blksize: %s blksize=%d\n" , |
| 2047 | mode == AUMODE_PLAY ? "play" : "record" , rb->blksize)); |
| 2048 | } |
| 2049 | |
| 2050 | void |
| 2051 | audio_fill_silence(struct audio_params *params, uint8_t *p, int n) |
| 2052 | { |
| 2053 | uint8_t auzero0, auzero1; |
| 2054 | int nfill; |
| 2055 | |
| 2056 | auzero1 = 0; /* initialize to please gcc */ |
| 2057 | nfill = 1; |
| 2058 | switch (params->encoding) { |
| 2059 | case AUDIO_ENCODING_ULAW: |
| 2060 | auzero0 = 0x7f; |
| 2061 | break; |
| 2062 | case AUDIO_ENCODING_ALAW: |
| 2063 | auzero0 = 0x55; |
| 2064 | break; |
| 2065 | case AUDIO_ENCODING_MPEG_L1_STREAM: |
| 2066 | case AUDIO_ENCODING_MPEG_L1_PACKETS: |
| 2067 | case AUDIO_ENCODING_MPEG_L1_SYSTEM: |
| 2068 | case AUDIO_ENCODING_MPEG_L2_STREAM: |
| 2069 | case AUDIO_ENCODING_MPEG_L2_PACKETS: |
| 2070 | case AUDIO_ENCODING_MPEG_L2_SYSTEM: |
| 2071 | case AUDIO_ENCODING_AC3: |
| 2072 | case AUDIO_ENCODING_ADPCM: /* is this right XXX */ |
| 2073 | case AUDIO_ENCODING_SLINEAR_LE: |
| 2074 | case AUDIO_ENCODING_SLINEAR_BE: |
| 2075 | auzero0 = 0;/* fortunately this works for any number of bits */ |
| 2076 | break; |
| 2077 | case AUDIO_ENCODING_ULINEAR_LE: |
| 2078 | case AUDIO_ENCODING_ULINEAR_BE: |
| 2079 | if (params->precision > 8) { |
| 2080 | nfill = (params->precision + NBBY - 1)/ NBBY; |
| 2081 | auzero0 = 0x80; |
| 2082 | auzero1 = 0; |
| 2083 | } else |
| 2084 | auzero0 = 0x80; |
| 2085 | break; |
| 2086 | default: |
| 2087 | DPRINTF(("audio: bad encoding %d\n" , params->encoding)); |
| 2088 | auzero0 = 0; |
| 2089 | break; |
| 2090 | } |
| 2091 | if (nfill == 1) { |
| 2092 | while (--n >= 0) |
| 2093 | *p++ = auzero0; /* XXX memset */ |
| 2094 | } else /* nfill must no longer be 2 */ { |
| 2095 | if (params->encoding == AUDIO_ENCODING_ULINEAR_LE) { |
| 2096 | int k = nfill; |
| 2097 | while (--k > 0) |
| 2098 | *p++ = auzero1; |
| 2099 | n -= nfill - 1; |
| 2100 | } |
| 2101 | while (n >= nfill) { |
| 2102 | int k = nfill; |
| 2103 | *p++ = auzero0; |
| 2104 | while (--k > 0) |
| 2105 | *p++ = auzero1; |
| 2106 | |
| 2107 | n -= nfill; |
| 2108 | } |
| 2109 | if (n-- > 0) /* XXX must be 1 - DIAGNOSTIC check? */ |
| 2110 | *p++ = auzero0; |
| 2111 | } |
| 2112 | } |
| 2113 | |
| 2114 | int |
| 2115 | audio_silence_copyout(struct audio_softc *sc, int n, struct uio *uio) |
| 2116 | { |
| 2117 | uint8_t zerobuf[128]; |
| 2118 | int error; |
| 2119 | int k; |
| 2120 | |
| 2121 | audio_fill_silence(&sc->sc_rparams, zerobuf, sizeof zerobuf); |
| 2122 | |
| 2123 | error = 0; |
| 2124 | while (n > 0 && uio->uio_resid > 0 && !error) { |
| 2125 | k = min(n, min(uio->uio_resid, sizeof zerobuf)); |
| 2126 | mutex_exit(sc->sc_lock); |
| 2127 | error = uiomove(zerobuf, k, uio); |
| 2128 | mutex_enter(sc->sc_lock); |
| 2129 | n -= k; |
| 2130 | } |
| 2131 | |
| 2132 | return error; |
| 2133 | } |
| 2134 | |
| 2135 | static int |
| 2136 | uio_fetcher_fetch_to(struct audio_softc *sc, stream_fetcher_t *self, |
| 2137 | audio_stream_t *p, int max_used) |
| 2138 | { |
| 2139 | uio_fetcher_t *this; |
| 2140 | int size; |
| 2141 | int stream_space; |
| 2142 | int error; |
| 2143 | |
| 2144 | KASSERT(mutex_owned(sc->sc_lock)); |
| 2145 | KASSERT(!cpu_intr_p()); |
| 2146 | KASSERT(!cpu_softintr_p()); |
| 2147 | |
| 2148 | this = (uio_fetcher_t *)self; |
| 2149 | this->last_used = audio_stream_get_used(p); |
| 2150 | if (this->last_used >= this->usedhigh) |
| 2151 | return 0; |
| 2152 | /* |
| 2153 | * uio_fetcher ignores max_used and move the data as |
| 2154 | * much as possible in order to return the correct value |
| 2155 | * for audio_prinfo::seek and kfilters. |
| 2156 | */ |
| 2157 | stream_space = audio_stream_get_space(p); |
| 2158 | size = min(this->uio->uio_resid, stream_space); |
| 2159 | |
| 2160 | /* the first fragment of the space */ |
| 2161 | stream_space = p->end - p->inp; |
| 2162 | if (stream_space >= size) { |
| 2163 | mutex_exit(sc->sc_lock); |
| 2164 | error = uiomove(p->inp, size, this->uio); |
| 2165 | mutex_enter(sc->sc_lock); |
| 2166 | if (error) |
| 2167 | return error; |
| 2168 | p->inp = audio_stream_add_inp(p, p->inp, size); |
| 2169 | } else { |
| 2170 | mutex_exit(sc->sc_lock); |
| 2171 | error = uiomove(p->inp, stream_space, this->uio); |
| 2172 | mutex_enter(sc->sc_lock); |
| 2173 | if (error) |
| 2174 | return error; |
| 2175 | p->inp = audio_stream_add_inp(p, p->inp, stream_space); |
| 2176 | mutex_exit(sc->sc_lock); |
| 2177 | error = uiomove(p->start, size - stream_space, this->uio); |
| 2178 | mutex_enter(sc->sc_lock); |
| 2179 | if (error) |
| 2180 | return error; |
| 2181 | p->inp = audio_stream_add_inp(p, p->inp, size - stream_space); |
| 2182 | } |
| 2183 | this->last_used = audio_stream_get_used(p); |
| 2184 | return 0; |
| 2185 | } |
| 2186 | |
| 2187 | static int |
| 2188 | null_fetcher_fetch_to(struct audio_softc *sc, stream_fetcher_t *self, |
| 2189 | audio_stream_t *p, int max_used) |
| 2190 | { |
| 2191 | |
| 2192 | return 0; |
| 2193 | } |
| 2194 | |
| 2195 | static void |
| 2196 | uio_fetcher_ctor(uio_fetcher_t *this, struct uio *u, int h) |
| 2197 | { |
| 2198 | |
| 2199 | this->base.fetch_to = uio_fetcher_fetch_to; |
| 2200 | this->uio = u; |
| 2201 | this->usedhigh = h; |
| 2202 | } |
| 2203 | |
| 2204 | int |
| 2205 | audio_write(struct audio_softc *sc, struct uio *uio, int ioflag) |
| 2206 | { |
| 2207 | uio_fetcher_t ufetcher; |
| 2208 | audio_stream_t stream; |
| 2209 | struct audio_ringbuffer *cb; |
| 2210 | stream_fetcher_t *fetcher; |
| 2211 | stream_filter_t *filter; |
| 2212 | uint8_t *inp, *einp; |
| 2213 | int saveerror, error, n, cc, used; |
| 2214 | |
| 2215 | KASSERT(mutex_owned(sc->sc_lock)); |
| 2216 | |
| 2217 | DPRINTFN(2,("audio_write: sc=%p count=%zu used=%d(hi=%d)\n" , |
| 2218 | sc, uio->uio_resid, audio_stream_get_used(sc->sc_pustream), |
| 2219 | sc->sc_pr.usedhigh)); |
| 2220 | cb = &sc->sc_pr; |
| 2221 | if (cb->mmapped) |
| 2222 | return EINVAL; |
| 2223 | |
| 2224 | if (uio->uio_resid == 0) { |
| 2225 | sc->sc_eof++; |
| 2226 | return 0; |
| 2227 | } |
| 2228 | |
| 2229 | #ifdef AUDIO_PM_IDLE |
| 2230 | if (device_is_active(&sc->dev) || sc->sc_idle) |
| 2231 | device_active(&sc->dev, DVA_SYSTEM); |
| 2232 | #endif |
| 2233 | |
| 2234 | /* |
| 2235 | * If half-duplex and currently recording, throw away data. |
| 2236 | */ |
| 2237 | if (!sc->sc_full_duplex && |
| 2238 | (sc->sc_mode & AUMODE_RECORD)) { |
| 2239 | uio->uio_offset += uio->uio_resid; |
| 2240 | uio->uio_resid = 0; |
| 2241 | DPRINTF(("audio_write: half-dpx read busy\n" )); |
| 2242 | return 0; |
| 2243 | } |
| 2244 | |
| 2245 | if (!(sc->sc_mode & AUMODE_PLAY_ALL) && sc->sc_playdrop > 0) { |
| 2246 | n = min(sc->sc_playdrop, uio->uio_resid); |
| 2247 | DPRINTF(("audio_write: playdrop %d\n" , n)); |
| 2248 | uio->uio_offset += n; |
| 2249 | uio->uio_resid -= n; |
| 2250 | sc->sc_playdrop -= n; |
| 2251 | if (uio->uio_resid == 0) |
| 2252 | return 0; |
| 2253 | } |
| 2254 | |
| 2255 | /** |
| 2256 | * setup filter pipeline |
| 2257 | */ |
| 2258 | uio_fetcher_ctor(&ufetcher, uio, cb->usedhigh); |
| 2259 | if (sc->sc_npfilters > 0) { |
| 2260 | fetcher = &sc->sc_pfilters[sc->sc_npfilters - 1]->base; |
| 2261 | } else { |
| 2262 | fetcher = &ufetcher.base; |
| 2263 | } |
| 2264 | |
| 2265 | error = 0; |
| 2266 | mutex_enter(sc->sc_intr_lock); |
| 2267 | while (uio->uio_resid > 0 && !error) { |
| 2268 | /* wait if the first buffer is occupied */ |
| 2269 | while ((used = audio_stream_get_used(sc->sc_pustream)) |
| 2270 | >= cb->usedhigh) { |
| 2271 | DPRINTFN(2, ("audio_write: sleep used=%d lowat=%d " |
| 2272 | "hiwat=%d\n" , used, |
| 2273 | cb->usedlow, cb->usedhigh)); |
| 2274 | mutex_exit(sc->sc_intr_lock); |
| 2275 | if (ioflag & IO_NDELAY) |
| 2276 | return EWOULDBLOCK; |
| 2277 | error = audio_waitio(sc, &sc->sc_wchan); |
| 2278 | if (sc->sc_dying) |
| 2279 | error = EIO; |
| 2280 | if (error) |
| 2281 | return error; |
| 2282 | mutex_enter(sc->sc_intr_lock); |
| 2283 | } |
| 2284 | inp = cb->s.inp; |
| 2285 | cb->copying = true; |
| 2286 | stream = cb->s; |
| 2287 | used = stream.used; |
| 2288 | |
| 2289 | /* Write to the sc_pustream as much as possible. */ |
| 2290 | mutex_exit(sc->sc_intr_lock); |
| 2291 | if (sc->sc_npfilters > 0) { |
| 2292 | filter = sc->sc_pfilters[0]; |
| 2293 | filter->set_fetcher(filter, &ufetcher.base); |
| 2294 | fetcher = &sc->sc_pfilters[sc->sc_npfilters - 1]->base; |
| 2295 | cc = cb->blksize * 2; |
| 2296 | error = fetcher->fetch_to(sc, fetcher, &stream, cc); |
| 2297 | if (error != 0) { |
| 2298 | fetcher = &ufetcher.base; |
| 2299 | cc = sc->sc_pustream->end - sc->sc_pustream->start; |
| 2300 | error = fetcher->fetch_to(sc, fetcher, |
| 2301 | sc->sc_pustream, cc); |
| 2302 | } |
| 2303 | } else { |
| 2304 | fetcher = &ufetcher.base; |
| 2305 | cc = stream.end - stream.start; |
| 2306 | error = fetcher->fetch_to(sc, fetcher, &stream, cc); |
| 2307 | } |
| 2308 | mutex_enter(sc->sc_intr_lock); |
| 2309 | if (sc->sc_npfilters > 0) { |
| 2310 | cb->fstamp += ufetcher.last_used |
| 2311 | - audio_stream_get_used(sc->sc_pustream); |
| 2312 | } |
| 2313 | cb->s.used += stream.used - used; |
| 2314 | cb->s.inp = stream.inp; |
| 2315 | einp = cb->s.inp; |
| 2316 | |
| 2317 | /* |
| 2318 | * This is a very suboptimal way of keeping track of |
| 2319 | * silence in the buffer, but it is simple. |
| 2320 | */ |
| 2321 | sc->sc_sil_count = 0; |
| 2322 | |
| 2323 | /* |
| 2324 | * If the interrupt routine wants the last block filled AND |
| 2325 | * the copy did not fill the last block completely it needs to |
| 2326 | * be padded. |
| 2327 | */ |
| 2328 | if (cb->needfill && inp < einp && |
| 2329 | (inp - cb->s.start) / cb->blksize == |
| 2330 | (einp - cb->s.start) / cb->blksize) { |
| 2331 | /* Figure out how many bytes to a block boundary. */ |
| 2332 | cc = cb->blksize - (einp - cb->s.start) % cb->blksize; |
| 2333 | DPRINTF(("audio_write: partial fill %d\n" , cc)); |
| 2334 | } else |
| 2335 | cc = 0; |
| 2336 | cb->needfill = false; |
| 2337 | cb->copying = false; |
| 2338 | if (!sc->sc_pbus && !cb->pause) { |
| 2339 | saveerror = error; |
| 2340 | error = audiostartp(sc); |
| 2341 | if (saveerror != 0) { |
| 2342 | /* Report the first error that occurred. */ |
| 2343 | error = saveerror; |
| 2344 | } |
| 2345 | } |
| 2346 | if (cc != 0) { |
| 2347 | DPRINTFN(1, ("audio_write: fill %d\n" , cc)); |
| 2348 | audio_fill_silence(&cb->s.param, einp, cc); |
| 2349 | } |
| 2350 | } |
| 2351 | mutex_exit(sc->sc_intr_lock); |
| 2352 | |
| 2353 | return error; |
| 2354 | } |
| 2355 | |
| 2356 | int |
| 2357 | audio_ioctl(struct audio_softc *sc, u_long cmd, void *addr, int flag, |
| 2358 | struct lwp *l) |
| 2359 | { |
| 2360 | const struct audio_hw_if *hw; |
| 2361 | struct audio_offset *ao; |
| 2362 | u_long stamp; |
| 2363 | int error, offs, fd; |
| 2364 | bool rbus, pbus; |
| 2365 | |
| 2366 | KASSERT(mutex_owned(sc->sc_lock)); |
| 2367 | |
| 2368 | DPRINTF(("audio_ioctl(%lu,'%c',%lu)\n" , |
| 2369 | IOCPARM_LEN(cmd), (char)IOCGROUP(cmd), cmd&0xff)); |
| 2370 | hw = sc->hw_if; |
| 2371 | error = 0; |
| 2372 | switch (cmd) { |
| 2373 | case FIONBIO: |
| 2374 | /* All handled in the upper FS layer. */ |
| 2375 | break; |
| 2376 | |
| 2377 | case FIONREAD: |
| 2378 | *(int *)addr = audio_stream_get_used(sc->sc_rustream); |
| 2379 | break; |
| 2380 | |
| 2381 | case FIOASYNC: |
| 2382 | if (*(int *)addr) { |
| 2383 | if (sc->sc_async_audio != 0) |
| 2384 | error = EBUSY; |
| 2385 | else |
| 2386 | sc->sc_async_audio = curproc->p_pid; |
| 2387 | DPRINTF(("audio_ioctl: FIOASYNC pid %d\n" , |
| 2388 | curproc->p_pid)); |
| 2389 | } else |
| 2390 | sc->sc_async_audio = 0; |
| 2391 | break; |
| 2392 | |
| 2393 | case AUDIO_FLUSH: |
| 2394 | DPRINTF(("AUDIO_FLUSH\n" )); |
| 2395 | rbus = sc->sc_rbus; |
| 2396 | pbus = sc->sc_pbus; |
| 2397 | mutex_enter(sc->sc_intr_lock); |
| 2398 | audio_clear(sc); |
| 2399 | error = audio_initbufs(sc); |
| 2400 | if (error) { |
| 2401 | mutex_exit(sc->sc_intr_lock); |
| 2402 | return error; |
| 2403 | } |
| 2404 | if ((sc->sc_mode & AUMODE_PLAY) && !sc->sc_pbus && pbus) |
| 2405 | error = audiostartp(sc); |
| 2406 | if (!error && |
| 2407 | (sc->sc_mode & AUMODE_RECORD) && !sc->sc_rbus && rbus) |
| 2408 | error = audiostartr(sc); |
| 2409 | mutex_exit(sc->sc_intr_lock); |
| 2410 | break; |
| 2411 | |
| 2412 | /* |
| 2413 | * Number of read (write) samples dropped. We don't know where or |
| 2414 | * when they were dropped. |
| 2415 | */ |
| 2416 | case AUDIO_RERROR: |
| 2417 | *(int *)addr = sc->sc_rr.drops; |
| 2418 | break; |
| 2419 | |
| 2420 | case AUDIO_PERROR: |
| 2421 | *(int *)addr = sc->sc_pr.drops; |
| 2422 | break; |
| 2423 | |
| 2424 | /* |
| 2425 | * Offsets into buffer. |
| 2426 | */ |
| 2427 | case AUDIO_GETIOFFS: |
| 2428 | ao = (struct audio_offset *)addr; |
| 2429 | mutex_enter(sc->sc_intr_lock); |
| 2430 | /* figure out where next DMA will start */ |
| 2431 | stamp = sc->sc_rustream == &sc->sc_rr.s |
| 2432 | ? sc->sc_rr.stamp : sc->sc_rr.fstamp; |
| 2433 | offs = sc->sc_rustream->inp - sc->sc_rustream->start; |
| 2434 | mutex_exit(sc->sc_intr_lock); |
| 2435 | ao->samples = stamp; |
| 2436 | ao->deltablks = |
| 2437 | (stamp / sc->sc_rr.blksize) - |
| 2438 | (sc->sc_rr.stamp_last / sc->sc_rr.blksize); |
| 2439 | sc->sc_rr.stamp_last = stamp; |
| 2440 | ao->offset = offs; |
| 2441 | break; |
| 2442 | |
| 2443 | case AUDIO_GETOOFFS: |
| 2444 | ao = (struct audio_offset *)addr; |
| 2445 | mutex_enter(sc->sc_intr_lock); |
| 2446 | /* figure out where next DMA will start */ |
| 2447 | stamp = sc->sc_pustream == &sc->sc_pr.s |
| 2448 | ? sc->sc_pr.stamp : sc->sc_pr.fstamp; |
| 2449 | offs = sc->sc_pustream->outp - sc->sc_pustream->start |
| 2450 | + sc->sc_pr.blksize; |
| 2451 | mutex_exit(sc->sc_intr_lock); |
| 2452 | ao->samples = stamp; |
| 2453 | ao->deltablks = |
| 2454 | (stamp / sc->sc_pr.blksize) - |
| 2455 | (sc->sc_pr.stamp_last / sc->sc_pr.blksize); |
| 2456 | sc->sc_pr.stamp_last = stamp; |
| 2457 | if (sc->sc_pustream->start + offs >= sc->sc_pustream->end) |
| 2458 | offs = 0; |
| 2459 | ao->offset = offs; |
| 2460 | break; |
| 2461 | |
| 2462 | /* |
| 2463 | * How many bytes will elapse until mike hears the first |
| 2464 | * sample of what we write next? |
| 2465 | */ |
| 2466 | case AUDIO_WSEEK: |
| 2467 | *(u_long *)addr = audio_stream_get_used(sc->sc_pustream); |
| 2468 | break; |
| 2469 | |
| 2470 | case AUDIO_SETINFO: |
| 2471 | DPRINTF(("AUDIO_SETINFO mode=0x%x\n" , sc->sc_mode)); |
| 2472 | error = audiosetinfo(sc, (struct audio_info *)addr); |
| 2473 | break; |
| 2474 | |
| 2475 | case AUDIO_GETINFO: |
| 2476 | DPRINTF(("AUDIO_GETINFO\n" )); |
| 2477 | error = audiogetinfo(sc, (struct audio_info *)addr, 0); |
| 2478 | break; |
| 2479 | |
| 2480 | case AUDIO_GETBUFINFO: |
| 2481 | DPRINTF(("AUDIO_GETBUFINFO\n" )); |
| 2482 | error = audiogetinfo(sc, (struct audio_info *)addr, 1); |
| 2483 | break; |
| 2484 | |
| 2485 | case AUDIO_DRAIN: |
| 2486 | DPRINTF(("AUDIO_DRAIN\n" )); |
| 2487 | mutex_enter(sc->sc_intr_lock); |
| 2488 | error = audio_drain(sc); |
| 2489 | if (!error && hw->drain) |
| 2490 | error = hw->drain(sc->hw_hdl); |
| 2491 | mutex_exit(sc->sc_intr_lock); |
| 2492 | break; |
| 2493 | |
| 2494 | case AUDIO_GETDEV: |
| 2495 | DPRINTF(("AUDIO_GETDEV\n" )); |
| 2496 | error = hw->getdev(sc->hw_hdl, (audio_device_t *)addr); |
| 2497 | break; |
| 2498 | |
| 2499 | case AUDIO_GETENC: |
| 2500 | DPRINTF(("AUDIO_GETENC\n" )); |
| 2501 | error = hw->query_encoding(sc->hw_hdl, |
| 2502 | (struct audio_encoding *)addr); |
| 2503 | break; |
| 2504 | |
| 2505 | case AUDIO_GETFD: |
| 2506 | DPRINTF(("AUDIO_GETFD\n" )); |
| 2507 | *(int *)addr = sc->sc_full_duplex; |
| 2508 | break; |
| 2509 | |
| 2510 | case AUDIO_SETFD: |
| 2511 | DPRINTF(("AUDIO_SETFD\n" )); |
| 2512 | fd = *(int *)addr; |
| 2513 | if (audio_get_props(sc) & AUDIO_PROP_FULLDUPLEX) { |
| 2514 | if (hw->setfd) |
| 2515 | error = hw->setfd(sc->hw_hdl, fd); |
| 2516 | else |
| 2517 | error = 0; |
| 2518 | if (!error) |
| 2519 | sc->sc_full_duplex = fd; |
| 2520 | } else { |
| 2521 | if (fd) |
| 2522 | error = ENOTTY; |
| 2523 | else |
| 2524 | error = 0; |
| 2525 | } |
| 2526 | break; |
| 2527 | |
| 2528 | case AUDIO_GETPROPS: |
| 2529 | DPRINTF(("AUDIO_GETPROPS\n" )); |
| 2530 | *(int *)addr = audio_get_props(sc); |
| 2531 | break; |
| 2532 | |
| 2533 | default: |
| 2534 | if (hw->dev_ioctl) { |
| 2535 | error = hw->dev_ioctl(sc->hw_hdl, cmd, addr, flag, l); |
| 2536 | } else { |
| 2537 | DPRINTF(("audio_ioctl: unknown ioctl\n" )); |
| 2538 | error = EINVAL; |
| 2539 | } |
| 2540 | break; |
| 2541 | } |
| 2542 | DPRINTF(("audio_ioctl(%lu,'%c',%lu) result %d\n" , |
| 2543 | IOCPARM_LEN(cmd), (char)IOCGROUP(cmd), cmd&0xff, error)); |
| 2544 | return error; |
| 2545 | } |
| 2546 | |
| 2547 | int |
| 2548 | audio_poll(struct audio_softc *sc, int events, struct lwp *l) |
| 2549 | { |
| 2550 | int revents; |
| 2551 | int used; |
| 2552 | |
| 2553 | KASSERT(mutex_owned(sc->sc_lock)); |
| 2554 | |
| 2555 | DPRINTF(("audio_poll: events=0x%x mode=%d\n" , events, sc->sc_mode)); |
| 2556 | |
| 2557 | revents = 0; |
| 2558 | mutex_enter(sc->sc_intr_lock); |
| 2559 | if (events & (POLLIN | POLLRDNORM)) { |
| 2560 | used = audio_stream_get_used(sc->sc_rustream); |
| 2561 | /* |
| 2562 | * If half duplex and playing, audio_read() will generate |
| 2563 | * silence at the play rate; poll for silence being |
| 2564 | * available. Otherwise, poll for recorded sound. |
| 2565 | */ |
| 2566 | if ((!sc->sc_full_duplex && (sc->sc_mode & AUMODE_PLAY)) ? |
| 2567 | sc->sc_pr.stamp > sc->sc_wstamp : |
| 2568 | used > sc->sc_rr.usedlow) |
| 2569 | revents |= events & (POLLIN | POLLRDNORM); |
| 2570 | } |
| 2571 | |
| 2572 | if (events & (POLLOUT | POLLWRNORM)) { |
| 2573 | used = audio_stream_get_used(sc->sc_pustream); |
| 2574 | /* |
| 2575 | * If half duplex and recording, audio_write() will throw |
| 2576 | * away play data, which means we are always ready to write. |
| 2577 | * Otherwise, poll for play buffer being below its low water |
| 2578 | * mark. |
| 2579 | */ |
| 2580 | if ((!sc->sc_full_duplex && (sc->sc_mode & AUMODE_RECORD)) || |
| 2581 | (!(sc->sc_mode & AUMODE_PLAY_ALL) && sc->sc_playdrop > 0) || |
| 2582 | (used <= sc->sc_pr.usedlow)) |
| 2583 | revents |= events & (POLLOUT | POLLWRNORM); |
| 2584 | } |
| 2585 | mutex_exit(sc->sc_intr_lock); |
| 2586 | |
| 2587 | if (revents == 0) { |
| 2588 | if (events & (POLLIN | POLLRDNORM)) |
| 2589 | selrecord(l, &sc->sc_rsel); |
| 2590 | |
| 2591 | if (events & (POLLOUT | POLLWRNORM)) |
| 2592 | selrecord(l, &sc->sc_wsel); |
| 2593 | } |
| 2594 | |
| 2595 | return revents; |
| 2596 | } |
| 2597 | |
| 2598 | static void |
| 2599 | filt_audiordetach(struct knote *kn) |
| 2600 | { |
| 2601 | struct audio_softc *sc; |
| 2602 | |
| 2603 | sc = kn->kn_hook; |
| 2604 | mutex_enter(sc->sc_intr_lock); |
| 2605 | SLIST_REMOVE(&sc->sc_rsel.sel_klist, kn, knote, kn_selnext); |
| 2606 | mutex_exit(sc->sc_intr_lock); |
| 2607 | } |
| 2608 | |
| 2609 | static int |
| 2610 | filt_audioread(struct knote *kn, long hint) |
| 2611 | { |
| 2612 | struct audio_softc *sc; |
| 2613 | |
| 2614 | sc = kn->kn_hook; |
| 2615 | mutex_enter(sc->sc_intr_lock); |
| 2616 | if (!sc->sc_full_duplex && (sc->sc_mode & AUMODE_PLAY)) |
| 2617 | kn->kn_data = sc->sc_pr.stamp - sc->sc_wstamp; |
| 2618 | else |
| 2619 | kn->kn_data = audio_stream_get_used(sc->sc_rustream) |
| 2620 | - sc->sc_rr.usedlow; |
| 2621 | mutex_exit(sc->sc_intr_lock); |
| 2622 | |
| 2623 | return kn->kn_data > 0; |
| 2624 | } |
| 2625 | |
| 2626 | static const struct filterops audioread_filtops = |
| 2627 | { 1, NULL, filt_audiordetach, filt_audioread }; |
| 2628 | |
| 2629 | static void |
| 2630 | filt_audiowdetach(struct knote *kn) |
| 2631 | { |
| 2632 | struct audio_softc *sc; |
| 2633 | |
| 2634 | sc = kn->kn_hook; |
| 2635 | mutex_enter(sc->sc_intr_lock); |
| 2636 | SLIST_REMOVE(&sc->sc_wsel.sel_klist, kn, knote, kn_selnext); |
| 2637 | mutex_exit(sc->sc_intr_lock); |
| 2638 | } |
| 2639 | |
| 2640 | static int |
| 2641 | filt_audiowrite(struct knote *kn, long hint) |
| 2642 | { |
| 2643 | struct audio_softc *sc; |
| 2644 | audio_stream_t *stream; |
| 2645 | |
| 2646 | sc = kn->kn_hook; |
| 2647 | mutex_enter(sc->sc_intr_lock); |
| 2648 | stream = sc->sc_pustream; |
| 2649 | kn->kn_data = (stream->end - stream->start) |
| 2650 | - audio_stream_get_used(stream); |
| 2651 | mutex_exit(sc->sc_intr_lock); |
| 2652 | |
| 2653 | return kn->kn_data > 0; |
| 2654 | } |
| 2655 | |
| 2656 | static const struct filterops audiowrite_filtops = |
| 2657 | { 1, NULL, filt_audiowdetach, filt_audiowrite }; |
| 2658 | |
| 2659 | int |
| 2660 | audio_kqfilter(struct audio_softc *sc, struct knote *kn) |
| 2661 | { |
| 2662 | struct klist *klist; |
| 2663 | |
| 2664 | switch (kn->kn_filter) { |
| 2665 | case EVFILT_READ: |
| 2666 | klist = &sc->sc_rsel.sel_klist; |
| 2667 | kn->kn_fop = &audioread_filtops; |
| 2668 | break; |
| 2669 | |
| 2670 | case EVFILT_WRITE: |
| 2671 | klist = &sc->sc_wsel.sel_klist; |
| 2672 | kn->kn_fop = &audiowrite_filtops; |
| 2673 | break; |
| 2674 | |
| 2675 | default: |
| 2676 | return EINVAL; |
| 2677 | } |
| 2678 | |
| 2679 | kn->kn_hook = sc; |
| 2680 | |
| 2681 | mutex_enter(sc->sc_intr_lock); |
| 2682 | SLIST_INSERT_HEAD(klist, kn, kn_selnext); |
| 2683 | mutex_exit(sc->sc_intr_lock); |
| 2684 | |
| 2685 | return 0; |
| 2686 | } |
| 2687 | |
| 2688 | paddr_t |
| 2689 | audio_mmap(struct audio_softc *sc, off_t off, int prot) |
| 2690 | { |
| 2691 | const struct audio_hw_if *hw; |
| 2692 | struct audio_ringbuffer *cb; |
| 2693 | paddr_t rv; |
| 2694 | |
| 2695 | KASSERT(mutex_owned(sc->sc_lock)); |
| 2696 | KASSERT(sc->sc_dvlock > 0); |
| 2697 | |
| 2698 | DPRINTF(("audio_mmap: off=%lld, prot=%d\n" , (long long)off, prot)); |
| 2699 | hw = sc->hw_if; |
| 2700 | if (!(audio_get_props(sc) & AUDIO_PROP_MMAP) || !hw->mappage) |
| 2701 | return -1; |
| 2702 | #if 0 |
| 2703 | /* XXX |
| 2704 | * The idea here was to use the protection to determine if |
| 2705 | * we are mapping the read or write buffer, but it fails. |
| 2706 | * The VM system is broken in (at least) two ways. |
| 2707 | * 1) If you map memory VM_PROT_WRITE you SIGSEGV |
| 2708 | * when writing to it, so VM_PROT_READ|VM_PROT_WRITE |
| 2709 | * has to be used for mmapping the play buffer. |
| 2710 | * 2) Even if calling mmap() with VM_PROT_READ|VM_PROT_WRITE |
| 2711 | * audio_mmap will get called at some point with VM_PROT_READ |
| 2712 | * only. |
| 2713 | * So, alas, we always map the play buffer for now. |
| 2714 | */ |
| 2715 | if (prot == (VM_PROT_READ|VM_PROT_WRITE) || |
| 2716 | prot == VM_PROT_WRITE) |
| 2717 | cb = &sc->sc_pr; |
| 2718 | else if (prot == VM_PROT_READ) |
| 2719 | cb = &sc->sc_rr; |
| 2720 | else |
| 2721 | return -1; |
| 2722 | #else |
| 2723 | cb = &sc->sc_pr; |
| 2724 | #endif |
| 2725 | |
| 2726 | if ((u_int)off >= cb->s.bufsize) |
| 2727 | return -1; |
| 2728 | if (!cb->mmapped) { |
| 2729 | cb->mmapped = true; |
| 2730 | if (cb == &sc->sc_pr) { |
| 2731 | audio_fill_silence(&cb->s.param, cb->s.start, |
| 2732 | cb->s.bufsize); |
| 2733 | mutex_enter(sc->sc_intr_lock); |
| 2734 | sc->sc_pustream = &cb->s; |
| 2735 | if (!sc->sc_pbus && !sc->sc_pr.pause) |
| 2736 | (void)audiostartp(sc); |
| 2737 | mutex_exit(sc->sc_intr_lock); |
| 2738 | } else { |
| 2739 | mutex_enter(sc->sc_intr_lock); |
| 2740 | sc->sc_rustream = &cb->s; |
| 2741 | if (!sc->sc_rbus && !sc->sc_rr.pause) |
| 2742 | (void)audiostartr(sc); |
| 2743 | mutex_exit(sc->sc_intr_lock); |
| 2744 | } |
| 2745 | } |
| 2746 | |
| 2747 | mutex_exit(sc->sc_lock); |
| 2748 | rv = hw->mappage(sc->hw_hdl, cb->s.start, off, prot); |
| 2749 | mutex_enter(sc->sc_lock); |
| 2750 | |
| 2751 | return rv; |
| 2752 | } |
| 2753 | |
| 2754 | int |
| 2755 | audiostartr(struct audio_softc *sc) |
| 2756 | { |
| 2757 | int error; |
| 2758 | |
| 2759 | KASSERT(mutex_owned(sc->sc_lock)); |
| 2760 | KASSERT(mutex_owned(sc->sc_intr_lock)); |
| 2761 | |
| 2762 | DPRINTF(("audiostartr: start=%p used=%d(hi=%d) mmapped=%d\n" , |
| 2763 | sc->sc_rr.s.start, audio_stream_get_used(&sc->sc_rr.s), |
| 2764 | sc->sc_rr.usedhigh, sc->sc_rr.mmapped)); |
| 2765 | |
| 2766 | if (!audio_can_capture(sc)) |
| 2767 | return EINVAL; |
| 2768 | |
| 2769 | if (sc->hw_if->trigger_input) |
| 2770 | error = sc->hw_if->trigger_input(sc->hw_hdl, sc->sc_rr.s.start, |
| 2771 | sc->sc_rr.s.end, sc->sc_rr.blksize, |
| 2772 | audio_rint, (void *)sc, &sc->sc_rr.s.param); |
| 2773 | else |
| 2774 | error = sc->hw_if->start_input(sc->hw_hdl, sc->sc_rr.s.start, |
| 2775 | sc->sc_rr.blksize, audio_rint, (void *)sc); |
| 2776 | if (error) { |
| 2777 | DPRINTF(("audiostartr failed: %d\n" , error)); |
| 2778 | return error; |
| 2779 | } |
| 2780 | sc->sc_rbus = true; |
| 2781 | return 0; |
| 2782 | } |
| 2783 | |
| 2784 | int |
| 2785 | audiostartp(struct audio_softc *sc) |
| 2786 | { |
| 2787 | int error; |
| 2788 | int used; |
| 2789 | |
| 2790 | KASSERT(mutex_owned(sc->sc_lock)); |
| 2791 | KASSERT(mutex_owned(sc->sc_intr_lock)); |
| 2792 | |
| 2793 | used = audio_stream_get_used(&sc->sc_pr.s); |
| 2794 | DPRINTF(("audiostartp: start=%p used=%d(hi=%d blk=%d) mmapped=%d\n" , |
| 2795 | sc->sc_pr.s.start, used, sc->sc_pr.usedhigh, |
| 2796 | sc->sc_pr.blksize, sc->sc_pr.mmapped)); |
| 2797 | |
| 2798 | if (!audio_can_playback(sc)) |
| 2799 | return EINVAL; |
| 2800 | |
| 2801 | if (!sc->sc_pr.mmapped && used < sc->sc_pr.blksize) { |
| 2802 | cv_broadcast(&sc->sc_wchan); |
| 2803 | DPRINTF(("%s: wakeup and return\n" , __func__)); |
| 2804 | return 0; |
| 2805 | } |
| 2806 | |
| 2807 | if (sc->hw_if->trigger_output) { |
| 2808 | DPRINTF(("%s: call trigger_output\n" , __func__)); |
| 2809 | error = sc->hw_if->trigger_output(sc->hw_hdl, sc->sc_pr.s.start, |
| 2810 | sc->sc_pr.s.end, sc->sc_pr.blksize, |
| 2811 | audio_pint, (void *)sc, &sc->sc_pr.s.param); |
| 2812 | } else { |
| 2813 | DPRINTF(("%s: call start_output\n" , __func__)); |
| 2814 | error = sc->hw_if->start_output(sc->hw_hdl, |
| 2815 | __UNCONST(sc->sc_pr.s.outp), sc->sc_pr.blksize, |
| 2816 | audio_pint, (void *)sc); |
| 2817 | } |
| 2818 | if (error) { |
| 2819 | DPRINTF(("audiostartp failed: %d\n" , error)); |
| 2820 | return error; |
| 2821 | } |
| 2822 | sc->sc_pbus = true; |
| 2823 | return 0; |
| 2824 | } |
| 2825 | |
| 2826 | /* |
| 2827 | * When the play interrupt routine finds that the write isn't keeping |
| 2828 | * the buffer filled it will insert silence in the buffer to make up |
| 2829 | * for this. The part of the buffer that is filled with silence |
| 2830 | * is kept track of in a very approximate way: it starts at sc_sil_start |
| 2831 | * and extends sc_sil_count bytes. If there is already silence in |
| 2832 | * the requested area nothing is done; so when the whole buffer is |
| 2833 | * silent nothing happens. When the writer starts again sc_sil_count |
| 2834 | * is set to 0. |
| 2835 | * |
| 2836 | * XXX |
| 2837 | * Putting silence into the output buffer should not really be done |
| 2838 | * from the device interrupt handler. Consider deferring to the soft |
| 2839 | * interrupt. |
| 2840 | */ |
| 2841 | static inline void |
| 2842 | audio_pint_silence(struct audio_softc *sc, struct audio_ringbuffer *cb, |
| 2843 | uint8_t *inp, int cc) |
| 2844 | { |
| 2845 | uint8_t *s, *e, *p, *q; |
| 2846 | |
| 2847 | KASSERT(mutex_owned(sc->sc_intr_lock)); |
| 2848 | |
| 2849 | if (sc->sc_sil_count > 0) { |
| 2850 | s = sc->sc_sil_start; /* start of silence */ |
| 2851 | e = s + sc->sc_sil_count; /* end of sil., may be beyond end */ |
| 2852 | p = inp; /* adjusted pointer to area to fill */ |
| 2853 | if (p < s) |
| 2854 | p += cb->s.end - cb->s.start; |
| 2855 | q = p + cc; |
| 2856 | /* Check if there is already silence. */ |
| 2857 | if (!(s <= p && p < e && |
| 2858 | s <= q && q <= e)) { |
| 2859 | if (s <= p) |
| 2860 | sc->sc_sil_count = max(sc->sc_sil_count, q-s); |
| 2861 | DPRINTFN(5,("audio_pint_silence: fill cc=%d inp=%p, " |
| 2862 | "count=%d size=%d\n" , |
| 2863 | cc, inp, sc->sc_sil_count, |
| 2864 | (int)(cb->s.end - cb->s.start))); |
| 2865 | audio_fill_silence(&cb->s.param, inp, cc); |
| 2866 | } else { |
| 2867 | DPRINTFN(5,("audio_pint_silence: already silent " |
| 2868 | "cc=%d inp=%p\n" , cc, inp)); |
| 2869 | |
| 2870 | } |
| 2871 | } else { |
| 2872 | sc->sc_sil_start = inp; |
| 2873 | sc->sc_sil_count = cc; |
| 2874 | DPRINTFN(5, ("audio_pint_silence: start fill %p %d\n" , |
| 2875 | inp, cc)); |
| 2876 | audio_fill_silence(&cb->s.param, inp, cc); |
| 2877 | } |
| 2878 | } |
| 2879 | |
| 2880 | static void |
| 2881 | audio_softintr_rd(void *cookie) |
| 2882 | { |
| 2883 | struct audio_softc *sc = cookie; |
| 2884 | proc_t *p; |
| 2885 | pid_t pid; |
| 2886 | |
| 2887 | mutex_enter(sc->sc_lock); |
| 2888 | cv_broadcast(&sc->sc_rchan); |
| 2889 | selnotify(&sc->sc_rsel, 0, NOTE_SUBMIT); |
| 2890 | if ((pid = sc->sc_async_audio) != 0) { |
| 2891 | DPRINTFN(3, ("audio_softintr_rd: sending SIGIO %d\n" , pid)); |
| 2892 | mutex_enter(proc_lock); |
| 2893 | if ((p = proc_find(pid)) != NULL) |
| 2894 | psignal(p, SIGIO); |
| 2895 | mutex_exit(proc_lock); |
| 2896 | } |
| 2897 | mutex_exit(sc->sc_lock); |
| 2898 | } |
| 2899 | |
| 2900 | static void |
| 2901 | audio_softintr_wr(void *cookie) |
| 2902 | { |
| 2903 | struct audio_softc *sc = cookie; |
| 2904 | proc_t *p; |
| 2905 | pid_t pid; |
| 2906 | |
| 2907 | mutex_enter(sc->sc_lock); |
| 2908 | cv_broadcast(&sc->sc_wchan); |
| 2909 | selnotify(&sc->sc_wsel, 0, NOTE_SUBMIT); |
| 2910 | if ((pid = sc->sc_async_audio) != 0) { |
| 2911 | DPRINTFN(3, ("audio_softintr_wr: sending SIGIO %d\n" , pid)); |
| 2912 | mutex_enter(proc_lock); |
| 2913 | if ((p = proc_find(pid)) != NULL) |
| 2914 | psignal(p, SIGIO); |
| 2915 | mutex_exit(proc_lock); |
| 2916 | } |
| 2917 | mutex_exit(sc->sc_lock); |
| 2918 | } |
| 2919 | |
| 2920 | /* |
| 2921 | * Called from HW driver module on completion of DMA output. |
| 2922 | * Start output of new block, wrap in ring buffer if needed. |
| 2923 | * If no more buffers to play, output zero instead. |
| 2924 | * Do a wakeup if necessary. |
| 2925 | */ |
| 2926 | void |
| 2927 | audio_pint(void *v) |
| 2928 | { |
| 2929 | stream_fetcher_t null_fetcher; |
| 2930 | struct audio_softc *sc; |
| 2931 | const struct audio_hw_if *hw; |
| 2932 | struct audio_ringbuffer *cb; |
| 2933 | stream_fetcher_t *fetcher; |
| 2934 | uint8_t *inp; |
| 2935 | int cc, used; |
| 2936 | int blksize; |
| 2937 | int error; |
| 2938 | |
| 2939 | sc = v; |
| 2940 | |
| 2941 | KASSERT(mutex_owned(sc->sc_intr_lock)); |
| 2942 | |
| 2943 | if (!sc->sc_open) |
| 2944 | return; /* ignore interrupt if not open */ |
| 2945 | |
| 2946 | hw = sc->hw_if; |
| 2947 | cb = &sc->sc_pr; |
| 2948 | blksize = cb->blksize; |
| 2949 | cb->s.outp = audio_stream_add_outp(&cb->s, cb->s.outp, blksize); |
| 2950 | cb->stamp += blksize; |
| 2951 | if (cb->mmapped) { |
| 2952 | DPRINTFN(5, ("audio_pint: mmapped outp=%p cc=%d inp=%p\n" , |
| 2953 | cb->s.outp, blksize, cb->s.inp)); |
| 2954 | if (hw->trigger_output == NULL) |
| 2955 | (void)hw->start_output(sc->hw_hdl, __UNCONST(cb->s.outp), |
| 2956 | blksize, audio_pint, (void *)sc); |
| 2957 | return; |
| 2958 | } |
| 2959 | |
| 2960 | #ifdef AUDIO_INTR_TIME |
| 2961 | { |
| 2962 | struct timeval tv; |
| 2963 | u_long t; |
| 2964 | microtime(&tv); |
| 2965 | t = tv.tv_usec + 1000000 * tv.tv_sec; |
| 2966 | if (sc->sc_pnintr) { |
| 2967 | long lastdelta, totdelta; |
| 2968 | lastdelta = t - sc->sc_plastintr - sc->sc_pblktime; |
| 2969 | if (lastdelta > sc->sc_pblktime / 3) { |
| 2970 | printf("audio: play interrupt(%d) off " |
| 2971 | "relative by %ld us (%lu)\n" , |
| 2972 | sc->sc_pnintr, lastdelta, |
| 2973 | sc->sc_pblktime); |
| 2974 | } |
| 2975 | totdelta = t - sc->sc_pfirstintr - |
| 2976 | sc->sc_pblktime * sc->sc_pnintr; |
| 2977 | if (totdelta > sc->sc_pblktime) { |
| 2978 | printf("audio: play interrupt(%d) off " |
| 2979 | "absolute by %ld us (%lu) (LOST)\n" , |
| 2980 | sc->sc_pnintr, totdelta, |
| 2981 | sc->sc_pblktime); |
| 2982 | sc->sc_pnintr++; /* avoid repeated messages */ |
| 2983 | } |
| 2984 | } else |
| 2985 | sc->sc_pfirstintr = t; |
| 2986 | sc->sc_plastintr = t; |
| 2987 | sc->sc_pnintr++; |
| 2988 | } |
| 2989 | #endif |
| 2990 | |
| 2991 | used = audio_stream_get_used(&cb->s); |
| 2992 | /* |
| 2993 | * "used <= cb->usedlow" should be "used < blksize" ideally. |
| 2994 | * Some HW drivers such as uaudio(4) does not call audio_pint() |
| 2995 | * at accurate timing. If used < blksize, uaudio(4) already |
| 2996 | * request transfer of garbage data. |
| 2997 | */ |
| 2998 | if (used <= cb->usedlow && !cb->copying && sc->sc_npfilters > 0) { |
| 2999 | /* we might have data in filter pipeline */ |
| 3000 | null_fetcher.fetch_to = null_fetcher_fetch_to; |
| 3001 | fetcher = &sc->sc_pfilters[sc->sc_npfilters - 1]->base; |
| 3002 | sc->sc_pfilters[0]->set_fetcher(sc->sc_pfilters[0], |
| 3003 | &null_fetcher); |
| 3004 | used = audio_stream_get_used(sc->sc_pustream); |
| 3005 | cc = cb->s.end - cb->s.start; |
| 3006 | if (blksize * 2 < cc) |
| 3007 | cc = blksize * 2; |
| 3008 | fetcher->fetch_to(sc, fetcher, &cb->s, cc); |
| 3009 | cb->fstamp += used - audio_stream_get_used(sc->sc_pustream); |
| 3010 | used = audio_stream_get_used(&cb->s); |
| 3011 | } |
| 3012 | if (used < blksize) { |
| 3013 | /* we don't have a full block to use */ |
| 3014 | if (cb->copying) { |
| 3015 | /* writer is in progress, don't disturb */ |
| 3016 | cb->needfill = true; |
| 3017 | DPRINTFN(1, ("audio_pint: copying in progress\n" )); |
| 3018 | } else { |
| 3019 | inp = cb->s.inp; |
| 3020 | cc = blksize - (inp - cb->s.start) % blksize; |
| 3021 | if (cb->pause) |
| 3022 | cb->pdrops += cc; |
| 3023 | else { |
| 3024 | cb->drops += cc; |
| 3025 | sc->sc_playdrop += cc; |
| 3026 | } |
| 3027 | audio_pint_silence(sc, cb, inp, cc); |
| 3028 | cb->s.inp = audio_stream_add_inp(&cb->s, inp, cc); |
| 3029 | |
| 3030 | /* Clear next block so we keep ahead of the DMA. */ |
| 3031 | used = audio_stream_get_used(&cb->s); |
| 3032 | if (used + blksize < cb->s.end - cb->s.start) |
| 3033 | audio_pint_silence(sc, cb, cb->s.inp, blksize); |
| 3034 | } |
| 3035 | } |
| 3036 | |
| 3037 | DPRINTFN(5, ("audio_pint: outp=%p cc=%d\n" , cb->s.outp, blksize)); |
| 3038 | if (hw->trigger_output == NULL) { |
| 3039 | error = hw->start_output(sc->hw_hdl, __UNCONST(cb->s.outp), |
| 3040 | blksize, audio_pint, (void *)sc); |
| 3041 | if (error) { |
| 3042 | /* XXX does this really help? */ |
| 3043 | DPRINTF(("audio_pint restart failed: %d\n" , error)); |
| 3044 | audio_clear(sc); |
| 3045 | } |
| 3046 | } |
| 3047 | |
| 3048 | DPRINTFN(2, ("audio_pint: mode=%d pause=%d used=%d lowat=%d\n" , |
| 3049 | sc->sc_mode, cb->pause, |
| 3050 | audio_stream_get_used(sc->sc_pustream), cb->usedlow)); |
| 3051 | if ((sc->sc_mode & AUMODE_PLAY) && !cb->pause) { |
| 3052 | if (audio_stream_get_used(sc->sc_pustream) <= cb->usedlow) |
| 3053 | softint_schedule(sc->sc_sih_wr); |
| 3054 | } |
| 3055 | |
| 3056 | /* Possible to return one or more "phantom blocks" now. */ |
| 3057 | if (!sc->sc_full_duplex) |
| 3058 | softint_schedule(sc->sc_sih_rd); |
| 3059 | } |
| 3060 | |
| 3061 | /* |
| 3062 | * Called from HW driver module on completion of DMA input. |
| 3063 | * Mark it as input in the ring buffer (fiddle pointers). |
| 3064 | * Do a wakeup if necessary. |
| 3065 | */ |
| 3066 | void |
| 3067 | audio_rint(void *v) |
| 3068 | { |
| 3069 | stream_fetcher_t null_fetcher; |
| 3070 | struct audio_softc *sc; |
| 3071 | const struct audio_hw_if *hw; |
| 3072 | struct audio_ringbuffer *cb; |
| 3073 | stream_fetcher_t *last_fetcher; |
| 3074 | int cc; |
| 3075 | int used; |
| 3076 | int blksize; |
| 3077 | int error; |
| 3078 | |
| 3079 | sc = v; |
| 3080 | cb = &sc->sc_rr; |
| 3081 | |
| 3082 | KASSERT(mutex_owned(sc->sc_intr_lock)); |
| 3083 | |
| 3084 | if (!sc->sc_open) |
| 3085 | return; /* ignore interrupt if not open */ |
| 3086 | |
| 3087 | hw = sc->hw_if; |
| 3088 | blksize = cb->blksize; |
| 3089 | cb->s.inp = audio_stream_add_inp(&cb->s, cb->s.inp, blksize); |
| 3090 | cb->stamp += blksize; |
| 3091 | if (cb->mmapped) { |
| 3092 | DPRINTFN(2, ("audio_rint: mmapped inp=%p cc=%d\n" , |
| 3093 | cb->s.inp, blksize)); |
| 3094 | if (hw->trigger_input == NULL) |
| 3095 | (void)hw->start_input(sc->hw_hdl, cb->s.inp, blksize, |
| 3096 | audio_rint, (void *)sc); |
| 3097 | return; |
| 3098 | } |
| 3099 | |
| 3100 | #ifdef AUDIO_INTR_TIME |
| 3101 | { |
| 3102 | struct timeval tv; |
| 3103 | u_long t; |
| 3104 | microtime(&tv); |
| 3105 | t = tv.tv_usec + 1000000 * tv.tv_sec; |
| 3106 | if (sc->sc_rnintr) { |
| 3107 | long lastdelta, totdelta; |
| 3108 | lastdelta = t - sc->sc_rlastintr - sc->sc_rblktime; |
| 3109 | if (lastdelta > sc->sc_rblktime / 5) { |
| 3110 | printf("audio: record interrupt(%d) off " |
| 3111 | "relative by %ld us (%lu)\n" , |
| 3112 | sc->sc_rnintr, lastdelta, |
| 3113 | sc->sc_rblktime); |
| 3114 | } |
| 3115 | totdelta = t - sc->sc_rfirstintr - |
| 3116 | sc->sc_rblktime * sc->sc_rnintr; |
| 3117 | if (totdelta > sc->sc_rblktime / 2) { |
| 3118 | sc->sc_rnintr++; |
| 3119 | printf("audio: record interrupt(%d) off " |
| 3120 | "absolute by %ld us (%lu)\n" , |
| 3121 | sc->sc_rnintr, totdelta, |
| 3122 | sc->sc_rblktime); |
| 3123 | sc->sc_rnintr++; /* avoid repeated messages */ |
| 3124 | } |
| 3125 | } else |
| 3126 | sc->sc_rfirstintr = t; |
| 3127 | sc->sc_rlastintr = t; |
| 3128 | sc->sc_rnintr++; |
| 3129 | } |
| 3130 | #endif |
| 3131 | |
| 3132 | if (!cb->pause && sc->sc_nrfilters > 0) { |
| 3133 | null_fetcher.fetch_to = null_fetcher_fetch_to; |
| 3134 | last_fetcher = &sc->sc_rfilters[sc->sc_nrfilters - 1]->base; |
| 3135 | sc->sc_rfilters[0]->set_fetcher(sc->sc_rfilters[0], |
| 3136 | &null_fetcher); |
| 3137 | used = audio_stream_get_used(sc->sc_rustream); |
| 3138 | cc = sc->sc_rustream->end - sc->sc_rustream->start; |
| 3139 | error = last_fetcher->fetch_to |
| 3140 | (sc, last_fetcher, sc->sc_rustream, cc); |
| 3141 | cb->fstamp += audio_stream_get_used(sc->sc_rustream) - used; |
| 3142 | /* XXX what should do for error? */ |
| 3143 | } |
| 3144 | used = audio_stream_get_used(&sc->sc_rr.s); |
| 3145 | if (cb->pause) { |
| 3146 | DPRINTFN(1, ("audio_rint: pdrops %lu\n" , cb->pdrops)); |
| 3147 | cb->pdrops += blksize; |
| 3148 | cb->s.outp = audio_stream_add_outp(&cb->s, cb->s.outp, blksize); |
| 3149 | } else if (used + blksize > cb->s.end - cb->s.start && !cb->copying) { |
| 3150 | DPRINTFN(1, ("audio_rint: drops %lu\n" , cb->drops)); |
| 3151 | cb->drops += blksize; |
| 3152 | cb->s.outp = audio_stream_add_outp(&cb->s, cb->s.outp, blksize); |
| 3153 | } |
| 3154 | |
| 3155 | DPRINTFN(2, ("audio_rint: inp=%p cc=%d\n" , cb->s.inp, blksize)); |
| 3156 | if (hw->trigger_input == NULL) { |
| 3157 | error = hw->start_input(sc->hw_hdl, cb->s.inp, blksize, |
| 3158 | audio_rint, (void *)sc); |
| 3159 | if (error) { |
| 3160 | /* XXX does this really help? */ |
| 3161 | DPRINTF(("audio_rint: restart failed: %d\n" , error)); |
| 3162 | audio_clear(sc); |
| 3163 | } |
| 3164 | } |
| 3165 | |
| 3166 | softint_schedule(sc->sc_sih_rd); |
| 3167 | } |
| 3168 | |
| 3169 | int |
| 3170 | audio_check_params(struct audio_params *p) |
| 3171 | { |
| 3172 | |
| 3173 | if (p->encoding == AUDIO_ENCODING_PCM16) { |
| 3174 | if (p->precision == 8) |
| 3175 | p->encoding = AUDIO_ENCODING_ULINEAR; |
| 3176 | else |
| 3177 | p->encoding = AUDIO_ENCODING_SLINEAR; |
| 3178 | } else if (p->encoding == AUDIO_ENCODING_PCM8) { |
| 3179 | if (p->precision == 8) |
| 3180 | p->encoding = AUDIO_ENCODING_ULINEAR; |
| 3181 | else |
| 3182 | return EINVAL; |
| 3183 | } |
| 3184 | |
| 3185 | if (p->encoding == AUDIO_ENCODING_SLINEAR) |
| 3186 | #if BYTE_ORDER == LITTLE_ENDIAN |
| 3187 | p->encoding = AUDIO_ENCODING_SLINEAR_LE; |
| 3188 | #else |
| 3189 | p->encoding = AUDIO_ENCODING_SLINEAR_BE; |
| 3190 | #endif |
| 3191 | if (p->encoding == AUDIO_ENCODING_ULINEAR) |
| 3192 | #if BYTE_ORDER == LITTLE_ENDIAN |
| 3193 | p->encoding = AUDIO_ENCODING_ULINEAR_LE; |
| 3194 | #else |
| 3195 | p->encoding = AUDIO_ENCODING_ULINEAR_BE; |
| 3196 | #endif |
| 3197 | |
| 3198 | switch (p->encoding) { |
| 3199 | case AUDIO_ENCODING_ULAW: |
| 3200 | case AUDIO_ENCODING_ALAW: |
| 3201 | if (p->precision != 8) |
| 3202 | return EINVAL; |
| 3203 | break; |
| 3204 | case AUDIO_ENCODING_ADPCM: |
| 3205 | if (p->precision != 4 && p->precision != 8) |
| 3206 | return EINVAL; |
| 3207 | break; |
| 3208 | case AUDIO_ENCODING_SLINEAR_LE: |
| 3209 | case AUDIO_ENCODING_SLINEAR_BE: |
| 3210 | case AUDIO_ENCODING_ULINEAR_LE: |
| 3211 | case AUDIO_ENCODING_ULINEAR_BE: |
| 3212 | /* XXX is: our zero-fill can handle any multiple of 8 */ |
| 3213 | if (p->precision != 8 && p->precision != 16 && |
| 3214 | p->precision != 24 && p->precision != 32) |
| 3215 | return EINVAL; |
| 3216 | if (p->precision == 8 && p->encoding == AUDIO_ENCODING_SLINEAR_BE) |
| 3217 | p->encoding = AUDIO_ENCODING_SLINEAR_LE; |
| 3218 | if (p->precision == 8 && p->encoding == AUDIO_ENCODING_ULINEAR_BE) |
| 3219 | p->encoding = AUDIO_ENCODING_ULINEAR_LE; |
| 3220 | if (p->validbits > p->precision) |
| 3221 | return EINVAL; |
| 3222 | break; |
| 3223 | case AUDIO_ENCODING_MPEG_L1_STREAM: |
| 3224 | case AUDIO_ENCODING_MPEG_L1_PACKETS: |
| 3225 | case AUDIO_ENCODING_MPEG_L1_SYSTEM: |
| 3226 | case AUDIO_ENCODING_MPEG_L2_STREAM: |
| 3227 | case AUDIO_ENCODING_MPEG_L2_PACKETS: |
| 3228 | case AUDIO_ENCODING_MPEG_L2_SYSTEM: |
| 3229 | case AUDIO_ENCODING_AC3: |
| 3230 | break; |
| 3231 | default: |
| 3232 | return EINVAL; |
| 3233 | } |
| 3234 | |
| 3235 | /* sanity check # of channels*/ |
| 3236 | if (p->channels < 1 || p->channels > AUDIO_MAX_CHANNELS) |
| 3237 | return EINVAL; |
| 3238 | |
| 3239 | return 0; |
| 3240 | } |
| 3241 | |
| 3242 | int |
| 3243 | audio_set_defaults(struct audio_softc *sc, u_int mode) |
| 3244 | { |
| 3245 | struct audio_info ai; |
| 3246 | |
| 3247 | KASSERT(mutex_owned(sc->sc_lock)); |
| 3248 | |
| 3249 | /* default parameters */ |
| 3250 | sc->sc_rparams = audio_default; |
| 3251 | sc->sc_pparams = audio_default; |
| 3252 | sc->sc_blkset = false; |
| 3253 | |
| 3254 | AUDIO_INITINFO(&ai); |
| 3255 | ai.record.sample_rate = sc->sc_rparams.sample_rate; |
| 3256 | ai.record.encoding = sc->sc_rparams.encoding; |
| 3257 | ai.record.channels = sc->sc_rparams.channels; |
| 3258 | ai.record.precision = sc->sc_rparams.precision; |
| 3259 | ai.record.pause = false; |
| 3260 | ai.play.sample_rate = sc->sc_pparams.sample_rate; |
| 3261 | ai.play.encoding = sc->sc_pparams.encoding; |
| 3262 | ai.play.channels = sc->sc_pparams.channels; |
| 3263 | ai.play.precision = sc->sc_pparams.precision; |
| 3264 | ai.play.pause = false; |
| 3265 | ai.mode = mode; |
| 3266 | |
| 3267 | return audiosetinfo(sc, &ai); |
| 3268 | } |
| 3269 | |
| 3270 | int |
| 3271 | au_set_lr_value(struct audio_softc *sc, mixer_ctrl_t *ct, int l, int r) |
| 3272 | { |
| 3273 | |
| 3274 | KASSERT(mutex_owned(sc->sc_lock)); |
| 3275 | |
| 3276 | ct->type = AUDIO_MIXER_VALUE; |
| 3277 | ct->un.value.num_channels = 2; |
| 3278 | ct->un.value.level[AUDIO_MIXER_LEVEL_LEFT] = l; |
| 3279 | ct->un.value.level[AUDIO_MIXER_LEVEL_RIGHT] = r; |
| 3280 | if (sc->hw_if->set_port(sc->hw_hdl, ct) == 0) |
| 3281 | return 0; |
| 3282 | ct->un.value.num_channels = 1; |
| 3283 | ct->un.value.level[AUDIO_MIXER_LEVEL_MONO] = (l+r)/2; |
| 3284 | return sc->hw_if->set_port(sc->hw_hdl, ct); |
| 3285 | } |
| 3286 | |
| 3287 | int |
| 3288 | au_set_gain(struct audio_softc *sc, struct au_mixer_ports *ports, |
| 3289 | int gain, int balance) |
| 3290 | { |
| 3291 | mixer_ctrl_t ct; |
| 3292 | int i, error; |
| 3293 | int l, r; |
| 3294 | u_int mask; |
| 3295 | int nset; |
| 3296 | |
| 3297 | KASSERT(mutex_owned(sc->sc_lock)); |
| 3298 | |
| 3299 | if (balance == AUDIO_MID_BALANCE) { |
| 3300 | l = r = gain; |
| 3301 | } else if (balance < AUDIO_MID_BALANCE) { |
| 3302 | l = gain; |
| 3303 | r = (balance * gain) / AUDIO_MID_BALANCE; |
| 3304 | } else { |
| 3305 | r = gain; |
| 3306 | l = ((AUDIO_RIGHT_BALANCE - balance) * gain) |
| 3307 | / AUDIO_MID_BALANCE; |
| 3308 | } |
| 3309 | DPRINTF(("au_set_gain: gain=%d balance=%d, l=%d r=%d\n" , |
| 3310 | gain, balance, l, r)); |
| 3311 | |
| 3312 | if (ports->index == -1) { |
| 3313 | usemaster: |
| 3314 | if (ports->master == -1) |
| 3315 | return 0; /* just ignore it silently */ |
| 3316 | ct.dev = ports->master; |
| 3317 | error = au_set_lr_value(sc, &ct, l, r); |
| 3318 | } else { |
| 3319 | ct.dev = ports->index; |
| 3320 | if (ports->isenum) { |
| 3321 | ct.type = AUDIO_MIXER_ENUM; |
| 3322 | error = sc->hw_if->get_port(sc->hw_hdl, &ct); |
| 3323 | if (error) |
| 3324 | return error; |
| 3325 | if (ports->isdual) { |
| 3326 | if (ports->cur_port == -1) |
| 3327 | ct.dev = ports->master; |
| 3328 | else |
| 3329 | ct.dev = ports->miport[ports->cur_port]; |
| 3330 | error = au_set_lr_value(sc, &ct, l, r); |
| 3331 | } else { |
| 3332 | for(i = 0; i < ports->nports; i++) |
| 3333 | if (ports->misel[i] == ct.un.ord) { |
| 3334 | ct.dev = ports->miport[i]; |
| 3335 | if (ct.dev == -1 || |
| 3336 | au_set_lr_value(sc, &ct, l, r)) |
| 3337 | goto usemaster; |
| 3338 | else |
| 3339 | break; |
| 3340 | } |
| 3341 | } |
| 3342 | } else { |
| 3343 | ct.type = AUDIO_MIXER_SET; |
| 3344 | error = sc->hw_if->get_port(sc->hw_hdl, &ct); |
| 3345 | if (error) |
| 3346 | return error; |
| 3347 | mask = ct.un.mask; |
| 3348 | nset = 0; |
| 3349 | for(i = 0; i < ports->nports; i++) { |
| 3350 | if (ports->misel[i] & mask) { |
| 3351 | ct.dev = ports->miport[i]; |
| 3352 | if (ct.dev != -1 && |
| 3353 | au_set_lr_value(sc, &ct, l, r) == 0) |
| 3354 | nset++; |
| 3355 | } |
| 3356 | } |
| 3357 | if (nset == 0) |
| 3358 | goto usemaster; |
| 3359 | } |
| 3360 | } |
| 3361 | if (!error) |
| 3362 | mixer_signal(sc); |
| 3363 | return error; |
| 3364 | } |
| 3365 | |
| 3366 | int |
| 3367 | au_get_lr_value(struct audio_softc *sc, mixer_ctrl_t *ct, int *l, int *r) |
| 3368 | { |
| 3369 | int error; |
| 3370 | |
| 3371 | KASSERT(mutex_owned(sc->sc_lock)); |
| 3372 | |
| 3373 | ct->un.value.num_channels = 2; |
| 3374 | if (sc->hw_if->get_port(sc->hw_hdl, ct) == 0) { |
| 3375 | *l = ct->un.value.level[AUDIO_MIXER_LEVEL_LEFT]; |
| 3376 | *r = ct->un.value.level[AUDIO_MIXER_LEVEL_RIGHT]; |
| 3377 | } else { |
| 3378 | ct->un.value.num_channels = 1; |
| 3379 | error = sc->hw_if->get_port(sc->hw_hdl, ct); |
| 3380 | if (error) |
| 3381 | return error; |
| 3382 | *r = *l = ct->un.value.level[AUDIO_MIXER_LEVEL_MONO]; |
| 3383 | } |
| 3384 | return 0; |
| 3385 | } |
| 3386 | |
| 3387 | void |
| 3388 | au_get_gain(struct audio_softc *sc, struct au_mixer_ports *ports, |
| 3389 | u_int *pgain, u_char *pbalance) |
| 3390 | { |
| 3391 | mixer_ctrl_t ct; |
| 3392 | int i, l, r, n; |
| 3393 | int lgain, rgain; |
| 3394 | |
| 3395 | KASSERT(mutex_owned(sc->sc_lock)); |
| 3396 | |
| 3397 | lgain = AUDIO_MAX_GAIN / 2; |
| 3398 | rgain = AUDIO_MAX_GAIN / 2; |
| 3399 | if (ports->index == -1) { |
| 3400 | usemaster: |
| 3401 | if (ports->master == -1) |
| 3402 | goto bad; |
| 3403 | ct.dev = ports->master; |
| 3404 | ct.type = AUDIO_MIXER_VALUE; |
| 3405 | if (au_get_lr_value(sc, &ct, &lgain, &rgain)) |
| 3406 | goto bad; |
| 3407 | } else { |
| 3408 | ct.dev = ports->index; |
| 3409 | if (ports->isenum) { |
| 3410 | ct.type = AUDIO_MIXER_ENUM; |
| 3411 | if (sc->hw_if->get_port(sc->hw_hdl, &ct)) |
| 3412 | goto bad; |
| 3413 | ct.type = AUDIO_MIXER_VALUE; |
| 3414 | if (ports->isdual) { |
| 3415 | if (ports->cur_port == -1) |
| 3416 | ct.dev = ports->master; |
| 3417 | else |
| 3418 | ct.dev = ports->miport[ports->cur_port]; |
| 3419 | au_get_lr_value(sc, &ct, &lgain, &rgain); |
| 3420 | } else { |
| 3421 | for(i = 0; i < ports->nports; i++) |
| 3422 | if (ports->misel[i] == ct.un.ord) { |
| 3423 | ct.dev = ports->miport[i]; |
| 3424 | if (ct.dev == -1 || |
| 3425 | au_get_lr_value(sc, &ct, |
| 3426 | &lgain, &rgain)) |
| 3427 | goto usemaster; |
| 3428 | else |
| 3429 | break; |
| 3430 | } |
| 3431 | } |
| 3432 | } else { |
| 3433 | ct.type = AUDIO_MIXER_SET; |
| 3434 | if (sc->hw_if->get_port(sc->hw_hdl, &ct)) |
| 3435 | goto bad; |
| 3436 | ct.type = AUDIO_MIXER_VALUE; |
| 3437 | lgain = rgain = n = 0; |
| 3438 | for(i = 0; i < ports->nports; i++) { |
| 3439 | if (ports->misel[i] & ct.un.mask) { |
| 3440 | ct.dev = ports->miport[i]; |
| 3441 | if (ct.dev == -1 || |
| 3442 | au_get_lr_value(sc, &ct, &l, &r)) |
| 3443 | goto usemaster; |
| 3444 | else { |
| 3445 | lgain += l; |
| 3446 | rgain += r; |
| 3447 | n++; |
| 3448 | } |
| 3449 | } |
| 3450 | } |
| 3451 | if (n != 0) { |
| 3452 | lgain /= n; |
| 3453 | rgain /= n; |
| 3454 | } |
| 3455 | } |
| 3456 | } |
| 3457 | bad: |
| 3458 | if (lgain == rgain) { /* handles lgain==rgain==0 */ |
| 3459 | *pgain = lgain; |
| 3460 | *pbalance = AUDIO_MID_BALANCE; |
| 3461 | } else if (lgain < rgain) { |
| 3462 | *pgain = rgain; |
| 3463 | /* balance should be > AUDIO_MID_BALANCE */ |
| 3464 | *pbalance = AUDIO_RIGHT_BALANCE - |
| 3465 | (AUDIO_MID_BALANCE * lgain) / rgain; |
| 3466 | } else /* lgain > rgain */ { |
| 3467 | *pgain = lgain; |
| 3468 | /* balance should be < AUDIO_MID_BALANCE */ |
| 3469 | *pbalance = (AUDIO_MID_BALANCE * rgain) / lgain; |
| 3470 | } |
| 3471 | } |
| 3472 | |
| 3473 | int |
| 3474 | au_set_port(struct audio_softc *sc, struct au_mixer_ports *ports, u_int port) |
| 3475 | { |
| 3476 | mixer_ctrl_t ct; |
| 3477 | int i, error, use_mixerout; |
| 3478 | |
| 3479 | KASSERT(mutex_owned(sc->sc_lock)); |
| 3480 | |
| 3481 | use_mixerout = 1; |
| 3482 | if (port == 0) { |
| 3483 | if (ports->allports == 0) |
| 3484 | return 0; /* Allow this special case. */ |
| 3485 | else if (ports->isdual) { |
| 3486 | if (ports->cur_port == -1) { |
| 3487 | return 0; |
| 3488 | } else { |
| 3489 | port = ports->aumask[ports->cur_port]; |
| 3490 | ports->cur_port = -1; |
| 3491 | use_mixerout = 0; |
| 3492 | } |
| 3493 | } |
| 3494 | } |
| 3495 | if (ports->index == -1) |
| 3496 | return EINVAL; |
| 3497 | ct.dev = ports->index; |
| 3498 | if (ports->isenum) { |
| 3499 | if (port & (port-1)) |
| 3500 | return EINVAL; /* Only one port allowed */ |
| 3501 | ct.type = AUDIO_MIXER_ENUM; |
| 3502 | error = EINVAL; |
| 3503 | for(i = 0; i < ports->nports; i++) |
| 3504 | if (ports->aumask[i] == port) { |
| 3505 | if (ports->isdual && use_mixerout) { |
| 3506 | ct.un.ord = ports->mixerout; |
| 3507 | ports->cur_port = i; |
| 3508 | } else { |
| 3509 | ct.un.ord = ports->misel[i]; |
| 3510 | } |
| 3511 | error = sc->hw_if->set_port(sc->hw_hdl, &ct); |
| 3512 | break; |
| 3513 | } |
| 3514 | } else { |
| 3515 | ct.type = AUDIO_MIXER_SET; |
| 3516 | ct.un.mask = 0; |
| 3517 | for(i = 0; i < ports->nports; i++) |
| 3518 | if (ports->aumask[i] & port) |
| 3519 | ct.un.mask |= ports->misel[i]; |
| 3520 | if (port != 0 && ct.un.mask == 0) |
| 3521 | error = EINVAL; |
| 3522 | else |
| 3523 | error = sc->hw_if->set_port(sc->hw_hdl, &ct); |
| 3524 | } |
| 3525 | if (!error) |
| 3526 | mixer_signal(sc); |
| 3527 | return error; |
| 3528 | } |
| 3529 | |
| 3530 | int |
| 3531 | au_get_port(struct audio_softc *sc, struct au_mixer_ports *ports) |
| 3532 | { |
| 3533 | mixer_ctrl_t ct; |
| 3534 | int i, aumask; |
| 3535 | |
| 3536 | KASSERT(mutex_owned(sc->sc_lock)); |
| 3537 | |
| 3538 | if (ports->index == -1) |
| 3539 | return 0; |
| 3540 | ct.dev = ports->index; |
| 3541 | ct.type = ports->isenum ? AUDIO_MIXER_ENUM : AUDIO_MIXER_SET; |
| 3542 | if (sc->hw_if->get_port(sc->hw_hdl, &ct)) |
| 3543 | return 0; |
| 3544 | aumask = 0; |
| 3545 | if (ports->isenum) { |
| 3546 | if (ports->isdual && ports->cur_port != -1) { |
| 3547 | if (ports->mixerout == ct.un.ord) |
| 3548 | aumask = ports->aumask[ports->cur_port]; |
| 3549 | else |
| 3550 | ports->cur_port = -1; |
| 3551 | } |
| 3552 | if (aumask == 0) |
| 3553 | for(i = 0; i < ports->nports; i++) |
| 3554 | if (ports->misel[i] == ct.un.ord) |
| 3555 | aumask = ports->aumask[i]; |
| 3556 | } else { |
| 3557 | for(i = 0; i < ports->nports; i++) |
| 3558 | if (ct.un.mask & ports->misel[i]) |
| 3559 | aumask |= ports->aumask[i]; |
| 3560 | } |
| 3561 | return aumask; |
| 3562 | } |
| 3563 | |
| 3564 | int |
| 3565 | audiosetinfo(struct audio_softc *sc, struct audio_info *ai) |
| 3566 | { |
| 3567 | stream_filter_list_t pfilters, rfilters; |
| 3568 | audio_params_t pp, rp; |
| 3569 | struct audio_prinfo *r, *p; |
| 3570 | const struct audio_hw_if *hw; |
| 3571 | audio_stream_t *oldpus, *oldrus; |
| 3572 | int setmode; |
| 3573 | int error; |
| 3574 | int np, nr; |
| 3575 | unsigned int blks; |
| 3576 | int oldpblksize, oldrblksize; |
| 3577 | u_int gain; |
| 3578 | bool rbus, pbus; |
| 3579 | bool cleared, modechange, pausechange; |
| 3580 | u_char balance; |
| 3581 | |
| 3582 | KASSERT(mutex_owned(sc->sc_lock)); |
| 3583 | |
| 3584 | hw = sc->hw_if; |
| 3585 | if (hw == NULL) /* HW has not attached */ |
| 3586 | return ENXIO; |
| 3587 | |
| 3588 | DPRINTF(("%s sc=%p ai=%p\n" , __func__, sc, ai)); |
| 3589 | r = &ai->record; |
| 3590 | p = &ai->play; |
| 3591 | rbus = sc->sc_rbus; |
| 3592 | pbus = sc->sc_pbus; |
| 3593 | error = 0; |
| 3594 | cleared = false; |
| 3595 | modechange = false; |
| 3596 | pausechange = false; |
| 3597 | |
| 3598 | pp = sc->sc_pparams; /* Temporary encoding storage in */ |
| 3599 | rp = sc->sc_rparams; /* case setting the modes fails. */ |
| 3600 | nr = np = 0; |
| 3601 | |
| 3602 | if (SPECIFIED(p->sample_rate)) { |
| 3603 | pp.sample_rate = p->sample_rate; |
| 3604 | np++; |
| 3605 | } |
| 3606 | if (SPECIFIED(r->sample_rate)) { |
| 3607 | rp.sample_rate = r->sample_rate; |
| 3608 | nr++; |
| 3609 | } |
| 3610 | if (SPECIFIED(p->encoding)) { |
| 3611 | pp.encoding = p->encoding; |
| 3612 | np++; |
| 3613 | } |
| 3614 | if (SPECIFIED(r->encoding)) { |
| 3615 | rp.encoding = r->encoding; |
| 3616 | nr++; |
| 3617 | } |
| 3618 | if (SPECIFIED(p->precision)) { |
| 3619 | pp.precision = p->precision; |
| 3620 | /* we don't have API to specify validbits */ |
| 3621 | pp.validbits = p->precision; |
| 3622 | np++; |
| 3623 | } |
| 3624 | if (SPECIFIED(r->precision)) { |
| 3625 | rp.precision = r->precision; |
| 3626 | /* we don't have API to specify validbits */ |
| 3627 | rp.validbits = r->precision; |
| 3628 | nr++; |
| 3629 | } |
| 3630 | if (SPECIFIED(p->channels)) { |
| 3631 | pp.channels = p->channels; |
| 3632 | np++; |
| 3633 | } |
| 3634 | if (SPECIFIED(r->channels)) { |
| 3635 | rp.channels = r->channels; |
| 3636 | nr++; |
| 3637 | } |
| 3638 | |
| 3639 | if (!audio_can_capture(sc)) |
| 3640 | nr = 0; |
| 3641 | if (!audio_can_playback(sc)) |
| 3642 | np = 0; |
| 3643 | |
| 3644 | #ifdef AUDIO_DEBUG |
| 3645 | if (audiodebug && nr > 0) |
| 3646 | audio_print_params("audiosetinfo() Setting record params:" , &rp); |
| 3647 | if (audiodebug && np > 0) |
| 3648 | audio_print_params("audiosetinfo() Setting play params:" , &pp); |
| 3649 | #endif |
| 3650 | if (nr > 0 && (error = audio_check_params(&rp))) |
| 3651 | return error; |
| 3652 | if (np > 0 && (error = audio_check_params(&pp))) |
| 3653 | return error; |
| 3654 | |
| 3655 | oldpblksize = sc->sc_pr.blksize; |
| 3656 | oldrblksize = sc->sc_rr.blksize; |
| 3657 | |
| 3658 | setmode = 0; |
| 3659 | if (nr > 0) { |
| 3660 | if (!cleared) { |
| 3661 | audio_clear_intr_unlocked(sc); |
| 3662 | cleared = true; |
| 3663 | } |
| 3664 | modechange = true; |
| 3665 | setmode |= AUMODE_RECORD; |
| 3666 | } |
| 3667 | if (np > 0) { |
| 3668 | if (!cleared) { |
| 3669 | audio_clear_intr_unlocked(sc); |
| 3670 | cleared = true; |
| 3671 | } |
| 3672 | modechange = true; |
| 3673 | setmode |= AUMODE_PLAY; |
| 3674 | } |
| 3675 | |
| 3676 | if (SPECIFIED(ai->mode)) { |
| 3677 | if (!cleared) { |
| 3678 | audio_clear_intr_unlocked(sc); |
| 3679 | cleared = true; |
| 3680 | } |
| 3681 | modechange = true; |
| 3682 | sc->sc_mode = ai->mode; |
| 3683 | if (sc->sc_mode & AUMODE_PLAY_ALL) |
| 3684 | sc->sc_mode |= AUMODE_PLAY; |
| 3685 | if ((sc->sc_mode & AUMODE_PLAY) && !sc->sc_full_duplex) |
| 3686 | /* Play takes precedence */ |
| 3687 | sc->sc_mode &= ~AUMODE_RECORD; |
| 3688 | } |
| 3689 | |
| 3690 | oldpus = sc->sc_pustream; |
| 3691 | oldrus = sc->sc_rustream; |
| 3692 | if (modechange) { |
| 3693 | int indep; |
| 3694 | |
| 3695 | indep = audio_get_props(sc) & AUDIO_PROP_INDEPENDENT; |
| 3696 | if (!indep) { |
| 3697 | if (setmode == AUMODE_RECORD) |
| 3698 | pp = rp; |
| 3699 | else if (setmode == AUMODE_PLAY) |
| 3700 | rp = pp; |
| 3701 | } |
| 3702 | memset(&pfilters, 0, sizeof(pfilters)); |
| 3703 | memset(&rfilters, 0, sizeof(rfilters)); |
| 3704 | pfilters.append = stream_filter_list_append; |
| 3705 | pfilters.prepend = stream_filter_list_prepend; |
| 3706 | pfilters.set = stream_filter_list_set; |
| 3707 | rfilters.append = stream_filter_list_append; |
| 3708 | rfilters.prepend = stream_filter_list_prepend; |
| 3709 | rfilters.set = stream_filter_list_set; |
| 3710 | /* Some device drivers change channels/sample_rate and change |
| 3711 | * no channels/sample_rate. */ |
| 3712 | error = hw->set_params(sc->hw_hdl, setmode, |
| 3713 | sc->sc_mode & (AUMODE_PLAY | AUMODE_RECORD), &pp, &rp, |
| 3714 | &pfilters, &rfilters); |
| 3715 | if (error) { |
| 3716 | DPRINTF(("%s: hw->set_params() failed with %d\n" , |
| 3717 | __func__, error)); |
| 3718 | goto cleanup; |
| 3719 | } |
| 3720 | |
| 3721 | audio_check_params(&pp); |
| 3722 | audio_check_params(&rp); |
| 3723 | if (!indep) { |
| 3724 | /* XXX for !indep device, we have to use the same |
| 3725 | * parameters for the hardware, not userland */ |
| 3726 | if (setmode == AUMODE_RECORD) { |
| 3727 | pp = rp; |
| 3728 | } else if (setmode == AUMODE_PLAY) { |
| 3729 | rp = pp; |
| 3730 | } |
| 3731 | } |
| 3732 | |
| 3733 | if (sc->sc_pr.mmapped && pfilters.req_size > 0) { |
| 3734 | DPRINTF(("%s: mmapped, and filters are requested.\n" , |
| 3735 | __func__)); |
| 3736 | error = EINVAL; |
| 3737 | goto cleanup; |
| 3738 | } |
| 3739 | |
| 3740 | /* construct new filter chain */ |
| 3741 | if (setmode & AUMODE_PLAY) { |
| 3742 | error = audio_setup_pfilters(sc, &pp, &pfilters); |
| 3743 | if (error) |
| 3744 | goto cleanup; |
| 3745 | } |
| 3746 | if (setmode & AUMODE_RECORD) { |
| 3747 | error = audio_setup_rfilters(sc, &rp, &rfilters); |
| 3748 | if (error) |
| 3749 | goto cleanup; |
| 3750 | } |
| 3751 | DPRINTF(("%s: filter setup is completed.\n" , __func__)); |
| 3752 | |
| 3753 | /* userland formats */ |
| 3754 | sc->sc_pparams = pp; |
| 3755 | sc->sc_rparams = rp; |
| 3756 | } |
| 3757 | |
| 3758 | /* Play params can affect the record params, so recalculate blksize. */ |
| 3759 | if (nr > 0 || np > 0) { |
| 3760 | audio_calc_blksize(sc, AUMODE_RECORD); |
| 3761 | audio_calc_blksize(sc, AUMODE_PLAY); |
| 3762 | } |
| 3763 | #ifdef AUDIO_DEBUG |
| 3764 | if (audiodebug > 1 && nr > 0) |
| 3765 | audio_print_params("audiosetinfo() After setting record params:" , |
| 3766 | &sc->sc_rparams); |
| 3767 | if (audiodebug > 1 && np > 0) |
| 3768 | audio_print_params("audiosetinfo() After setting play params:" , |
| 3769 | &sc->sc_pparams); |
| 3770 | #endif |
| 3771 | |
| 3772 | if (SPECIFIED(p->port)) { |
| 3773 | if (!cleared) { |
| 3774 | audio_clear_intr_unlocked(sc); |
| 3775 | cleared = true; |
| 3776 | } |
| 3777 | error = au_set_port(sc, &sc->sc_outports, p->port); |
| 3778 | if (error) |
| 3779 | goto cleanup; |
| 3780 | } |
| 3781 | if (SPECIFIED(r->port)) { |
| 3782 | if (!cleared) { |
| 3783 | audio_clear_intr_unlocked(sc); |
| 3784 | cleared = true; |
| 3785 | } |
| 3786 | error = au_set_port(sc, &sc->sc_inports, r->port); |
| 3787 | if (error) |
| 3788 | goto cleanup; |
| 3789 | } |
| 3790 | if (SPECIFIED(p->gain)) { |
| 3791 | au_get_gain(sc, &sc->sc_outports, &gain, &balance); |
| 3792 | error = au_set_gain(sc, &sc->sc_outports, p->gain, balance); |
| 3793 | if (error) |
| 3794 | goto cleanup; |
| 3795 | } |
| 3796 | if (SPECIFIED(r->gain)) { |
| 3797 | au_get_gain(sc, &sc->sc_inports, &gain, &balance); |
| 3798 | error = au_set_gain(sc, &sc->sc_inports, r->gain, balance); |
| 3799 | if (error) |
| 3800 | goto cleanup; |
| 3801 | } |
| 3802 | |
| 3803 | if (SPECIFIED_CH(p->balance)) { |
| 3804 | au_get_gain(sc, &sc->sc_outports, &gain, &balance); |
| 3805 | error = au_set_gain(sc, &sc->sc_outports, gain, p->balance); |
| 3806 | if (error) |
| 3807 | goto cleanup; |
| 3808 | } |
| 3809 | if (SPECIFIED_CH(r->balance)) { |
| 3810 | au_get_gain(sc, &sc->sc_inports, &gain, &balance); |
| 3811 | error = au_set_gain(sc, &sc->sc_inports, gain, r->balance); |
| 3812 | if (error) |
| 3813 | goto cleanup; |
| 3814 | } |
| 3815 | |
| 3816 | if (SPECIFIED(ai->monitor_gain) && sc->sc_monitor_port != -1) { |
| 3817 | mixer_ctrl_t ct; |
| 3818 | |
| 3819 | ct.dev = sc->sc_monitor_port; |
| 3820 | ct.type = AUDIO_MIXER_VALUE; |
| 3821 | ct.un.value.num_channels = 1; |
| 3822 | ct.un.value.level[AUDIO_MIXER_LEVEL_MONO] = ai->monitor_gain; |
| 3823 | error = sc->hw_if->set_port(sc->hw_hdl, &ct); |
| 3824 | if (error) |
| 3825 | goto cleanup; |
| 3826 | } |
| 3827 | |
| 3828 | if (SPECIFIED_CH(p->pause)) { |
| 3829 | sc->sc_pr.pause = p->pause; |
| 3830 | pbus = !p->pause; |
| 3831 | pausechange = true; |
| 3832 | } |
| 3833 | if (SPECIFIED_CH(r->pause)) { |
| 3834 | sc->sc_rr.pause = r->pause; |
| 3835 | rbus = !r->pause; |
| 3836 | pausechange = true; |
| 3837 | } |
| 3838 | |
| 3839 | if (SPECIFIED(ai->blocksize)) { |
| 3840 | int pblksize, rblksize; |
| 3841 | |
| 3842 | /* Block size specified explicitly. */ |
| 3843 | if (ai->blocksize == 0) { |
| 3844 | if (!cleared) { |
| 3845 | audio_clear_intr_unlocked(sc); |
| 3846 | cleared = true; |
| 3847 | } |
| 3848 | sc->sc_blkset = false; |
| 3849 | audio_calc_blksize(sc, AUMODE_RECORD); |
| 3850 | audio_calc_blksize(sc, AUMODE_PLAY); |
| 3851 | } else { |
| 3852 | sc->sc_blkset = true; |
| 3853 | /* check whether new blocksize changes actually */ |
| 3854 | if (hw->round_blocksize == NULL) { |
| 3855 | if (!cleared) { |
| 3856 | audio_clear_intr_unlocked(sc); |
| 3857 | cleared = true; |
| 3858 | } |
| 3859 | sc->sc_pr.blksize = ai->blocksize; |
| 3860 | sc->sc_rr.blksize = ai->blocksize; |
| 3861 | } else { |
| 3862 | pblksize = hw->round_blocksize(sc->hw_hdl, |
| 3863 | ai->blocksize, AUMODE_PLAY, &sc->sc_pr.s.param); |
| 3864 | rblksize = hw->round_blocksize(sc->hw_hdl, |
| 3865 | ai->blocksize, AUMODE_RECORD, &sc->sc_rr.s.param); |
| 3866 | if (pblksize != sc->sc_pr.blksize || |
| 3867 | rblksize != sc->sc_rr.blksize) { |
| 3868 | if (!cleared) { |
| 3869 | audio_clear_intr_unlocked(sc); |
| 3870 | cleared = true; |
| 3871 | } |
| 3872 | sc->sc_pr.blksize = ai->blocksize; |
| 3873 | sc->sc_rr.blksize = ai->blocksize; |
| 3874 | } |
| 3875 | } |
| 3876 | } |
| 3877 | } |
| 3878 | |
| 3879 | if (SPECIFIED(ai->mode)) { |
| 3880 | if (sc->sc_mode & AUMODE_PLAY) |
| 3881 | audio_init_play(sc); |
| 3882 | if (sc->sc_mode & AUMODE_RECORD) |
| 3883 | audio_init_record(sc); |
| 3884 | } |
| 3885 | |
| 3886 | if (hw->commit_settings) { |
| 3887 | error = hw->commit_settings(sc->hw_hdl); |
| 3888 | if (error) |
| 3889 | goto cleanup; |
| 3890 | } |
| 3891 | |
| 3892 | sc->sc_lastinfo = *ai; |
| 3893 | sc->sc_lastinfovalid = true; |
| 3894 | |
| 3895 | cleanup: |
| 3896 | if (cleared || pausechange) { |
| 3897 | int init_error; |
| 3898 | |
| 3899 | mutex_enter(sc->sc_intr_lock); |
| 3900 | init_error = audio_initbufs(sc); |
| 3901 | if (init_error) goto err; |
| 3902 | if (sc->sc_pr.blksize != oldpblksize || |
| 3903 | sc->sc_rr.blksize != oldrblksize || |
| 3904 | sc->sc_pustream != oldpus || |
| 3905 | sc->sc_rustream != oldrus) |
| 3906 | audio_calcwater(sc); |
| 3907 | if ((sc->sc_mode & AUMODE_PLAY) && |
| 3908 | pbus && !sc->sc_pbus) |
| 3909 | init_error = audiostartp(sc); |
| 3910 | if (!init_error && |
| 3911 | (sc->sc_mode & AUMODE_RECORD) && |
| 3912 | rbus && !sc->sc_rbus) |
| 3913 | init_error = audiostartr(sc); |
| 3914 | err: |
| 3915 | mutex_exit(sc->sc_intr_lock); |
| 3916 | if (init_error) |
| 3917 | return init_error; |
| 3918 | } |
| 3919 | |
| 3920 | /* Change water marks after initializing the buffers. */ |
| 3921 | if (SPECIFIED(ai->hiwat)) { |
| 3922 | blks = ai->hiwat; |
| 3923 | if (blks > sc->sc_pr.maxblks) |
| 3924 | blks = sc->sc_pr.maxblks; |
| 3925 | if (blks < 2) |
| 3926 | blks = 2; |
| 3927 | sc->sc_pr.usedhigh = blks * sc->sc_pr.blksize; |
| 3928 | } |
| 3929 | if (SPECIFIED(ai->lowat)) { |
| 3930 | blks = ai->lowat; |
| 3931 | if (blks > sc->sc_pr.maxblks - 1) |
| 3932 | blks = sc->sc_pr.maxblks - 1; |
| 3933 | sc->sc_pr.usedlow = blks * sc->sc_pr.blksize; |
| 3934 | } |
| 3935 | if (SPECIFIED(ai->hiwat) || SPECIFIED(ai->lowat)) { |
| 3936 | if (sc->sc_pr.usedlow > sc->sc_pr.usedhigh - sc->sc_pr.blksize) |
| 3937 | sc->sc_pr.usedlow = |
| 3938 | sc->sc_pr.usedhigh - sc->sc_pr.blksize; |
| 3939 | } |
| 3940 | |
| 3941 | return error; |
| 3942 | } |
| 3943 | |
| 3944 | int |
| 3945 | audiogetinfo(struct audio_softc *sc, struct audio_info *ai, int buf_only_mode) |
| 3946 | { |
| 3947 | struct audio_prinfo *r, *p; |
| 3948 | const struct audio_hw_if *hw; |
| 3949 | |
| 3950 | KASSERT(mutex_owned(sc->sc_lock)); |
| 3951 | |
| 3952 | r = &ai->record; |
| 3953 | p = &ai->play; |
| 3954 | hw = sc->hw_if; |
| 3955 | if (hw == NULL) /* HW has not attached */ |
| 3956 | return ENXIO; |
| 3957 | |
| 3958 | p->sample_rate = sc->sc_pparams.sample_rate; |
| 3959 | r->sample_rate = sc->sc_rparams.sample_rate; |
| 3960 | p->channels = sc->sc_pparams.channels; |
| 3961 | r->channels = sc->sc_rparams.channels; |
| 3962 | p->precision = sc->sc_pparams.precision; |
| 3963 | r->precision = sc->sc_rparams.precision; |
| 3964 | p->encoding = sc->sc_pparams.encoding; |
| 3965 | r->encoding = sc->sc_rparams.encoding; |
| 3966 | |
| 3967 | if (buf_only_mode) { |
| 3968 | r->port = 0; |
| 3969 | p->port = 0; |
| 3970 | |
| 3971 | r->avail_ports = 0; |
| 3972 | p->avail_ports = 0; |
| 3973 | |
| 3974 | r->gain = 0; |
| 3975 | r->balance = 0; |
| 3976 | |
| 3977 | p->gain = 0; |
| 3978 | p->balance = 0; |
| 3979 | } else { |
| 3980 | r->port = au_get_port(sc, &sc->sc_inports); |
| 3981 | p->port = au_get_port(sc, &sc->sc_outports); |
| 3982 | |
| 3983 | r->avail_ports = sc->sc_inports.allports; |
| 3984 | p->avail_ports = sc->sc_outports.allports; |
| 3985 | |
| 3986 | au_get_gain(sc, &sc->sc_inports, &r->gain, &r->balance); |
| 3987 | au_get_gain(sc, &sc->sc_outports, &p->gain, &p->balance); |
| 3988 | } |
| 3989 | |
| 3990 | if (sc->sc_monitor_port != -1 && buf_only_mode == 0) { |
| 3991 | mixer_ctrl_t ct; |
| 3992 | |
| 3993 | ct.dev = sc->sc_monitor_port; |
| 3994 | ct.type = AUDIO_MIXER_VALUE; |
| 3995 | ct.un.value.num_channels = 1; |
| 3996 | if (sc->hw_if->get_port(sc->hw_hdl, &ct)) |
| 3997 | ai->monitor_gain = 0; |
| 3998 | else |
| 3999 | ai->monitor_gain = |
| 4000 | ct.un.value.level[AUDIO_MIXER_LEVEL_MONO]; |
| 4001 | } else |
| 4002 | ai->monitor_gain = 0; |
| 4003 | |
| 4004 | p->seek = audio_stream_get_used(sc->sc_pustream); |
| 4005 | r->seek = audio_stream_get_used(sc->sc_rustream); |
| 4006 | |
| 4007 | /* |
| 4008 | * XXX samples should be a value for userland data. |
| 4009 | * But drops is a value for HW data. |
| 4010 | */ |
| 4011 | p->samples = (sc->sc_pustream == &sc->sc_pr.s |
| 4012 | ? sc->sc_pr.stamp : sc->sc_pr.fstamp) - sc->sc_pr.drops; |
| 4013 | r->samples = (sc->sc_rustream == &sc->sc_rr.s |
| 4014 | ? sc->sc_rr.stamp : sc->sc_rr.fstamp) - sc->sc_rr.drops; |
| 4015 | |
| 4016 | p->eof = sc->sc_eof; |
| 4017 | r->eof = 0; |
| 4018 | |
| 4019 | p->pause = sc->sc_pr.pause; |
| 4020 | r->pause = sc->sc_rr.pause; |
| 4021 | |
| 4022 | p->error = sc->sc_pr.drops != 0; |
| 4023 | r->error = sc->sc_rr.drops != 0; |
| 4024 | |
| 4025 | p->waiting = r->waiting = 0; /* open never hangs */ |
| 4026 | |
| 4027 | p->open = (sc->sc_open & AUOPEN_WRITE) != 0; |
| 4028 | r->open = (sc->sc_open & AUOPEN_READ) != 0; |
| 4029 | |
| 4030 | p->active = sc->sc_pbus; |
| 4031 | r->active = sc->sc_rbus; |
| 4032 | |
| 4033 | p->buffer_size = sc->sc_pustream ? sc->sc_pustream->bufsize : 0; |
| 4034 | r->buffer_size = sc->sc_rustream ? sc->sc_rustream->bufsize : 0; |
| 4035 | |
| 4036 | ai->blocksize = sc->sc_pr.blksize; |
| 4037 | if (sc->sc_pr.blksize > 0) { |
| 4038 | ai->hiwat = sc->sc_pr.usedhigh / sc->sc_pr.blksize; |
| 4039 | ai->lowat = sc->sc_pr.usedlow / sc->sc_pr.blksize; |
| 4040 | } else |
| 4041 | ai->hiwat = ai->lowat = 0; |
| 4042 | ai->mode = sc->sc_mode; |
| 4043 | |
| 4044 | return 0; |
| 4045 | } |
| 4046 | |
| 4047 | /* |
| 4048 | * Mixer driver |
| 4049 | */ |
| 4050 | int |
| 4051 | mixer_open(dev_t dev, struct audio_softc *sc, int flags, |
| 4052 | int ifmt, struct lwp *l) |
| 4053 | { |
| 4054 | |
| 4055 | KASSERT(mutex_owned(sc->sc_lock)); |
| 4056 | |
| 4057 | if (sc->hw_if == NULL) |
| 4058 | return ENXIO; |
| 4059 | |
| 4060 | DPRINTF(("mixer_open: flags=0x%x sc=%p\n" , flags, sc)); |
| 4061 | |
| 4062 | return 0; |
| 4063 | } |
| 4064 | |
| 4065 | /* |
| 4066 | * Remove a process from those to be signalled on mixer activity. |
| 4067 | */ |
| 4068 | static void |
| 4069 | mixer_remove(struct audio_softc *sc) |
| 4070 | { |
| 4071 | struct mixer_asyncs **pm, *m; |
| 4072 | pid_t pid; |
| 4073 | |
| 4074 | KASSERT(mutex_owned(sc->sc_lock)); |
| 4075 | |
| 4076 | pid = curproc->p_pid; |
| 4077 | for (pm = &sc->sc_async_mixer; *pm; pm = &(*pm)->next) { |
| 4078 | if ((*pm)->pid == pid) { |
| 4079 | m = *pm; |
| 4080 | *pm = m->next; |
| 4081 | kmem_free(m, sizeof(*m)); |
| 4082 | return; |
| 4083 | } |
| 4084 | } |
| 4085 | } |
| 4086 | |
| 4087 | /* |
| 4088 | * Signal all processes waiting for the mixer. |
| 4089 | */ |
| 4090 | static void |
| 4091 | mixer_signal(struct audio_softc *sc) |
| 4092 | { |
| 4093 | struct mixer_asyncs *m; |
| 4094 | proc_t *p; |
| 4095 | |
| 4096 | for (m = sc->sc_async_mixer; m; m = m->next) { |
| 4097 | mutex_enter(proc_lock); |
| 4098 | if ((p = proc_find(m->pid)) != NULL) |
| 4099 | psignal(p, SIGIO); |
| 4100 | mutex_exit(proc_lock); |
| 4101 | } |
| 4102 | } |
| 4103 | |
| 4104 | /* |
| 4105 | * Close a mixer device |
| 4106 | */ |
| 4107 | /* ARGSUSED */ |
| 4108 | int |
| 4109 | mixer_close(struct audio_softc *sc, int flags, int ifmt, struct lwp *l) |
| 4110 | { |
| 4111 | |
| 4112 | KASSERT(mutex_owned(sc->sc_lock)); |
| 4113 | |
| 4114 | DPRINTF(("mixer_close: sc %p\n" , sc)); |
| 4115 | mixer_remove(sc); |
| 4116 | return 0; |
| 4117 | } |
| 4118 | |
| 4119 | int |
| 4120 | mixer_ioctl(struct audio_softc *sc, u_long cmd, void *addr, int flag, |
| 4121 | struct lwp *l) |
| 4122 | { |
| 4123 | const struct audio_hw_if *hw; |
| 4124 | struct mixer_asyncs *ma; |
| 4125 | mixer_ctrl_t *mc; |
| 4126 | int error; |
| 4127 | |
| 4128 | DPRINTF(("mixer_ioctl(%lu,'%c',%lu)\n" , |
| 4129 | IOCPARM_LEN(cmd), (char)IOCGROUP(cmd), cmd&0xff)); |
| 4130 | hw = sc->hw_if; |
| 4131 | error = EINVAL; |
| 4132 | |
| 4133 | /* we can return cached values if we are sleeping */ |
| 4134 | if (cmd != AUDIO_MIXER_READ) |
| 4135 | device_active(sc->dev, DVA_SYSTEM); |
| 4136 | |
| 4137 | switch (cmd) { |
| 4138 | case FIOASYNC: |
| 4139 | if (*(int *)addr) { |
| 4140 | mutex_exit(sc->sc_lock); |
| 4141 | ma = kmem_alloc(sizeof(struct mixer_asyncs), KM_SLEEP); |
| 4142 | mutex_enter(sc->sc_lock); |
| 4143 | } else { |
| 4144 | ma = NULL; |
| 4145 | } |
| 4146 | mixer_remove(sc); /* remove old entry */ |
| 4147 | if (ma != NULL) { |
| 4148 | ma->next = sc->sc_async_mixer; |
| 4149 | ma->pid = curproc->p_pid; |
| 4150 | sc->sc_async_mixer = ma; |
| 4151 | } |
| 4152 | error = 0; |
| 4153 | break; |
| 4154 | |
| 4155 | case AUDIO_GETDEV: |
| 4156 | DPRINTF(("AUDIO_GETDEV\n" )); |
| 4157 | error = hw->getdev(sc->hw_hdl, (audio_device_t *)addr); |
| 4158 | break; |
| 4159 | |
| 4160 | case AUDIO_MIXER_DEVINFO: |
| 4161 | DPRINTF(("AUDIO_MIXER_DEVINFO\n" )); |
| 4162 | ((mixer_devinfo_t *)addr)->un.v.delta = 0; /* default */ |
| 4163 | error = hw->query_devinfo(sc->hw_hdl, (mixer_devinfo_t *)addr); |
| 4164 | break; |
| 4165 | |
| 4166 | case AUDIO_MIXER_READ: |
| 4167 | DPRINTF(("AUDIO_MIXER_READ\n" )); |
| 4168 | mc = (mixer_ctrl_t *)addr; |
| 4169 | |
| 4170 | if (device_is_active(sc->sc_dev)) |
| 4171 | error = hw->get_port(sc->hw_hdl, mc); |
| 4172 | else if (mc->dev >= sc->sc_nmixer_states) |
| 4173 | error = ENXIO; |
| 4174 | else { |
| 4175 | int dev = mc->dev; |
| 4176 | memcpy(mc, &sc->sc_mixer_state[dev], |
| 4177 | sizeof(mixer_ctrl_t)); |
| 4178 | error = 0; |
| 4179 | } |
| 4180 | break; |
| 4181 | |
| 4182 | case AUDIO_MIXER_WRITE: |
| 4183 | DPRINTF(("AUDIO_MIXER_WRITE\n" )); |
| 4184 | error = hw->set_port(sc->hw_hdl, (mixer_ctrl_t *)addr); |
| 4185 | if (!error && hw->commit_settings) |
| 4186 | error = hw->commit_settings(sc->hw_hdl); |
| 4187 | if (!error) |
| 4188 | mixer_signal(sc); |
| 4189 | break; |
| 4190 | |
| 4191 | default: |
| 4192 | if (hw->dev_ioctl) |
| 4193 | error = hw->dev_ioctl(sc->hw_hdl, cmd, addr, flag, l); |
| 4194 | else |
| 4195 | error = EINVAL; |
| 4196 | break; |
| 4197 | } |
| 4198 | DPRINTF(("mixer_ioctl(%lu,'%c',%lu) result %d\n" , |
| 4199 | IOCPARM_LEN(cmd), (char)IOCGROUP(cmd), cmd&0xff, error)); |
| 4200 | return error; |
| 4201 | } |
| 4202 | #endif /* NAUDIO > 0 */ |
| 4203 | |
| 4204 | #include "midi.h" |
| 4205 | |
| 4206 | #if NAUDIO == 0 && (NMIDI > 0 || NMIDIBUS > 0) |
| 4207 | #include <sys/param.h> |
| 4208 | #include <sys/systm.h> |
| 4209 | #include <sys/device.h> |
| 4210 | #include <sys/audioio.h> |
| 4211 | #include <dev/audio_if.h> |
| 4212 | #endif |
| 4213 | |
| 4214 | #if NAUDIO > 0 || (NMIDI > 0 || NMIDIBUS > 0) |
| 4215 | int |
| 4216 | audioprint(void *aux, const char *pnp) |
| 4217 | { |
| 4218 | struct audio_attach_args *arg; |
| 4219 | const char *type; |
| 4220 | |
| 4221 | if (pnp != NULL) { |
| 4222 | arg = aux; |
| 4223 | switch (arg->type) { |
| 4224 | case AUDIODEV_TYPE_AUDIO: |
| 4225 | type = "audio" ; |
| 4226 | break; |
| 4227 | case AUDIODEV_TYPE_MIDI: |
| 4228 | type = "midi" ; |
| 4229 | break; |
| 4230 | case AUDIODEV_TYPE_OPL: |
| 4231 | type = "opl" ; |
| 4232 | break; |
| 4233 | case AUDIODEV_TYPE_MPU: |
| 4234 | type = "mpu" ; |
| 4235 | break; |
| 4236 | default: |
| 4237 | panic("audioprint: unknown type %d" , arg->type); |
| 4238 | } |
| 4239 | aprint_normal("%s at %s" , type, pnp); |
| 4240 | } |
| 4241 | return UNCONF; |
| 4242 | } |
| 4243 | |
| 4244 | #endif /* NAUDIO > 0 || (NMIDI > 0 || NMIDIBUS > 0) */ |
| 4245 | |
| 4246 | #if NAUDIO > 0 |
| 4247 | device_t |
| 4248 | audio_get_device(struct audio_softc *sc) |
| 4249 | { |
| 4250 | return sc->sc_dev; |
| 4251 | } |
| 4252 | #endif |
| 4253 | |
| 4254 | #if NAUDIO > 0 |
| 4255 | static void |
| 4256 | audio_mixer_capture(struct audio_softc *sc) |
| 4257 | { |
| 4258 | mixer_devinfo_t mi; |
| 4259 | mixer_ctrl_t *mc; |
| 4260 | |
| 4261 | KASSERT(mutex_owned(sc->sc_lock)); |
| 4262 | |
| 4263 | for (mi.index = 0;; mi.index++) { |
| 4264 | if (sc->hw_if->query_devinfo(sc->hw_hdl, &mi) != 0) |
| 4265 | break; |
| 4266 | KASSERT(mi.index < sc->sc_nmixer_states); |
| 4267 | if (mi.type == AUDIO_MIXER_CLASS) |
| 4268 | continue; |
| 4269 | mc = &sc->sc_mixer_state[mi.index]; |
| 4270 | mc->dev = mi.index; |
| 4271 | mc->type = mi.type; |
| 4272 | mc->un.value.num_channels = mi.un.v.num_channels; |
| 4273 | (void)sc->hw_if->get_port(sc->hw_hdl, mc); |
| 4274 | } |
| 4275 | |
| 4276 | return; |
| 4277 | } |
| 4278 | |
| 4279 | static void |
| 4280 | audio_mixer_restore(struct audio_softc *sc) |
| 4281 | { |
| 4282 | mixer_devinfo_t mi; |
| 4283 | mixer_ctrl_t *mc; |
| 4284 | |
| 4285 | KASSERT(mutex_owned(sc->sc_lock)); |
| 4286 | |
| 4287 | for (mi.index = 0; ; mi.index++) { |
| 4288 | if (sc->hw_if->query_devinfo(sc->hw_hdl, &mi) != 0) |
| 4289 | break; |
| 4290 | if (mi.type == AUDIO_MIXER_CLASS) |
| 4291 | continue; |
| 4292 | mc = &sc->sc_mixer_state[mi.index]; |
| 4293 | (void)sc->hw_if->set_port(sc->hw_hdl, mc); |
| 4294 | } |
| 4295 | if (sc->hw_if->commit_settings) |
| 4296 | sc->hw_if->commit_settings(sc->hw_hdl); |
| 4297 | |
| 4298 | return; |
| 4299 | } |
| 4300 | |
| 4301 | #ifdef AUDIO_PM_IDLE |
| 4302 | static void |
| 4303 | audio_idle(void *arg) |
| 4304 | { |
| 4305 | device_t dv = arg; |
| 4306 | struct audio_softc *sc = device_private(dv); |
| 4307 | |
| 4308 | #ifdef PNP_DEBUG |
| 4309 | extern int pnp_debug_idle; |
| 4310 | if (pnp_debug_idle) |
| 4311 | printf("%s: idle handler called\n" , device_xname(dv)); |
| 4312 | #endif |
| 4313 | |
| 4314 | sc->sc_idle = true; |
| 4315 | |
| 4316 | /* XXX joerg Make pmf_device_suspend handle children? */ |
| 4317 | if (!pmf_device_suspend(dv, PMF_Q_SELF)) |
| 4318 | return; |
| 4319 | |
| 4320 | if (!pmf_device_suspend(sc->sc_dev, PMF_Q_SELF)) |
| 4321 | pmf_device_resume(dv, PMF_Q_SELF); |
| 4322 | } |
| 4323 | |
| 4324 | static void |
| 4325 | audio_activity(device_t dv, devactive_t type) |
| 4326 | { |
| 4327 | struct audio_softc *sc = device_private(dv); |
| 4328 | |
| 4329 | if (type != DVA_SYSTEM) |
| 4330 | return; |
| 4331 | |
| 4332 | callout_schedule(&sc->sc_idle_counter, audio_idle_timeout * hz); |
| 4333 | |
| 4334 | sc->sc_idle = false; |
| 4335 | if (!device_is_active(dv)) { |
| 4336 | /* XXX joerg How to deal with a failing resume... */ |
| 4337 | pmf_device_resume(sc->sc_dev, PMF_Q_SELF); |
| 4338 | pmf_device_resume(dv, PMF_Q_SELF); |
| 4339 | } |
| 4340 | } |
| 4341 | #endif |
| 4342 | |
| 4343 | static bool |
| 4344 | audio_suspend(device_t dv, const pmf_qual_t *qual) |
| 4345 | { |
| 4346 | struct audio_softc *sc = device_private(dv); |
| 4347 | const struct audio_hw_if *hwp = sc->hw_if; |
| 4348 | |
| 4349 | mutex_enter(sc->sc_lock); |
| 4350 | audio_mixer_capture(sc); |
| 4351 | mutex_enter(sc->sc_intr_lock); |
| 4352 | if (sc->sc_pbus == true) |
| 4353 | hwp->halt_output(sc->hw_hdl); |
| 4354 | if (sc->sc_rbus == true) |
| 4355 | hwp->halt_input(sc->hw_hdl); |
| 4356 | mutex_exit(sc->sc_intr_lock); |
| 4357 | #ifdef AUDIO_PM_IDLE |
| 4358 | callout_halt(&sc->sc_idle_counter, sc->sc_lock); |
| 4359 | #endif |
| 4360 | mutex_exit(sc->sc_lock); |
| 4361 | |
| 4362 | return true; |
| 4363 | } |
| 4364 | |
| 4365 | static bool |
| 4366 | audio_resume(device_t dv, const pmf_qual_t *qual) |
| 4367 | { |
| 4368 | struct audio_softc *sc = device_private(dv); |
| 4369 | |
| 4370 | mutex_enter(sc->sc_lock); |
| 4371 | if (sc->sc_lastinfovalid) |
| 4372 | audiosetinfo(sc, &sc->sc_lastinfo); |
| 4373 | audio_mixer_restore(sc); |
| 4374 | mutex_enter(sc->sc_intr_lock); |
| 4375 | if ((sc->sc_pbus == true) && !sc->sc_pr.pause) |
| 4376 | audiostartp(sc); |
| 4377 | if ((sc->sc_rbus == true) && !sc->sc_rr.pause) |
| 4378 | audiostartr(sc); |
| 4379 | mutex_exit(sc->sc_intr_lock); |
| 4380 | mutex_exit(sc->sc_lock); |
| 4381 | |
| 4382 | return true; |
| 4383 | } |
| 4384 | |
| 4385 | static void |
| 4386 | audio_volume_down(device_t dv) |
| 4387 | { |
| 4388 | struct audio_softc *sc = device_private(dv); |
| 4389 | mixer_devinfo_t mi; |
| 4390 | int newgain; |
| 4391 | u_int gain; |
| 4392 | u_char balance; |
| 4393 | |
| 4394 | mutex_enter(sc->sc_lock); |
| 4395 | if (sc->sc_outports.index == -1 && sc->sc_outports.master != -1) { |
| 4396 | mi.index = sc->sc_outports.master; |
| 4397 | mi.un.v.delta = 0; |
| 4398 | if (sc->hw_if->query_devinfo(sc->hw_hdl, &mi) == 0) { |
| 4399 | au_get_gain(sc, &sc->sc_outports, &gain, &balance); |
| 4400 | newgain = gain - mi.un.v.delta; |
| 4401 | if (newgain < AUDIO_MIN_GAIN) |
| 4402 | newgain = AUDIO_MIN_GAIN; |
| 4403 | au_set_gain(sc, &sc->sc_outports, newgain, balance); |
| 4404 | } |
| 4405 | } |
| 4406 | mutex_exit(sc->sc_lock); |
| 4407 | } |
| 4408 | |
| 4409 | static void |
| 4410 | audio_volume_up(device_t dv) |
| 4411 | { |
| 4412 | struct audio_softc *sc = device_private(dv); |
| 4413 | mixer_devinfo_t mi; |
| 4414 | u_int gain, newgain; |
| 4415 | u_char balance; |
| 4416 | |
| 4417 | mutex_enter(sc->sc_lock); |
| 4418 | if (sc->sc_outports.index == -1 && sc->sc_outports.master != -1) { |
| 4419 | mi.index = sc->sc_outports.master; |
| 4420 | mi.un.v.delta = 0; |
| 4421 | if (sc->hw_if->query_devinfo(sc->hw_hdl, &mi) == 0) { |
| 4422 | au_get_gain(sc, &sc->sc_outports, &gain, &balance); |
| 4423 | newgain = gain + mi.un.v.delta; |
| 4424 | if (newgain > AUDIO_MAX_GAIN) |
| 4425 | newgain = AUDIO_MAX_GAIN; |
| 4426 | au_set_gain(sc, &sc->sc_outports, newgain, balance); |
| 4427 | } |
| 4428 | } |
| 4429 | mutex_exit(sc->sc_lock); |
| 4430 | } |
| 4431 | |
| 4432 | static void |
| 4433 | audio_volume_toggle(device_t dv) |
| 4434 | { |
| 4435 | struct audio_softc *sc = device_private(dv); |
| 4436 | u_int gain, newgain; |
| 4437 | u_char balance; |
| 4438 | |
| 4439 | mutex_enter(sc->sc_lock); |
| 4440 | au_get_gain(sc, &sc->sc_outports, &gain, &balance); |
| 4441 | if (gain != 0) { |
| 4442 | sc->sc_lastgain = gain; |
| 4443 | newgain = 0; |
| 4444 | } else |
| 4445 | newgain = sc->sc_lastgain; |
| 4446 | au_set_gain(sc, &sc->sc_outports, newgain, balance); |
| 4447 | mutex_exit(sc->sc_lock); |
| 4448 | } |
| 4449 | |
| 4450 | static int |
| 4451 | audio_get_props(struct audio_softc *sc) |
| 4452 | { |
| 4453 | const struct audio_hw_if *hw; |
| 4454 | int props; |
| 4455 | |
| 4456 | KASSERT(mutex_owned(sc->sc_lock)); |
| 4457 | |
| 4458 | hw = sc->hw_if; |
| 4459 | props = hw->get_props(sc->hw_hdl); |
| 4460 | |
| 4461 | /* |
| 4462 | * if neither playback nor capture properties are reported, |
| 4463 | * assume both are supported by the device driver |
| 4464 | */ |
| 4465 | if ((props & (AUDIO_PROP_PLAYBACK|AUDIO_PROP_CAPTURE)) == 0) |
| 4466 | props |= (AUDIO_PROP_PLAYBACK | AUDIO_PROP_CAPTURE); |
| 4467 | |
| 4468 | return props; |
| 4469 | } |
| 4470 | |
| 4471 | static bool |
| 4472 | audio_can_playback(struct audio_softc *sc) |
| 4473 | { |
| 4474 | return audio_get_props(sc) & AUDIO_PROP_PLAYBACK ? true : false; |
| 4475 | } |
| 4476 | |
| 4477 | static bool |
| 4478 | audio_can_capture(struct audio_softc *sc) |
| 4479 | { |
| 4480 | return audio_get_props(sc) & AUDIO_PROP_CAPTURE ? true : false; |
| 4481 | } |
| 4482 | |
| 4483 | #endif /* NAUDIO > 0 */ |
| 4484 | |