1 /** $MirOS: src/lib/libossaudio/ossaudio.c,v 1.2 2007/05/16 23:46:48 tg Exp $ */
2 /* $OpenBSD: ossaudio.c,v 1.7 2005/01/19 18:35:04 jason Exp $ */
3 /* $NetBSD: ossaudio.c,v 1.14 2001/05/10 01:53:48 augustss Exp $ */
4
5 /*-
6 * Copyright (c) 1997 The NetBSD Foundation, Inc.
7 * All rights reserved.
8 *
9 * Redistribution and use in source and binary forms, with or without
10 * modification, are permitted provided that the following conditions
11 * are met:
12 * 1. Redistributions of source code must retain the above copyright
13 * notice, this list of conditions and the following disclaimer.
14 * 2. Redistributions in binary form must reproduce the above copyright
15 * notice, this list of conditions and the following disclaimer in the
16 * documentation and/or other materials provided with the distribution.
17 * 3. All advertising materials mentioning features or use of this software
18 * must display the following acknowledgement:
19 * This product includes software developed by the NetBSD
20 * Foundation, Inc. and its contributors.
21 * 4. Neither the name of The NetBSD Foundation nor the names of its
22 * contributors may be used to endorse or promote products derived
23 * from this software without specific prior written permission.
24 *
25 * THIS SOFTWARE IS PROVIDED BY THE NETBSD FOUNDATION, INC. AND CONTRIBUTORS
26 * ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED
27 * TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR
28 * PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE FOUNDATION OR CONTRIBUTORS
29 * BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
30 * CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
31 * SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
32 * INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
33 * CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
34 * ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
35 * POSSIBILITY OF SUCH DAMAGE.
36 */
37
38 /*
39 * This is an OSS (Linux) sound API emulator.
40 * It provides the essentials of the API.
41 */
42
43 /* XXX This file is essentially the same as sys/compat/ossaudio.c.
44 * With some preprocessor magic it could be the same file.
45 */
46
47 #include <stdarg.h>
48 #include <string.h>
49 #include <sys/types.h>
50 #include <sys/ioctl.h>
51 #include <sys/audioio.h>
52 #include <sys/stat.h>
53 #include <errno.h>
54
55 __RCSID("$MirOS: src/lib/libossaudio/ossaudio.c,v 1.2 2007/05/16 23:46:48 tg Exp $");
56
57 #include "soundcard.h"
58 #undef ioctl
59
60 #define GET_DEV(com) ((com) & 0xff)
61
62 #define TO_OSSVOL(x) (((x) * 100 + 127) / 255)
63 #define FROM_OSSVOL(x) ((((x) > 100 ? 100 : (x)) * 255 + 50) / 100)
64
65 static struct audiodevinfo *getdevinfo(int);
66
67 static void setblocksize(int, struct audio_info *);
68
69 static int audio_ioctl(int, unsigned long, void *);
70 static int mixer_ioctl(int, unsigned long, void *);
71 static int opaque_to_enum(struct audiodevinfo *di, audio_mixer_name_t *label, int opq);
72 static int enum_to_ord(struct audiodevinfo *di, int enm);
73 static int enum_to_mask(struct audiodevinfo *di, int enm);
74
75 #define INTARG (*(int*)argp)
76
77 int
_oss_ioctl(int fd,unsigned long com,...)78 _oss_ioctl(int fd, unsigned long com, ...)
79 {
80 va_list ap;
81 void *argp;
82
83 va_start(ap, com);
84 argp = va_arg(ap, void *);
85 va_end(ap);
86 if (IOCGROUP(com) == 'P')
87 return audio_ioctl(fd, com, argp);
88 else if (IOCGROUP(com) == 'M')
89 return mixer_ioctl(fd, com, argp);
90 else
91 return ioctl(fd, com, argp);
92 }
93
94 static int
audio_ioctl(int fd,unsigned long com,void * argp)95 audio_ioctl(int fd, unsigned long com, void *argp)
96 {
97
98 struct audio_info tmpinfo;
99 struct audio_offset tmpoffs;
100 struct audio_buf_info bufinfo;
101 struct count_info cntinfo;
102 struct audio_encoding tmpenc;
103 u_int u;
104 int idat, idata;
105 int retval;
106
107 switch (com) {
108 case SNDCTL_DSP_RESET:
109 retval = ioctl(fd, AUDIO_FLUSH, 0);
110 if (retval < 0)
111 return retval;
112 break;
113 case SNDCTL_DSP_SYNC:
114 retval = ioctl(fd, AUDIO_DRAIN, 0);
115 if (retval < 0)
116 return retval;
117 break;
118 case SNDCTL_DSP_POST:
119 /* This call is merely advisory, and may be a nop. */
120 break;
121 case SNDCTL_DSP_SPEED:
122 AUDIO_INITINFO(&tmpinfo);
123 tmpinfo.play.sample_rate =
124 tmpinfo.record.sample_rate = INTARG;
125 (void) ioctl(fd, AUDIO_SETINFO, &tmpinfo);
126 /* FALLTHRU */
127 case SOUND_PCM_READ_RATE:
128 retval = ioctl(fd, AUDIO_GETINFO, &tmpinfo);
129 if (retval < 0)
130 return retval;
131 INTARG = tmpinfo.play.sample_rate;
132 break;
133 case SNDCTL_DSP_STEREO:
134 AUDIO_INITINFO(&tmpinfo);
135 tmpinfo.play.channels =
136 tmpinfo.record.channels = INTARG ? 2 : 1;
137 (void) ioctl(fd, AUDIO_SETINFO, &tmpinfo);
138 retval = ioctl(fd, AUDIO_GETINFO, &tmpinfo);
139 if (retval < 0)
140 return retval;
141 INTARG = tmpinfo.play.channels - 1;
142 break;
143 case SNDCTL_DSP_GETBLKSIZE:
144 retval = ioctl(fd, AUDIO_GETINFO, &tmpinfo);
145 if (retval < 0)
146 return retval;
147 setblocksize(fd, &tmpinfo);
148 INTARG = tmpinfo.blocksize;
149 break;
150 case SNDCTL_DSP_SETFMT:
151 AUDIO_INITINFO(&tmpinfo);
152 switch (INTARG) {
153 case AFMT_MU_LAW:
154 tmpinfo.play.precision =
155 tmpinfo.record.precision = 8;
156 tmpinfo.play.encoding =
157 tmpinfo.record.encoding = AUDIO_ENCODING_ULAW;
158 break;
159 case AFMT_A_LAW:
160 tmpinfo.play.precision =
161 tmpinfo.record.precision = 8;
162 tmpinfo.play.encoding =
163 tmpinfo.record.encoding = AUDIO_ENCODING_ALAW;
164 break;
165 case AFMT_U8:
166 tmpinfo.play.precision =
167 tmpinfo.record.precision = 8;
168 tmpinfo.play.encoding =
169 tmpinfo.record.encoding = AUDIO_ENCODING_ULINEAR;
170 break;
171 case AFMT_S8:
172 tmpinfo.play.precision =
173 tmpinfo.record.precision = 8;
174 tmpinfo.play.encoding =
175 tmpinfo.record.encoding = AUDIO_ENCODING_SLINEAR;
176 break;
177 case AFMT_S16_LE:
178 tmpinfo.play.precision =
179 tmpinfo.record.precision = 16;
180 tmpinfo.play.encoding =
181 tmpinfo.record.encoding = AUDIO_ENCODING_SLINEAR_LE;
182 break;
183 case AFMT_S16_BE:
184 tmpinfo.play.precision =
185 tmpinfo.record.precision = 16;
186 tmpinfo.play.encoding =
187 tmpinfo.record.encoding = AUDIO_ENCODING_SLINEAR_BE;
188 break;
189 case AFMT_U16_LE:
190 tmpinfo.play.precision =
191 tmpinfo.record.precision = 16;
192 tmpinfo.play.encoding =
193 tmpinfo.record.encoding = AUDIO_ENCODING_ULINEAR_LE;
194 break;
195 case AFMT_U16_BE:
196 tmpinfo.play.precision =
197 tmpinfo.record.precision = 16;
198 tmpinfo.play.encoding =
199 tmpinfo.record.encoding = AUDIO_ENCODING_ULINEAR_BE;
200 break;
201 default:
202 return EINVAL;
203 }
204 (void) ioctl(fd, AUDIO_SETINFO, &tmpinfo);
205 /* FALLTHRU */
206 case SOUND_PCM_READ_BITS:
207 retval = ioctl(fd, AUDIO_GETINFO, &tmpinfo);
208 if (retval < 0)
209 return retval;
210 switch (tmpinfo.play.encoding) {
211 case AUDIO_ENCODING_ULAW:
212 idat = AFMT_MU_LAW;
213 break;
214 case AUDIO_ENCODING_ALAW:
215 idat = AFMT_A_LAW;
216 break;
217 case AUDIO_ENCODING_SLINEAR_LE:
218 if (tmpinfo.play.precision == 16)
219 idat = AFMT_S16_LE;
220 else
221 idat = AFMT_S8;
222 break;
223 case AUDIO_ENCODING_SLINEAR_BE:
224 if (tmpinfo.play.precision == 16)
225 idat = AFMT_S16_BE;
226 else
227 idat = AFMT_S8;
228 break;
229 case AUDIO_ENCODING_ULINEAR_LE:
230 if (tmpinfo.play.precision == 16)
231 idat = AFMT_U16_LE;
232 else
233 idat = AFMT_U8;
234 break;
235 case AUDIO_ENCODING_ULINEAR_BE:
236 if (tmpinfo.play.precision == 16)
237 idat = AFMT_U16_BE;
238 else
239 idat = AFMT_U8;
240 break;
241 case AUDIO_ENCODING_ADPCM:
242 idat = AFMT_IMA_ADPCM;
243 break;
244 }
245 INTARG = idat;
246 break;
247 case SNDCTL_DSP_CHANNELS:
248 AUDIO_INITINFO(&tmpinfo);
249 tmpinfo.play.channels =
250 tmpinfo.record.channels = INTARG;
251 (void) ioctl(fd, AUDIO_SETINFO, &tmpinfo);
252 /* FALLTHRU */
253 case SOUND_PCM_READ_CHANNELS:
254 retval = ioctl(fd, AUDIO_GETINFO, &tmpinfo);
255 if (retval < 0)
256 return retval;
257 INTARG = tmpinfo.play.channels;
258 break;
259 case SOUND_PCM_WRITE_FILTER:
260 case SOUND_PCM_READ_FILTER:
261 errno = EINVAL;
262 return -1; /* XXX unimplemented */
263 case SNDCTL_DSP_SUBDIVIDE:
264 retval = ioctl(fd, AUDIO_GETINFO, &tmpinfo);
265 if (retval < 0)
266 return retval;
267 setblocksize(fd, &tmpinfo);
268 idat = INTARG;
269 if (idat == 0)
270 idat = tmpinfo.play.buffer_size / tmpinfo.blocksize;
271 idat = (tmpinfo.play.buffer_size / idat) & -4;
272 AUDIO_INITINFO(&tmpinfo);
273 tmpinfo.blocksize = idat;
274 retval = ioctl(fd, AUDIO_SETINFO, &tmpinfo);
275 if (retval < 0)
276 return retval;
277 INTARG = tmpinfo.play.buffer_size / tmpinfo.blocksize;
278 break;
279 case SNDCTL_DSP_SETFRAGMENT:
280 AUDIO_INITINFO(&tmpinfo);
281 idat = INTARG;
282 if ((idat & 0xffff) < 4 || (idat & 0xffff) > 17)
283 return EINVAL;
284 tmpinfo.blocksize = 1 << (idat & 0xffff);
285 tmpinfo.hiwat = ((unsigned)idat >> 16) & 0x7fff;
286 if (tmpinfo.hiwat == 0) /* 0 means set to max */
287 tmpinfo.hiwat = 65536;
288 (void) ioctl(fd, AUDIO_SETINFO, &tmpinfo);
289 retval = ioctl(fd, AUDIO_GETINFO, &tmpinfo);
290 if (retval < 0)
291 return retval;
292 u = tmpinfo.blocksize;
293 for(idat = 0; u > 1; idat++, u >>= 1)
294 ;
295 idat |= (tmpinfo.hiwat & 0x7fff) << 16;
296 INTARG = idat;
297 break;
298 case SNDCTL_DSP_GETFMTS:
299 for(idat = 0, tmpenc.index = 0;
300 ioctl(fd, AUDIO_GETENC, &tmpenc) == 0;
301 tmpenc.index++) {
302 switch(tmpenc.encoding) {
303 case AUDIO_ENCODING_ULAW:
304 idat |= AFMT_MU_LAW;
305 break;
306 case AUDIO_ENCODING_ALAW:
307 idat |= AFMT_A_LAW;
308 break;
309 case AUDIO_ENCODING_SLINEAR:
310 idat |= AFMT_S8;
311 break;
312 case AUDIO_ENCODING_SLINEAR_LE:
313 if (tmpenc.precision == 16)
314 idat |= AFMT_S16_LE;
315 else
316 idat |= AFMT_S8;
317 break;
318 case AUDIO_ENCODING_SLINEAR_BE:
319 if (tmpenc.precision == 16)
320 idat |= AFMT_S16_BE;
321 else
322 idat |= AFMT_S8;
323 break;
324 case AUDIO_ENCODING_ULINEAR:
325 idat |= AFMT_U8;
326 break;
327 case AUDIO_ENCODING_ULINEAR_LE:
328 if (tmpenc.precision == 16)
329 idat |= AFMT_U16_LE;
330 else
331 idat |= AFMT_U8;
332 break;
333 case AUDIO_ENCODING_ULINEAR_BE:
334 if (tmpenc.precision == 16)
335 idat |= AFMT_U16_BE;
336 else
337 idat |= AFMT_U8;
338 break;
339 case AUDIO_ENCODING_ADPCM:
340 idat |= AFMT_IMA_ADPCM;
341 break;
342 default:
343 break;
344 }
345 }
346 INTARG = idat;
347 break;
348 case SNDCTL_DSP_GETOSPACE:
349 retval = ioctl(fd, AUDIO_GETINFO, &tmpinfo);
350 if (retval < 0)
351 return retval;
352 setblocksize(fd, &tmpinfo);
353 bufinfo.fragsize = tmpinfo.blocksize;
354 bufinfo.fragments = tmpinfo.hiwat -
355 (tmpinfo.play.seek + tmpinfo.blocksize - 1)/tmpinfo.blocksize;
356 bufinfo.fragstotal = tmpinfo.hiwat;
357 bufinfo.bytes = tmpinfo.hiwat * tmpinfo.blocksize - tmpinfo.play.seek;
358 *(struct audio_buf_info *)argp = bufinfo;
359 break;
360 case SNDCTL_DSP_GETISPACE:
361 retval = ioctl(fd, AUDIO_GETINFO, &tmpinfo);
362 if (retval < 0)
363 return retval;
364 setblocksize(fd, &tmpinfo);
365 bufinfo.fragsize = tmpinfo.blocksize;
366 bufinfo.fragments = tmpinfo.hiwat -
367 (tmpinfo.record.seek + tmpinfo.blocksize - 1)/tmpinfo.blocksize;
368 bufinfo.fragstotal = tmpinfo.hiwat;
369 bufinfo.bytes = tmpinfo.hiwat * tmpinfo.blocksize - tmpinfo.record.seek;
370 *(struct audio_buf_info *)argp = bufinfo;
371 break;
372 case SNDCTL_DSP_NONBLOCK:
373 idat = 1;
374 retval = ioctl(fd, FIONBIO, &idat);
375 if (retval < 0)
376 return retval;
377 break;
378 case SNDCTL_DSP_GETCAPS:
379 retval = ioctl(fd, AUDIO_GETPROPS, &idata);
380 if (retval < 0)
381 return retval;
382 idat = DSP_CAP_TRIGGER; /* pretend we have trigger */
383 if (idata & AUDIO_PROP_FULLDUPLEX)
384 idat |= DSP_CAP_DUPLEX;
385 if (idata & AUDIO_PROP_MMAP)
386 idat |= DSP_CAP_MMAP;
387 INTARG = idat;
388 break;
389 #if 0
390 case SNDCTL_DSP_GETTRIGGER:
391 retval = ioctl(fd, AUDIO_GETINFO, &tmpinfo);
392 if (retval < 0)
393 return retval;
394 idat = (tmpinfo.play.pause ? 0 : PCM_ENABLE_OUTPUT) |
395 (tmpinfo.record.pause ? 0 : PCM_ENABLE_INPUT);
396 retval = copyout(&idat, SCARG(uap, data), sizeof idat);
397 if (retval < 0)
398 return retval;
399 break;
400 case SNDCTL_DSP_SETTRIGGER:
401 AUDIO_INITINFO(&tmpinfo);
402 retval = copyin(SCARG(uap, data), &idat, sizeof idat);
403 if (retval < 0)
404 return retval;
405 tmpinfo.play.pause = (idat & PCM_ENABLE_OUTPUT) == 0;
406 tmpinfo.record.pause = (idat & PCM_ENABLE_INPUT) == 0;
407 (void) ioctl(fd, AUDIO_SETINFO, &tmpinfo);
408 retval = copyout(&idat, SCARG(uap, data), sizeof idat);
409 if (retval < 0)
410 return retval;
411 break;
412 #else
413 case SNDCTL_DSP_GETTRIGGER:
414 case SNDCTL_DSP_SETTRIGGER:
415 /* XXX Do nothing for now. */
416 INTARG = PCM_ENABLE_OUTPUT;
417 break;
418 #endif
419 case SNDCTL_DSP_GETIPTR:
420 retval = ioctl(fd, AUDIO_GETIOFFS, &tmpoffs);
421 if (retval < 0)
422 return retval;
423 cntinfo.bytes = tmpoffs.samples;
424 cntinfo.blocks = tmpoffs.deltablks;
425 cntinfo.ptr = tmpoffs.offset;
426 *(struct count_info *)argp = cntinfo;
427 break;
428 case SNDCTL_DSP_GETOPTR:
429 retval = ioctl(fd, AUDIO_GETOOFFS, &tmpoffs);
430 if (retval < 0)
431 return retval;
432 cntinfo.bytes = tmpoffs.samples;
433 cntinfo.blocks = tmpoffs.deltablks;
434 cntinfo.ptr = tmpoffs.offset;
435 *(struct count_info *)argp = cntinfo;
436 break;
437 case SNDCTL_DSP_SETDUPLEX:
438 idat = 1;
439 retval = ioctl(fd, AUDIO_SETFD, &idat);
440 if (retval < 0)
441 return retval;
442 break;
443 case SNDCTL_DSP_MAPINBUF:
444 case SNDCTL_DSP_MAPOUTBUF:
445 case SNDCTL_DSP_SETSYNCRO:
446 case SNDCTL_DSP_PROFILE:
447 errno = EINVAL;
448 return -1; /* XXX unimplemented */
449 default:
450 errno = EINVAL;
451 return -1;
452 }
453
454 return 0;
455 }
456
457
458 /* If the NetBSD mixer device should have more than NETBSD_MAXDEVS devices
459 * some will not be available to Linux */
460 #define NETBSD_MAXDEVS 64
461 struct audiodevinfo {
462 int done;
463 dev_t dev;
464 ino_t ino;
465 int16_t devmap[SOUND_MIXER_NRDEVICES],
466 rdevmap[NETBSD_MAXDEVS];
467 char names[NETBSD_MAXDEVS][MAX_AUDIO_DEV_LEN];
468 int enum2opaque[NETBSD_MAXDEVS];
469 u_long devmask, recmask, stereomask;
470 u_long caps, source;
471 };
472
473 static int
opaque_to_enum(struct audiodevinfo * di,audio_mixer_name_t * label,int opq)474 opaque_to_enum(struct audiodevinfo *di, audio_mixer_name_t *label, int opq)
475 {
476 int i, o;
477
478 for (i = 0; i < NETBSD_MAXDEVS; i++) {
479 o = di->enum2opaque[i];
480 if (o == opq)
481 break;
482 if (o == -1 && label != NULL &&
483 !strncmp(di->names[i], label->name, sizeof di->names[i])) {
484 di->enum2opaque[i] = opq;
485 break;
486 }
487 }
488 if (i >= NETBSD_MAXDEVS)
489 i = -1;
490 /*printf("opq_to_enum %s %d -> %d\n", label->name, opq, i);*/
491 return (i);
492 }
493
494 static int
enum_to_ord(struct audiodevinfo * di,int enm)495 enum_to_ord(struct audiodevinfo *di, int enm)
496 {
497 if (enm >= NETBSD_MAXDEVS)
498 return (-1);
499
500 /*printf("enum_to_ord %d -> %d\n", enm, di->enum2opaque[enm]);*/
501 return (di->enum2opaque[enm]);
502 }
503
504 static int
enum_to_mask(struct audiodevinfo * di,int enm)505 enum_to_mask(struct audiodevinfo *di, int enm)
506 {
507 int m;
508 if (enm >= NETBSD_MAXDEVS)
509 return (0);
510
511 m = di->enum2opaque[enm];
512 if (m == -1)
513 m = 0;
514 /*printf("enum_to_mask %d -> %d\n", enm, di->enum2opaque[enm]);*/
515 return (m);
516 }
517
518 /*
519 * Collect the audio device information to allow faster
520 * emulation of the Linux mixer ioctls. Cache the information
521 * to eliminate the overhead of repeating all the ioctls needed
522 * to collect the information.
523 */
524 static struct audiodevinfo *
getdevinfo(int fd)525 getdevinfo(int fd)
526 {
527 mixer_devinfo_t mi;
528 int i, j, e;
529 static struct {
530 const char *name;
531 int code;
532 } *dp, devs[] = {
533 { AudioNmicrophone, SOUND_MIXER_MIC },
534 { AudioNline, SOUND_MIXER_LINE },
535 { AudioNcd, SOUND_MIXER_CD },
536 { AudioNdac, SOUND_MIXER_PCM },
537 { AudioNaux, SOUND_MIXER_LINE1 },
538 { AudioNrecord, SOUND_MIXER_IMIX },
539 { AudioNmaster, SOUND_MIXER_VOLUME },
540 { AudioNtreble, SOUND_MIXER_TREBLE },
541 { AudioNbass, SOUND_MIXER_BASS },
542 { AudioNspeaker, SOUND_MIXER_SPEAKER },
543 /* { AudioNheadphone, ?? },*/
544 { AudioNoutput, SOUND_MIXER_OGAIN },
545 { AudioNinput, SOUND_MIXER_IGAIN },
546 /* { AudioNmaster, SOUND_MIXER_SPEAKER },*/
547 /* { AudioNstereo, ?? },*/
548 /* { AudioNmono, ?? },*/
549 { AudioNfmsynth, SOUND_MIXER_SYNTH },
550 /* { AudioNwave, SOUND_MIXER_PCM },*/
551 { AudioNmidi, SOUND_MIXER_SYNTH },
552 /* { AudioNmixerout, ?? },*/
553 { 0, -1 }
554 };
555 static struct audiodevinfo devcache = { 0 };
556 struct audiodevinfo *di = &devcache;
557 struct stat sb;
558
559 /* Figure out what device it is so we can check if the
560 * cached data is valid.
561 */
562 if (fstat(fd, &sb) < 0)
563 return 0;
564 if (di->done && (di->dev == sb.st_dev && di->ino == sb.st_ino))
565 return di;
566
567 di->done = 1;
568 di->dev = sb.st_dev;
569 di->ino = sb.st_ino;
570 di->devmask = 0;
571 di->recmask = 0;
572 di->stereomask = 0;
573 di->source = ~0;
574 di->caps = 0;
575 for(i = 0; i < SOUND_MIXER_NRDEVICES; i++)
576 di->devmap[i] = -1;
577 for(i = 0; i < NETBSD_MAXDEVS; i++) {
578 di->rdevmap[i] = -1;
579 di->names[i][0] = '\0';
580 di->enum2opaque[i] = -1;
581 }
582 for(i = 0; i < NETBSD_MAXDEVS; i++) {
583 mi.index = i;
584 if (ioctl(fd, AUDIO_MIXER_DEVINFO, &mi) < 0)
585 break;
586 switch(mi.type) {
587 case AUDIO_MIXER_VALUE:
588 for(dp = devs; dp->name; dp++)
589 if (strcmp(dp->name, mi.label.name) == 0)
590 break;
591 if (dp->code >= 0) {
592 di->devmap[dp->code] = i;
593 di->rdevmap[i] = dp->code;
594 di->devmask |= 1 << dp->code;
595 if (mi.un.v.num_channels == 2)
596 di->stereomask |= 1 << dp->code;
597 strncpy(di->names[i], mi.label.name,
598 sizeof di->names[i]);
599 }
600 break;
601 }
602 }
603 for(i = 0; i < NETBSD_MAXDEVS; i++) {
604 mi.index = i;
605 if (ioctl(fd, AUDIO_MIXER_DEVINFO, &mi) < 0)
606 break;
607 if (strcmp(mi.label.name, AudioNsource) != 0)
608 continue;
609 di->source = i;
610 switch(mi.type) {
611 case AUDIO_MIXER_ENUM:
612 for(j = 0; j < mi.un.e.num_mem; j++) {
613 e = opaque_to_enum(di,
614 &mi.un.e.member[j].label,
615 mi.un.e.member[j].ord);
616 if (e >= 0)
617 di->recmask |= 1 << di->rdevmap[e];
618 }
619 di->caps = SOUND_CAP_EXCL_INPUT;
620 break;
621 case AUDIO_MIXER_SET:
622 for(j = 0; j < mi.un.s.num_mem; j++) {
623 e = opaque_to_enum(di,
624 &mi.un.s.member[j].label,
625 mi.un.s.member[j].mask);
626 if (e >= 0)
627 di->recmask |= 1 << di->rdevmap[e];
628 }
629 break;
630 }
631 }
632 return di;
633 }
634
635 int
mixer_ioctl(int fd,unsigned long com,void * argp)636 mixer_ioctl(int fd, unsigned long com, void *argp)
637 {
638 struct audiodevinfo *di;
639 struct mixer_info *omi;
640 struct audio_device adev;
641 mixer_ctrl_t mc;
642 int idat = 0;
643 int i;
644 int retval;
645 int l, r, n, error, e;
646
647 di = getdevinfo(fd);
648 if (di == 0)
649 return -1;
650
651 switch (com) {
652 case OSS_GETVERSION:
653 idat = SOUND_VERSION;
654 break;
655 case SOUND_MIXER_INFO:
656 case SOUND_OLD_MIXER_INFO:
657 error = ioctl(fd, AUDIO_GETDEV, &adev);
658 if (error)
659 return (error);
660 omi = argp;
661 if (com == SOUND_MIXER_INFO)
662 omi->modify_counter = 1;
663 strncpy(omi->id, adev.name, sizeof omi->id);
664 strncpy(omi->name, adev.name, sizeof omi->name);
665 return 0;
666 case SOUND_MIXER_READ_RECSRC:
667 if (di->source == (u_long)-1)
668 return EINVAL;
669 mc.dev = di->source;
670 if (di->caps & SOUND_CAP_EXCL_INPUT) {
671 mc.type = AUDIO_MIXER_ENUM;
672 retval = ioctl(fd, AUDIO_MIXER_READ, &mc);
673 if (retval < 0)
674 return retval;
675 e = opaque_to_enum(di, NULL, mc.un.ord);
676 if (e >= 0)
677 idat = 1 << di->rdevmap[e];
678 } else {
679 mc.type = AUDIO_MIXER_SET;
680 retval = ioctl(fd, AUDIO_MIXER_READ, &mc);
681 if (retval < 0)
682 return retval;
683 e = opaque_to_enum(di, NULL, mc.un.mask);
684 if (e >= 0)
685 idat = 1 << di->rdevmap[e];
686 }
687 break;
688 case SOUND_MIXER_READ_DEVMASK:
689 idat = di->devmask;
690 break;
691 case SOUND_MIXER_READ_RECMASK:
692 idat = di->recmask;
693 break;
694 case SOUND_MIXER_READ_STEREODEVS:
695 idat = di->stereomask;
696 break;
697 case SOUND_MIXER_READ_CAPS:
698 idat = di->caps;
699 break;
700 case SOUND_MIXER_WRITE_RECSRC:
701 case SOUND_MIXER_WRITE_R_RECSRC:
702 if (di->source == (u_long)-1)
703 return EINVAL;
704 mc.dev = di->source;
705 idat = INTARG;
706 if (di->caps & SOUND_CAP_EXCL_INPUT) {
707 mc.type = AUDIO_MIXER_ENUM;
708 for(i = 0; i < SOUND_MIXER_NRDEVICES; i++)
709 if (idat & (1 << i))
710 break;
711 if (i >= SOUND_MIXER_NRDEVICES ||
712 di->devmap[i] == -1)
713 return EINVAL;
714 mc.un.ord = enum_to_ord(di, di->devmap[i]);
715 } else {
716 mc.type = AUDIO_MIXER_SET;
717 mc.un.mask = 0;
718 for(i = 0; i < SOUND_MIXER_NRDEVICES; i++) {
719 if (idat & (1 << i)) {
720 if (di->devmap[i] == -1)
721 return EINVAL;
722 mc.un.mask |= enum_to_mask(di, di->devmap[i]);
723 }
724 }
725 }
726 return ioctl(fd, AUDIO_MIXER_WRITE, &mc);
727 default:
728 if (MIXER_READ(SOUND_MIXER_FIRST) <= com &&
729 com < MIXER_READ(SOUND_MIXER_NRDEVICES)) {
730 n = GET_DEV(com);
731 if (di->devmap[n] == -1)
732 return EINVAL;
733 mc.dev = di->devmap[n];
734 mc.type = AUDIO_MIXER_VALUE;
735 doread:
736 mc.un.value.num_channels = di->stereomask & (1<<n) ? 2 : 1;
737 retval = ioctl(fd, AUDIO_MIXER_READ, &mc);
738 if (retval < 0)
739 return retval;
740 if (mc.type != AUDIO_MIXER_VALUE)
741 return EINVAL;
742 if (mc.un.value.num_channels != 2) {
743 l = r = mc.un.value.level[AUDIO_MIXER_LEVEL_MONO];
744 } else {
745 l = mc.un.value.level[AUDIO_MIXER_LEVEL_LEFT];
746 r = mc.un.value.level[AUDIO_MIXER_LEVEL_RIGHT];
747 }
748 idat = TO_OSSVOL(l) | (TO_OSSVOL(r) << 8);
749 break;
750 } else if ((MIXER_WRITE_R(SOUND_MIXER_FIRST) <= com &&
751 com < MIXER_WRITE_R(SOUND_MIXER_NRDEVICES)) ||
752 (MIXER_WRITE(SOUND_MIXER_FIRST) <= com &&
753 com < MIXER_WRITE(SOUND_MIXER_NRDEVICES))) {
754 n = GET_DEV(com);
755 if (di->devmap[n] == -1)
756 return EINVAL;
757 idat = INTARG;
758 l = FROM_OSSVOL( idat & 0xff);
759 r = FROM_OSSVOL((idat >> 8) & 0xff);
760 mc.dev = di->devmap[n];
761 mc.type = AUDIO_MIXER_VALUE;
762 if (di->stereomask & (1<<n)) {
763 mc.un.value.num_channels = 2;
764 mc.un.value.level[AUDIO_MIXER_LEVEL_LEFT] = l;
765 mc.un.value.level[AUDIO_MIXER_LEVEL_RIGHT] = r;
766 } else {
767 mc.un.value.num_channels = 1;
768 mc.un.value.level[AUDIO_MIXER_LEVEL_MONO] = (l+r)/2;
769 }
770 retval = ioctl(fd, AUDIO_MIXER_WRITE, &mc);
771 if (retval < 0)
772 return retval;
773 if (MIXER_WRITE(SOUND_MIXER_FIRST) <= com &&
774 com < MIXER_WRITE(SOUND_MIXER_NRDEVICES))
775 return 0;
776 goto doread;
777 } else {
778 errno = EINVAL;
779 return -1;
780 }
781 }
782 INTARG = idat;
783 return 0;
784 }
785
786 /*
787 * Check that the blocksize is a power of 2 as OSS wants.
788 * If not, set it to be.
789 */
790 static void
setblocksize(int fd,struct audio_info * info)791 setblocksize(int fd, struct audio_info *info)
792 {
793 struct audio_info set;
794 u_int s;
795
796 if (info->blocksize & (info->blocksize-1)) {
797 for(s = 32; s < info->blocksize; s <<= 1)
798 ;
799 AUDIO_INITINFO(&set);
800 set.blocksize = s;
801 ioctl(fd, AUDIO_SETINFO, &set);
802 ioctl(fd, AUDIO_GETINFO, info);
803 }
804 }
805