sbdsp.c revision 1.25
1 /* $NetBSD: sbdsp.c,v 1.25 1996/04/29 20:03:31 christos Exp $ */ 2 3 /* 4 * Copyright (c) 1991-1993 Regents of the University of California. 5 * All rights reserved. 6 * 7 * Redistribution and use in source and binary forms, with or without 8 * modification, are permitted provided that the following conditions 9 * are met: 10 * 1. Redistributions of source code must retain the above copyright 11 * notice, this list of conditions and the following disclaimer. 12 * 2. Redistributions in binary form must reproduce the above copyright 13 * notice, this list of conditions and the following disclaimer in the 14 * documentation and/or other materials provided with the distribution. 15 * 3. All advertising materials mentioning features or use of this software 16 * must display the following acknowledgement: 17 * This product includes software developed by the Computer Systems 18 * Engineering Group at Lawrence Berkeley Laboratory. 19 * 4. Neither the name of the University nor of the Laboratory may be used 20 * to endorse or promote products derived from this software without 21 * specific prior written permission. 22 * 23 * THIS SOFTWARE IS PROVIDED BY THE REGENTS AND CONTRIBUTORS ``AS IS'' AND 24 * ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE 25 * IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE 26 * ARE DISCLAIMED. IN NO EVENT SHALL THE REGENTS OR CONTRIBUTORS BE LIABLE 27 * FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL 28 * DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS 29 * OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) 30 * HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT 31 * LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY 32 * OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF 33 * SUCH DAMAGE. 34 * 35 */ 36 /* 37 * SoundBlaster Pro code provided by John Kohl, based on lots of 38 * information he gleaned from Steve Haehnichen <steve (at) vigra.com>'s 39 * SBlast driver for 386BSD and DOS driver code from Daniel Sachs 40 * <sachs (at) meibm15.cen.uiuc.edu>. 41 */ 42 43 #include <sys/param.h> 44 #include <sys/systm.h> 45 #include <sys/errno.h> 46 #include <sys/ioctl.h> 47 #include <sys/syslog.h> 48 #include <sys/device.h> 49 #include <sys/proc.h> 50 #include <sys/buf.h> 51 #include <vm/vm.h> 52 53 #include <machine/cpu.h> 54 #include <machine/pio.h> 55 56 #include <sys/audioio.h> 57 #include <dev/audio_if.h> 58 59 #include <dev/isa/isavar.h> 60 #include <dev/isa/isadmavar.h> 61 #include <i386/isa/icu.h> /* XXX BROKEN; WHY? */ 62 63 #include <dev/isa/sbreg.h> 64 #include <dev/isa/sbdspvar.h> 65 66 #ifdef AUDIO_DEBUG 67 extern void Dprintf __P((const char *, ...)); 68 #define DPRINTF(x) if (sbdspdebug) Dprintf x 69 int sbdspdebug = 0; 70 #else 71 #define DPRINTF(x) 72 #endif 73 74 #ifndef SBDSP_NPOLL 75 #define SBDSP_NPOLL 3000 76 #endif 77 78 struct { 79 int wdsp; 80 int rdsp; 81 int wmidi; 82 } sberr; 83 84 int sbdsp_srtotc __P((struct sbdsp_softc *sc, int sr, int isdac, 85 int *tcp, int *modep)); 86 u_int sbdsp_jazz16_probe __P((struct sbdsp_softc *)); 87 88 /* 89 * Time constant routines follow. See SBK, section 12. 90 * Although they don't come out and say it (in the docs), 91 * the card clearly uses a 1MHz countdown timer, as the 92 * low-speed formula (p. 12-4) is: 93 * tc = 256 - 10^6 / sr 94 * In high-speed mode, the constant is the upper byte of a 16-bit counter, 95 * and a 256MHz clock is used: 96 * tc = 65536 - 256 * 10^ 6 / sr 97 * Since we can only use the upper byte of the HS TC, the two formulae 98 * are equivalent. (Why didn't they say so?) E.g., 99 * (65536 - 256 * 10 ^ 6 / x) >> 8 = 256 - 10^6 / x 100 * 101 * The crossover point (from low- to high-speed modes) is different 102 * for the SBPRO and SB20. The table on p. 12-5 gives the following data: 103 * 104 * SBPRO SB20 105 * ----- -------- 106 * input ls min 4 KHz 4 KHz 107 * input ls max 23 KHz 13 KHz 108 * input hs max 44.1 KHz 15 KHz 109 * output ls min 4 KHz 4 KHz 110 * output ls max 23 KHz 23 KHz 111 * output hs max 44.1 KHz 44.1 KHz 112 */ 113 #define SB_LS_MIN 0x06 /* 4000 Hz */ 114 #define SB_8K 0x83 /* 8000 Hz */ 115 #define SBPRO_ADC_LS_MAX 0xd4 /* 22727 Hz */ 116 #define SBPRO_ADC_HS_MAX 0xea /* 45454 Hz */ 117 #define SBCLA_ADC_LS_MAX 0xb3 /* 12987 Hz */ 118 #define SBCLA_ADC_HS_MAX 0xbd /* 14925 Hz */ 119 #define SB_DAC_LS_MAX 0xd4 /* 22727 Hz */ 120 #define SB_DAC_HS_MAX 0xea /* 45454 Hz */ 121 122 int sbdsp16_wait __P((int)); 123 void sbdsp_to __P((void *)); 124 void sbdsp_pause __P((struct sbdsp_softc *)); 125 int sbdsp_setrate __P((struct sbdsp_softc *, int, int, int *)); 126 int sbdsp_tctosr __P((struct sbdsp_softc *, int)); 127 int sbdsp_set_timeconst __P((struct sbdsp_softc *, int)); 128 129 #ifdef AUDIO_DEBUG 130 void sb_printsc __P((struct sbdsp_softc *)); 131 #endif 132 133 #ifdef AUDIO_DEBUG 134 void 135 sb_printsc(sc) 136 struct sbdsp_softc *sc; 137 { 138 int i; 139 140 printf("open %d dmachan %d iobase %x\n", 141 sc->sc_open, sc->sc_drq, sc->sc_iobase); 142 printf("irate %d itc %d imode %d orate %d otc %d omode %d encoding %x\n", 143 sc->sc_irate, sc->sc_itc, sc->sc_imode, 144 sc->sc_orate, sc->sc_otc, sc->sc_omode, sc->encoding); 145 printf("outport %d inport %d spkron %d nintr %d\n", 146 sc->out_port, sc->in_port, sc->spkr_state, sc->sc_interrupts); 147 printf("precision %d channels %d intr %x arg %x\n", 148 sc->sc_precision, sc->sc_channels, sc->sc_intr, sc->sc_arg); 149 printf("gain: "); 150 for (i = 0; i < SB_NDEVS; i++) 151 printf("%d ", sc->gain[i]); 152 printf("\n"); 153 } 154 #endif 155 156 /* 157 * Probe / attach routines. 158 */ 159 160 /* 161 * Probe for the soundblaster hardware. 162 */ 163 int 164 sbdsp_probe(sc) 165 struct sbdsp_softc *sc; 166 { 167 168 if (sbdsp_reset(sc) < 0) { 169 DPRINTF(("sbdsp: couldn't reset card\n")); 170 return 0; 171 } 172 /* if flags set, go and probe the jazz16 stuff */ 173 if (sc->sc_dev.dv_cfdata->cf_flags != 0) 174 sc->sc_model = sbdsp_jazz16_probe(sc); 175 else 176 sc->sc_model = sbversion(sc); 177 178 return 1; 179 } 180 181 /* 182 * Try add-on stuff for Jazz16. 183 */ 184 u_int 185 sbdsp_jazz16_probe(sc) 186 struct sbdsp_softc *sc; 187 { 188 static u_char jazz16_irq_conf[16] = { 189 -1, -1, 0x02, 0x03, 190 -1, 0x01, -1, 0x04, 191 -1, 0x02, 0x05, -1, 192 -1, -1, -1, 0x06}; 193 static u_char jazz16_drq_conf[8] = { 194 -1, 0x01, -1, 0x02, 195 -1, 0x03, -1, 0x04}; 196 197 u_int rval = sbversion(sc); 198 register int iobase = sc->sc_iobase; 199 200 if (jazz16_drq_conf[sc->sc_drq] == (u_char)-1 || 201 jazz16_irq_conf[sc->sc_irq] == (u_char)-1) 202 return rval; /* give up, we can't do it. */ 203 outb(JAZZ16_CONFIG_PORT, JAZZ16_WAKEUP); 204 delay(10000); /* delay 10 ms */ 205 outb(JAZZ16_CONFIG_PORT, JAZZ16_SETBASE); 206 outb(JAZZ16_CONFIG_PORT, iobase & 0x70); 207 208 if (sbdsp_reset(sc) < 0) 209 return rval; /* XXX? what else could we do? */ 210 211 if (sbdsp_wdsp(iobase, JAZZ16_READ_VER)) 212 return rval; 213 if (sbdsp_rdsp(iobase) != JAZZ16_VER_JAZZ) 214 return rval; 215 216 if (sbdsp_wdsp(iobase, JAZZ16_SET_DMAINTR) || 217 /* set both 8 & 16-bit drq to same channel, it works fine. */ 218 sbdsp_wdsp(iobase, 219 (jazz16_drq_conf[sc->sc_drq] << 4) | 220 jazz16_drq_conf[sc->sc_drq]) || 221 sbdsp_wdsp(iobase, jazz16_irq_conf[sc->sc_irq])) { 222 DPRINTF(("sbdsp: can't write jazz16 probe stuff")); 223 return rval; 224 } 225 return (rval | MODEL_JAZZ16); 226 } 227 228 /* 229 * Attach hardware to driver, attach hardware driver to audio 230 * pseudo-device driver . 231 */ 232 void 233 sbdsp_attach(sc) 234 struct sbdsp_softc *sc; 235 { 236 237 /* Set defaults */ 238 if (ISSB16CLASS(sc)) 239 sc->sc_irate = sc->sc_orate = 8000; 240 else if (ISSBPROCLASS(sc)) 241 sc->sc_itc = sc->sc_otc = SB_8K; 242 else 243 sc->sc_itc = sc->sc_otc = SB_8K; 244 sc->sc_precision = 8; 245 sc->sc_channels = 1; 246 sc->encoding = AUDIO_ENCODING_ULAW; 247 248 (void) sbdsp_set_in_port(sc, SB_MIC_PORT); 249 (void) sbdsp_set_out_port(sc, SB_SPEAKER); 250 251 if (ISSBPROCLASS(sc)) { 252 int i; 253 254 /* set mixer to default levels, by sending a mixer 255 reset command. */ 256 sbdsp_mix_write(sc, SBP_MIX_RESET, SBP_MIX_RESET); 257 /* then some adjustments :) */ 258 sbdsp_mix_write(sc, SBP_CD_VOL, 259 sbdsp_stereo_vol(SBP_MAXVOL, SBP_MAXVOL)); 260 sbdsp_mix_write(sc, SBP_DAC_VOL, 261 sbdsp_stereo_vol(SBP_MAXVOL, SBP_MAXVOL)); 262 sbdsp_mix_write(sc, SBP_MASTER_VOL, 263 sbdsp_stereo_vol(SBP_MAXVOL/2, SBP_MAXVOL/2)); 264 sbdsp_mix_write(sc, SBP_LINE_VOL, 265 sbdsp_stereo_vol(SBP_MAXVOL, SBP_MAXVOL)); 266 for (i = 0; i < SB_NDEVS; i++) 267 sc->gain[i] = sbdsp_stereo_vol(SBP_MAXVOL, SBP_MAXVOL); 268 sc->in_filter = 0; /* no filters turned on, please */ 269 } 270 271 printf(": dsp v%d.%02d%s\n", 272 SBVER_MAJOR(sc->sc_model), SBVER_MINOR(sc->sc_model), 273 ISJAZZ16(sc) ? ": <Jazz16>" : ""); 274 275 #ifdef notyet 276 sbdsp_mix_write(sc, SBP_SET_IRQ, 0x04); 277 sbdsp_mix_write(sc, SBP_SET_DRQ, 0x22); 278 279 printf("sbdsp_attach: irq=%02x, drq=%02x\n", 280 sbdsp_mix_read(sc, SBP_SET_IRQ), 281 sbdsp_mix_read(sc, SBP_SET_DRQ)); 282 #else 283 if (ISSB16CLASS(sc)) 284 sc->sc_model = 0x0300; 285 #endif 286 } 287 288 /* 289 * Various routines to interface to higher level audio driver 290 */ 291 292 void 293 sbdsp_mix_write(sc, mixerport, val) 294 struct sbdsp_softc *sc; 295 int mixerport; 296 int val; 297 { 298 int iobase = sc->sc_iobase; 299 outb(iobase + SBP_MIXER_ADDR, mixerport); 300 delay(10); 301 outb(iobase + SBP_MIXER_DATA, val); 302 delay(30); 303 } 304 305 int 306 sbdsp_mix_read(sc, mixerport) 307 struct sbdsp_softc *sc; 308 int mixerport; 309 { 310 int iobase = sc->sc_iobase; 311 outb(iobase + SBP_MIXER_ADDR, mixerport); 312 delay(10); 313 return inb(iobase + SBP_MIXER_DATA); 314 } 315 316 int 317 sbdsp_set_in_sr(addr, sr) 318 void *addr; 319 u_long sr; 320 { 321 register struct sbdsp_softc *sc = addr; 322 323 if (ISSB16CLASS(sc)) 324 return (sbdsp_setrate(sc, sr, SB_INPUT_RATE, &sc->sc_irate)); 325 else 326 return (sbdsp_srtotc(sc, sr, SB_INPUT_RATE, &sc->sc_itc, &sc->sc_imode)); 327 } 328 329 u_long 330 sbdsp_get_in_sr(addr) 331 void *addr; 332 { 333 register struct sbdsp_softc *sc = addr; 334 335 if (ISSB16CLASS(sc)) 336 return (sc->sc_irate); 337 else 338 return (sbdsp_tctosr(sc, sc->sc_itc)); 339 } 340 341 int 342 sbdsp_set_out_sr(addr, sr) 343 void *addr; 344 u_long sr; 345 { 346 register struct sbdsp_softc *sc = addr; 347 348 if (ISSB16CLASS(sc)) 349 return (sbdsp_setrate(sc, sr, SB_OUTPUT_RATE, &sc->sc_orate)); 350 else 351 return (sbdsp_srtotc(sc, sr, SB_OUTPUT_RATE, &sc->sc_otc, &sc->sc_omode)); 352 } 353 354 u_long 355 sbdsp_get_out_sr(addr) 356 void *addr; 357 { 358 register struct sbdsp_softc *sc = addr; 359 360 if (ISSB16CLASS(sc)) 361 return (sc->sc_orate); 362 else 363 return (sbdsp_tctosr(sc, sc->sc_otc)); 364 } 365 366 int 367 sbdsp_query_encoding(addr, fp) 368 void *addr; 369 struct audio_encoding *fp; 370 { 371 switch (fp->index) { 372 case 0: 373 strcpy(fp->name, AudioEmulaw); 374 fp->format_id = AUDIO_ENCODING_ULAW; 375 break; 376 case 1: 377 strcpy(fp->name, AudioEpcm16); 378 fp->format_id = AUDIO_ENCODING_PCM16; 379 break; 380 default: 381 return (EINVAL); 382 } 383 return (0); 384 } 385 386 int 387 sbdsp_set_encoding(addr, encoding) 388 void *addr; 389 u_int encoding; 390 { 391 register struct sbdsp_softc *sc = addr; 392 393 switch (encoding) { 394 case AUDIO_ENCODING_ULAW: 395 sc->encoding = AUDIO_ENCODING_ULAW; 396 break; 397 case AUDIO_ENCODING_LINEAR: 398 sc->encoding = AUDIO_ENCODING_LINEAR; 399 break; 400 default: 401 return (EINVAL); 402 } 403 404 return (0); 405 } 406 407 int 408 sbdsp_get_encoding(addr) 409 void *addr; 410 { 411 register struct sbdsp_softc *sc = addr; 412 413 return (sc->encoding); 414 } 415 416 int 417 sbdsp_set_precision(addr, precision) 418 void *addr; 419 u_int precision; 420 { 421 register struct sbdsp_softc *sc = addr; 422 423 if (ISSB16CLASS(sc) || ISJAZZ16(sc)) { 424 if (precision != 16 && precision != 8) 425 return (EINVAL); 426 sc->sc_precision = precision; 427 } else { 428 if (precision != 8) 429 return (EINVAL); 430 sc->sc_precision = precision; 431 } 432 433 return (0); 434 } 435 436 int 437 sbdsp_get_precision(addr) 438 void *addr; 439 { 440 register struct sbdsp_softc *sc = addr; 441 442 return (sc->sc_precision); 443 } 444 445 int 446 sbdsp_set_channels(addr, channels) 447 void *addr; 448 int channels; 449 { 450 register struct sbdsp_softc *sc = addr; 451 452 if (ISSBPROCLASS(sc)) { 453 if (channels != 1 && channels != 2) 454 return (EINVAL); 455 sc->sc_channels = channels; 456 sc->sc_dmadir = SB_DMA_NONE; 457 /* 458 * XXXX 459 * With 2 channels, SBPro can't do more than 22kHz. 460 * No framework to check this. 461 */ 462 } else { 463 if (channels != 1) 464 return (EINVAL); 465 sc->sc_channels = channels; 466 } 467 468 return (0); 469 } 470 471 int 472 sbdsp_get_channels(addr) 473 void *addr; 474 { 475 register struct sbdsp_softc *sc = addr; 476 477 return (sc->sc_channels); 478 } 479 480 int 481 sbdsp_set_ifilter(addr, which) 482 void *addr; 483 int which; 484 { 485 register struct sbdsp_softc *sc = addr; 486 int mixval; 487 488 if (ISSBPROCLASS(sc)) { 489 mixval = sbdsp_mix_read(sc, SBP_INFILTER) & ~SBP_IFILTER_MASK; 490 switch (which) { 491 case 0: 492 mixval |= SBP_FILTER_OFF; 493 break; 494 case SBP_TREBLE_EQ: 495 mixval |= SBP_FILTER_ON | SBP_IFILTER_HIGH; 496 break; 497 case SBP_BASS_EQ: 498 mixval |= SBP_FILTER_ON | SBP_IFILTER_LOW; 499 break; 500 default: 501 return (EINVAL); 502 } 503 sc->in_filter = mixval & SBP_IFILTER_MASK; 504 sbdsp_mix_write(sc, SBP_INFILTER, mixval); 505 return (0); 506 } else 507 return (EINVAL); 508 } 509 510 int 511 sbdsp_get_ifilter(addr) 512 void *addr; 513 { 514 register struct sbdsp_softc *sc = addr; 515 516 if (ISSBPROCLASS(sc)) { 517 sc->in_filter = 518 sbdsp_mix_read(sc, SBP_INFILTER) & SBP_IFILTER_MASK; 519 switch (sc->in_filter) { 520 case SBP_FILTER_ON|SBP_IFILTER_HIGH: 521 return (SBP_TREBLE_EQ); 522 case SBP_FILTER_ON|SBP_IFILTER_LOW: 523 return (SBP_BASS_EQ); 524 case SBP_FILTER_OFF: 525 default: 526 return (0); 527 } 528 } else 529 return (0); 530 } 531 532 int 533 sbdsp_set_out_port(addr, port) 534 void *addr; 535 int port; 536 { 537 register struct sbdsp_softc *sc = addr; 538 539 sc->out_port = port; /* Just record it */ 540 541 return (0); 542 } 543 544 int 545 sbdsp_get_out_port(addr) 546 void *addr; 547 { 548 register struct sbdsp_softc *sc = addr; 549 550 return (sc->out_port); 551 } 552 553 554 int 555 sbdsp_set_in_port(addr, port) 556 void *addr; 557 int port; 558 { 559 register struct sbdsp_softc *sc = addr; 560 int mixport, sbport; 561 562 if (ISSBPROCLASS(sc)) { 563 switch (port) { 564 case SB_MIC_PORT: 565 sbport = SBP_FROM_MIC; 566 mixport = SBP_MIC_VOL; 567 break; 568 case SB_LINE_IN_PORT: 569 sbport = SBP_FROM_LINE; 570 mixport = SBP_LINE_VOL; 571 break; 572 case SB_CD_PORT: 573 sbport = SBP_FROM_CD; 574 mixport = SBP_CD_VOL; 575 break; 576 case SB_DAC_PORT: 577 case SB_FM_PORT: 578 default: 579 return (EINVAL); 580 } 581 } else { 582 switch (port) { 583 case SB_MIC_PORT: 584 sbport = SBP_FROM_MIC; 585 mixport = SBP_MIC_VOL; 586 break; 587 default: 588 return (EINVAL); 589 } 590 } 591 592 sc->in_port = port; /* Just record it */ 593 594 if (ISSBPROCLASS(sc)) { 595 /* record from that port */ 596 sbdsp_mix_write(sc, SBP_RECORD_SOURCE, 597 SBP_RECORD_FROM(sbport, SBP_FILTER_OFF, SBP_IFILTER_HIGH)); 598 /* fetch gain from that port */ 599 sc->gain[port] = sbdsp_mix_read(sc, mixport); 600 } 601 602 return (0); 603 } 604 605 int 606 sbdsp_get_in_port(addr) 607 void *addr; 608 { 609 register struct sbdsp_softc *sc = addr; 610 611 return (sc->in_port); 612 } 613 614 615 int 616 sbdsp_speaker_ctl(addr, newstate) 617 void *addr; 618 int newstate; 619 { 620 register struct sbdsp_softc *sc = addr; 621 622 if ((newstate == SPKR_ON) && 623 (sc->spkr_state == SPKR_OFF)) { 624 sbdsp_spkron(sc); 625 sc->spkr_state = SPKR_ON; 626 } 627 if ((newstate == SPKR_OFF) && 628 (sc->spkr_state == SPKR_ON)) { 629 sbdsp_spkroff(sc); 630 sc->spkr_state = SPKR_OFF; 631 } 632 return(0); 633 } 634 635 int 636 sbdsp_round_blocksize(addr, blk) 637 void *addr; 638 int blk; 639 { 640 register struct sbdsp_softc *sc = addr; 641 642 sc->sc_last_hs_size = 0; 643 644 /* Higher speeds need bigger blocks to avoid popping and silence gaps. */ 645 if (blk < NBPG/4 || blk > NBPG/2) { 646 if (ISSB16CLASS(sc)) { 647 if (sc->sc_orate > 8000 || sc->sc_irate > 8000) 648 blk = NBPG/2; 649 } else { 650 if (sc->sc_otc > SB_8K || sc->sc_itc < SB_8K) 651 blk = NBPG/2; 652 } 653 } 654 /* don't try to DMA too much at once, though. */ 655 if (blk > NBPG) 656 blk = NBPG; 657 if (sc->sc_channels == 2) 658 return (blk & ~1); /* must be even to preserve stereo separation */ 659 else 660 return (blk); /* Anything goes :-) */ 661 } 662 663 int 664 sbdsp_commit_settings(addr) 665 void *addr; 666 { 667 register struct sbdsp_softc *sc = addr; 668 669 /* due to potentially unfortunate ordering in the above layers, 670 re-do a few sets which may be important--input gains 671 (adjust the proper channels), number of input channels (hit the 672 record rate and set mode) */ 673 674 if (ISSBPRO(sc)) { 675 /* 676 * With 2 channels, SBPro can't do more than 22kHz. 677 * Whack the rates down to speed if necessary. 678 * Reset the time constant anyway 679 * because it may have been adjusted with a different number 680 * of channels, which means it might have computed the wrong 681 * mode (low/high speed). 682 */ 683 if (sc->sc_channels == 2 && 684 sbdsp_tctosr(sc, sc->sc_itc) > 22727) { 685 sbdsp_srtotc(sc, 22727, SB_INPUT_RATE, 686 &sc->sc_itc, &sc->sc_imode); 687 } else 688 sbdsp_srtotc(sc, sbdsp_tctosr(sc, sc->sc_itc), 689 SB_INPUT_RATE, &sc->sc_itc, 690 &sc->sc_imode); 691 692 if (sc->sc_channels == 2 && 693 sbdsp_tctosr(sc, sc->sc_otc) > 22727) { 694 sbdsp_srtotc(sc, 22727, SB_OUTPUT_RATE, 695 &sc->sc_otc, &sc->sc_omode); 696 } else 697 sbdsp_srtotc(sc, sbdsp_tctosr(sc, sc->sc_otc), 698 SB_OUTPUT_RATE, &sc->sc_otc, 699 &sc->sc_omode); 700 } 701 if (ISSB16CLASS(sc) || ISJAZZ16(sc)) { 702 if (sc->encoding == AUDIO_ENCODING_ULAW && 703 sc->sc_precision == 16) { 704 sc->sc_precision = 8; 705 return EINVAL; /* XXX what should we really do? */ 706 } 707 } 708 /* 709 * XXX 710 * Should wait for chip to be idle. 711 */ 712 sc->sc_dmadir = SB_DMA_NONE; 713 714 return 0; 715 } 716 717 718 int 719 sbdsp_open(sc, dev, flags) 720 register struct sbdsp_softc *sc; 721 dev_t dev; 722 int flags; 723 { 724 DPRINTF(("sbdsp_open: sc=0x%x\n", sc)); 725 726 if (sc->sc_open != 0 || sbdsp_reset(sc) != 0) 727 return ENXIO; 728 729 sc->sc_open = 1; 730 sc->sc_mintr = 0; 731 if (ISSBPROCLASS(sc) && 732 sbdsp_wdsp(sc->sc_iobase, SB_DSP_RECORD_MONO) < 0) { 733 DPRINTF(("sbdsp_open: can't set mono mode\n")); 734 /* we'll readjust when it's time for DMA. */ 735 } 736 737 /* 738 * Leave most things as they were; users must change things if 739 * the previous process didn't leave it they way they wanted. 740 * Looked at another way, it's easy to set up a configuration 741 * in one program and leave it for another to inherit. 742 */ 743 DPRINTF(("sbdsp_open: opened\n")); 744 745 return 0; 746 } 747 748 void 749 sbdsp_close(addr) 750 void *addr; 751 { 752 struct sbdsp_softc *sc = addr; 753 754 DPRINTF(("sbdsp_close: sc=0x%x\n", sc)); 755 756 sc->sc_open = 0; 757 sbdsp_spkroff(sc); 758 sc->spkr_state = SPKR_OFF; 759 sc->sc_mintr = 0; 760 sbdsp_haltdma(sc); 761 762 DPRINTF(("sbdsp_close: closed\n")); 763 } 764 765 /* 766 * Lower-level routines 767 */ 768 769 /* 770 * Reset the card. 771 * Return non-zero if the card isn't detected. 772 */ 773 int 774 sbdsp_reset(sc) 775 register struct sbdsp_softc *sc; 776 { 777 register int iobase = sc->sc_iobase; 778 779 sc->sc_intr = 0; 780 if (sc->sc_dmadir != SB_DMA_NONE) { 781 isa_dmaabort(sc->sc_drq); 782 sc->sc_dmadir = SB_DMA_NONE; 783 } 784 sc->sc_last_hs_size = 0; 785 786 /* 787 * See SBK, section 11.3. 788 * We pulse a reset signal into the card. 789 * Gee, what a brilliant hardware design. 790 */ 791 outb(iobase + SBP_DSP_RESET, 1); 792 delay(10); 793 outb(iobase + SBP_DSP_RESET, 0); 794 delay(30); 795 if (sbdsp_rdsp(iobase) != SB_MAGIC) 796 return -1; 797 798 return 0; 799 } 800 801 int 802 sbdsp16_wait(iobase) 803 int iobase; 804 { 805 register int i; 806 807 for (i = SBDSP_NPOLL; --i >= 0; ) { 808 register u_char x; 809 x = inb(iobase + SBP_DSP_WSTAT); 810 delay(10); 811 if ((x & SB_DSP_BUSY) == 0) 812 continue; 813 return 0; 814 } 815 ++sberr.wdsp; 816 return -1; 817 } 818 819 /* 820 * Write a byte to the dsp. 821 * XXX We are at the mercy of the card as we use a 822 * polling loop and wait until it can take the byte. 823 */ 824 int 825 sbdsp_wdsp(int iobase, int v) 826 { 827 register int i; 828 829 for (i = SBDSP_NPOLL; --i >= 0; ) { 830 register u_char x; 831 x = inb(iobase + SBP_DSP_WSTAT); 832 delay(10); 833 if ((x & SB_DSP_BUSY) != 0) 834 continue; 835 outb(iobase + SBP_DSP_WRITE, v); 836 delay(10); 837 return 0; 838 } 839 ++sberr.wdsp; 840 return -1; 841 } 842 843 /* 844 * Read a byte from the DSP, using polling. 845 */ 846 int 847 sbdsp_rdsp(int iobase) 848 { 849 register int i; 850 851 for (i = SBDSP_NPOLL; --i >= 0; ) { 852 register u_char x; 853 x = inb(iobase + SBP_DSP_RSTAT); 854 delay(10); 855 if ((x & SB_DSP_READY) == 0) 856 continue; 857 x = inb(iobase + SBP_DSP_READ); 858 delay(10); 859 return x; 860 } 861 ++sberr.rdsp; 862 return -1; 863 } 864 865 /* 866 * Doing certain things (like toggling the speaker) make 867 * the SB hardware go away for a while, so pause a little. 868 */ 869 void 870 sbdsp_to(arg) 871 void *arg; 872 { 873 wakeup(arg); 874 } 875 876 void 877 sbdsp_pause(sc) 878 struct sbdsp_softc *sc; 879 { 880 extern int hz; 881 882 timeout(sbdsp_to, sbdsp_to, hz/8); 883 (void)tsleep(sbdsp_to, PWAIT, "sbpause", 0); 884 } 885 886 /* 887 * Turn on the speaker. The SBK documention says this operation 888 * can take up to 1/10 of a second. Higher level layers should 889 * probably let the task sleep for this amount of time after 890 * calling here. Otherwise, things might not work (because 891 * sbdsp_wdsp() and sbdsp_rdsp() will probably timeout.) 892 * 893 * These engineers had their heads up their ass when 894 * they designed this card. 895 */ 896 void 897 sbdsp_spkron(sc) 898 struct sbdsp_softc *sc; 899 { 900 (void)sbdsp_wdsp(sc->sc_iobase, SB_DSP_SPKR_ON); 901 sbdsp_pause(sc); 902 } 903 904 /* 905 * Turn off the speaker; see comment above. 906 */ 907 void 908 sbdsp_spkroff(sc) 909 struct sbdsp_softc *sc; 910 { 911 (void)sbdsp_wdsp(sc->sc_iobase, SB_DSP_SPKR_OFF); 912 sbdsp_pause(sc); 913 } 914 915 /* 916 * Read the version number out of the card. Return major code 917 * in high byte, and minor code in low byte. 918 */ 919 short 920 sbversion(sc) 921 struct sbdsp_softc *sc; 922 { 923 register int iobase = sc->sc_iobase; 924 short v; 925 926 if (sbdsp_wdsp(iobase, SB_DSP_VERSION) < 0) 927 return 0; 928 v = sbdsp_rdsp(iobase) << 8; 929 v |= sbdsp_rdsp(iobase); 930 return ((v >= 0) ? v : 0); 931 } 932 933 /* 934 * Halt a DMA in progress. A low-speed transfer can be 935 * resumed with sbdsp_contdma(). 936 */ 937 int 938 sbdsp_haltdma(addr) 939 void *addr; 940 { 941 register struct sbdsp_softc *sc = addr; 942 943 DPRINTF(("sbdsp_haltdma: sc=0x%x\n", sc)); 944 945 sbdsp_reset(sc); 946 return 0; 947 } 948 949 int 950 sbdsp_contdma(addr) 951 void *addr; 952 { 953 register struct sbdsp_softc *sc = addr; 954 955 DPRINTF(("sbdsp_contdma: sc=0x%x\n", sc)); 956 957 /* XXX how do we reinitialize the DMA controller state? do we care? */ 958 (void)sbdsp_wdsp(sc->sc_iobase, SB_DSP_CONT); 959 return(0); 960 } 961 962 int 963 sbdsp_setrate(sc, sr, isdac, ratep) 964 register struct sbdsp_softc *sc; 965 int sr; 966 int isdac; 967 int *ratep; 968 { 969 970 /* 971 * XXXX 972 * More checks here? 973 */ 974 if (sr < 5000 || sr > 44100) 975 return (EINVAL); 976 *ratep = sr; 977 return (0); 978 } 979 980 /* 981 * Convert a linear sampling rate into the DAC time constant. 982 * Set *mode to indicate the high/low-speed DMA operation. 983 * Because of limitations of the card, not all rates are possible. 984 * We return the time constant of the closest possible rate. 985 * The sampling rate limits are different for the DAC and ADC, 986 * so isdac indicates output, and !isdac indicates input. 987 */ 988 int 989 sbdsp_srtotc(sc, sr, isdac, tcp, modep) 990 register struct sbdsp_softc *sc; 991 int sr; 992 int isdac; 993 int *tcp, *modep; 994 { 995 int tc, realtc, mode; 996 997 /* 998 * Don't forget to compute which mode we'll be in based on whether 999 * we need to double the rate for stereo on SBPRO. 1000 */ 1001 1002 if (sr == 0) { 1003 tc = SB_LS_MIN; 1004 mode = SB_ADAC_LS; 1005 goto out; 1006 } 1007 1008 tc = 256 - (1000000 / sr); 1009 1010 if (sc->sc_channels == 2 && ISSBPRO(sc)) 1011 /* compute based on 2x sample rate when needed */ 1012 realtc = 256 - ( 500000 / sr); 1013 else 1014 realtc = tc; 1015 1016 if (tc < SB_LS_MIN) { 1017 tc = SB_LS_MIN; 1018 mode = SB_ADAC_LS; /* NB: 2x minimum speed is still low 1019 * speed mode. */ 1020 goto out; 1021 } else if (isdac) { 1022 if (realtc <= SB_DAC_LS_MAX) 1023 mode = SB_ADAC_LS; 1024 else { 1025 mode = SB_ADAC_HS; 1026 if (tc > SB_DAC_HS_MAX) 1027 tc = SB_DAC_HS_MAX; 1028 } 1029 } else { 1030 int adc_ls_max, adc_hs_max; 1031 1032 /* XXX use better rounding--compare distance to nearest tc on both 1033 sides of requested speed */ 1034 if (ISSBPROCLASS(sc)) { 1035 adc_ls_max = SBPRO_ADC_LS_MAX; 1036 adc_hs_max = SBPRO_ADC_HS_MAX; 1037 } else { 1038 adc_ls_max = SBCLA_ADC_LS_MAX; 1039 adc_hs_max = SBCLA_ADC_HS_MAX; 1040 } 1041 1042 if (realtc <= adc_ls_max) 1043 mode = SB_ADAC_LS; 1044 else { 1045 mode = SB_ADAC_HS; 1046 if (tc > adc_hs_max) 1047 tc = adc_hs_max; 1048 } 1049 } 1050 1051 out: 1052 *tcp = tc; 1053 *modep = mode; 1054 return (0); 1055 } 1056 1057 /* 1058 * Convert a DAC time constant to a sampling rate. 1059 * See SBK, section 12. 1060 */ 1061 int 1062 sbdsp_tctosr(sc, tc) 1063 register struct sbdsp_softc *sc; 1064 int tc; 1065 { 1066 int adc; 1067 1068 if (ISSBPROCLASS(sc)) 1069 adc = SBPRO_ADC_HS_MAX; 1070 else 1071 adc = SBCLA_ADC_HS_MAX; 1072 1073 if (tc > adc) 1074 tc = adc; 1075 1076 return (1000000 / (256 - tc)); 1077 } 1078 1079 int 1080 sbdsp_set_timeconst(sc, tc) 1081 register struct sbdsp_softc *sc; 1082 int tc; 1083 { 1084 register int iobase; 1085 1086 /* 1087 * A SBPro in stereo mode uses time constants at double the 1088 * actual rate. 1089 */ 1090 if (ISSBPRO(sc) && sc->sc_channels == 2) 1091 tc = 256 - ((256 - tc) / 2); 1092 1093 DPRINTF(("sbdsp_set_timeconst: sc=%p tc=%d\n", sc, tc)); 1094 1095 iobase = sc->sc_iobase; 1096 if (sbdsp_wdsp(iobase, SB_DSP_TIMECONST) < 0 || 1097 sbdsp_wdsp(iobase, tc) < 0) 1098 return (EIO); 1099 1100 return (0); 1101 } 1102 1103 int 1104 sbdsp_dma_input(addr, p, cc, intr, arg) 1105 void *addr; 1106 void *p; 1107 int cc; 1108 void (*intr) __P((void *)); 1109 void *arg; 1110 { 1111 register struct sbdsp_softc *sc = addr; 1112 register int iobase; 1113 1114 #ifdef AUDIO_DEBUG 1115 if (sbdspdebug > 1) 1116 Dprintf("sbdsp_dma_input: cc=%d 0x%x (0x%x)\n", cc, intr, arg); 1117 #endif 1118 if (sc->sc_channels == 2 && (cc & 1)) { 1119 DPRINTF(("sbdsp_dma_input: stereo input, odd bytecnt\n")); 1120 return EIO; 1121 } 1122 1123 iobase = sc->sc_iobase; 1124 if (sc->sc_dmadir != SB_DMA_IN) { 1125 if (ISSBPRO(sc)) { 1126 if (sc->sc_channels == 2) { 1127 if (ISJAZZ16(sc) && sc->sc_precision == 16) { 1128 if (sbdsp_wdsp(iobase, 1129 JAZZ16_RECORD_STEREO) < 0) { 1130 goto badmode; 1131 } 1132 } else if (sbdsp_wdsp(iobase, 1133 SB_DSP_RECORD_STEREO) < 0) 1134 goto badmode; 1135 sbdsp_mix_write(sc, SBP_INFILTER, 1136 (sbdsp_mix_read(sc, SBP_INFILTER) & 1137 ~SBP_IFILTER_MASK) | SBP_FILTER_OFF); 1138 } else { 1139 if (ISJAZZ16(sc) && sc->sc_precision == 16) { 1140 if (sbdsp_wdsp(iobase, 1141 JAZZ16_RECORD_MONO) < 0) 1142 { 1143 goto badmode; 1144 } 1145 } else if (sbdsp_wdsp(iobase, SB_DSP_RECORD_MONO) < 0) 1146 goto badmode; 1147 sbdsp_mix_write(sc, SBP_INFILTER, 1148 (sbdsp_mix_read(sc, SBP_INFILTER) & 1149 ~SBP_IFILTER_MASK) | sc->in_filter); 1150 } 1151 } 1152 1153 if (ISSB16CLASS(sc)) { 1154 if (sbdsp_wdsp(iobase, SB_DSP16_INPUTRATE) < 0 || 1155 sbdsp_wdsp(iobase, sc->sc_irate >> 8) < 0 || 1156 sbdsp_wdsp(iobase, sc->sc_irate) < 0) 1157 goto giveup; 1158 } else 1159 sbdsp_set_timeconst(sc, sc->sc_itc); 1160 sc->sc_dmadir = SB_DMA_IN; 1161 } 1162 1163 isa_dmastart(DMAMODE_READ, p, cc, sc->sc_drq); 1164 sc->sc_intr = intr; 1165 sc->sc_arg = arg; 1166 sc->dmaflags = DMAMODE_READ; 1167 sc->dmaaddr = p; 1168 sc->dmacnt = cc; /* DMA controller is strange...? */ 1169 1170 if ((ISSB16CLASS(sc) && sc->sc_precision == 16) || 1171 (ISJAZZ16(sc) && sc->sc_drq > 3)) 1172 cc >>= 1; 1173 --cc; 1174 if (ISSB16CLASS(sc)) { 1175 if (sbdsp_wdsp(iobase, sc->sc_precision == 16 ? SB_DSP16_RDMA_16 : 1176 SB_DSP16_RDMA_8) < 0 || 1177 sbdsp_wdsp(iobase, (sc->sc_precision == 16 ? 0x10 : 0x00) | 1178 (sc->sc_channels == 2 ? 0x20 : 0x00)) < 0 || 1179 sbdsp16_wait(iobase) || 1180 sbdsp_wdsp(iobase, cc) < 0 || 1181 sbdsp_wdsp(iobase, cc >> 8) < 0) { 1182 DPRINTF(("sbdsp_dma_input: SB16 DMA start failed\n")); 1183 goto giveup; 1184 } 1185 } else if (sc->sc_imode == SB_ADAC_LS) { 1186 if (sbdsp_wdsp(iobase, SB_DSP_RDMA) < 0 || 1187 sbdsp_wdsp(iobase, cc) < 0 || 1188 sbdsp_wdsp(iobase, cc >> 8) < 0) { 1189 DPRINTF(("sbdsp_dma_input: LS DMA start failed\n")); 1190 goto giveup; 1191 } 1192 } else { 1193 if (cc != sc->sc_last_hs_size) { 1194 if (sbdsp_wdsp(iobase, SB_DSP_BLOCKSIZE) < 0 || 1195 sbdsp_wdsp(iobase, cc) < 0 || 1196 sbdsp_wdsp(iobase, cc >> 8) < 0) { 1197 DPRINTF(("sbdsp_dma_input: HS DMA start failed\n")); 1198 goto giveup; 1199 } 1200 sc->sc_last_hs_size = cc; 1201 } 1202 if (sbdsp_wdsp(iobase, SB_DSP_HS_INPUT) < 0) { 1203 DPRINTF(("sbdsp_dma_input: HS DMA restart failed\n")); 1204 goto giveup; 1205 } 1206 } 1207 return 0; 1208 1209 giveup: 1210 sbdsp_reset(sc); 1211 return EIO; 1212 1213 badmode: 1214 DPRINTF(("sbdsp_dma_input: can't set %s mode\n", 1215 sc->sc_channels == 2 ? "stereo" : "mono")); 1216 return EIO; 1217 } 1218 1219 int 1220 sbdsp_dma_output(addr, p, cc, intr, arg) 1221 void *addr; 1222 void *p; 1223 int cc; 1224 void (*intr) __P((void *)); 1225 void *arg; 1226 { 1227 register struct sbdsp_softc *sc = addr; 1228 register int iobase; 1229 1230 #ifdef AUDIO_DEBUG 1231 if (sbdspdebug > 1) 1232 Dprintf("sbdsp_dma_output: cc=%d 0x%x (0x%x)\n", cc, intr, arg); 1233 #endif 1234 if (sc->sc_channels == 2 && (cc & 1)) { 1235 DPRINTF(("stereo playback odd bytes (%d)\n", cc)); 1236 return EIO; 1237 } 1238 1239 iobase = sc->sc_iobase; 1240 if (sc->sc_dmadir != SB_DMA_OUT) { 1241 if (ISSBPRO(sc)) { 1242 /* make sure we re-set stereo mixer bit when we start 1243 output. */ 1244 sbdsp_mix_write(sc, SBP_STEREO, 1245 (sbdsp_mix_read(sc, SBP_STEREO) & ~SBP_PLAYMODE_MASK) | 1246 (sc->sc_channels == 2 ? SBP_PLAYMODE_STEREO : SBP_PLAYMODE_MONO)); 1247 if (ISJAZZ16(sc)) { 1248 /* Yes, we write the record mode to set 1249 16-bit playback mode. weird, huh? */ 1250 if (sc->sc_precision == 16) { 1251 sbdsp_wdsp(iobase, 1252 sc->sc_channels == 2 ? 1253 JAZZ16_RECORD_STEREO : 1254 JAZZ16_RECORD_MONO); 1255 } else { 1256 sbdsp_wdsp(iobase, 1257 sc->sc_channels == 2 ? 1258 SB_DSP_RECORD_STEREO : 1259 SB_DSP_RECORD_MONO); 1260 } 1261 } 1262 } 1263 1264 if (ISSB16CLASS(sc)) { 1265 if (sbdsp_wdsp(iobase, SB_DSP16_OUTPUTRATE) < 0 || 1266 sbdsp_wdsp(iobase, sc->sc_orate >> 8) < 0 || 1267 sbdsp_wdsp(iobase, sc->sc_orate) < 0) 1268 goto giveup; 1269 } else 1270 sbdsp_set_timeconst(sc, sc->sc_otc); 1271 sc->sc_dmadir = SB_DMA_OUT; 1272 } 1273 1274 isa_dmastart(DMAMODE_WRITE, p, cc, sc->sc_drq); 1275 sc->sc_intr = intr; 1276 sc->sc_arg = arg; 1277 sc->dmaflags = DMAMODE_WRITE; 1278 sc->dmaaddr = p; 1279 sc->dmacnt = cc; /* a vagary of how DMA works, apparently. */ 1280 1281 if ((ISSB16CLASS(sc) && sc->sc_precision == 16) || 1282 (ISJAZZ16(sc) && sc->sc_drq > 3)) 1283 cc >>= 1; 1284 --cc; 1285 if (ISSB16CLASS(sc)) { 1286 if (sbdsp_wdsp(iobase, sc->sc_precision == 16 ? SB_DSP16_WDMA_16 : 1287 SB_DSP16_WDMA_8) < 0 || 1288 sbdsp_wdsp(iobase, (sc->sc_precision == 16 ? 0x10 : 0x00) | 1289 (sc->sc_channels == 2 ? 0x20 : 0x00)) < 0 || 1290 sbdsp16_wait(iobase) || 1291 sbdsp_wdsp(iobase, cc) < 0 || 1292 sbdsp_wdsp(iobase, cc >> 8) < 0) { 1293 DPRINTF(("sbdsp_dma_output: SB16 DMA start failed\n")); 1294 goto giveup; 1295 } 1296 } else if (sc->sc_omode == SB_ADAC_LS) { 1297 if (sbdsp_wdsp(iobase, SB_DSP_WDMA) < 0 || 1298 sbdsp_wdsp(iobase, cc) < 0 || 1299 sbdsp_wdsp(iobase, cc >> 8) < 0) { 1300 DPRINTF(("sbdsp_dma_output: LS DMA start failed\n")); 1301 goto giveup; 1302 } 1303 } else { 1304 if (cc != sc->sc_last_hs_size) { 1305 if (sbdsp_wdsp(iobase, SB_DSP_BLOCKSIZE) < 0 || 1306 sbdsp_wdsp(iobase, cc) < 0 || 1307 sbdsp_wdsp(iobase, cc >> 8) < 0) { 1308 DPRINTF(("sbdsp_dma_output: HS DMA start failed\n")); 1309 goto giveup; 1310 } 1311 sc->sc_last_hs_size = cc; 1312 } 1313 if (sbdsp_wdsp(iobase, SB_DSP_HS_OUTPUT) < 0) { 1314 DPRINTF(("sbdsp_dma_output: HS DMA restart failed\n")); 1315 goto giveup; 1316 } 1317 } 1318 return 0; 1319 1320 giveup: 1321 sbdsp_reset(sc); 1322 return EIO; 1323 } 1324 1325 /* 1326 * Only the DSP unit on the sound blaster generates interrupts. 1327 * There are three cases of interrupt: reception of a midi byte 1328 * (when mode is enabled), completion of dma transmission, or 1329 * completion of a dma reception. The three modes are mutually 1330 * exclusive so we know a priori which event has occurred. 1331 */ 1332 int 1333 sbdsp_intr(arg) 1334 void *arg; 1335 { 1336 register struct sbdsp_softc *sc = arg; 1337 u_char x; 1338 1339 #ifdef AUDIO_DEBUG 1340 if (sbdspdebug > 1) 1341 Dprintf("sbdsp_intr: intr=0x%x\n", sc->sc_intr); 1342 #endif 1343 if (!isa_dmafinished(sc->sc_drq)) { 1344 printf("sbdsp_intr: not finished\n"); 1345 return 0; 1346 } 1347 sc->sc_interrupts++; 1348 /* clear interrupt */ 1349 #ifdef notyet 1350 x = sbdsp_mix_read(sc, 0x82); 1351 x = inb(sc->sc_iobase + 15); 1352 #endif 1353 x = inb(sc->sc_iobase + SBP_DSP_RSTAT); 1354 delay(10); 1355 #if 0 1356 if (sc->sc_mintr != 0) { 1357 x = sbdsp_rdsp(sc->sc_iobase); 1358 (*sc->sc_mintr)(sc->sc_arg, x); 1359 } else 1360 #endif 1361 if (sc->sc_intr != 0) { 1362 isa_dmadone(sc->dmaflags, sc->dmaaddr, sc->dmacnt, sc->sc_drq); 1363 (*sc->sc_intr)(sc->sc_arg); 1364 } 1365 else 1366 return 0; 1367 return 1; 1368 } 1369 1370 #if 0 1371 /* 1372 * Enter midi uart mode and arrange for read interrupts 1373 * to vector to `intr'. This puts the card in a mode 1374 * which allows only midi I/O; the card must be reset 1375 * to leave this mode. Unfortunately, the card does not 1376 * use transmit interrupts, so bytes must be output 1377 * using polling. To keep the polling overhead to a 1378 * minimum, output should be driven off a timer. 1379 * This is a little tricky since only 320us separate 1380 * consecutive midi bytes. 1381 */ 1382 void 1383 sbdsp_set_midi_mode(sc, intr, arg) 1384 struct sbdsp_softc *sc; 1385 void (*intr)(); 1386 void *arg; 1387 { 1388 1389 sbdsp_wdsp(sc->sc_iobase, SB_MIDI_UART_INTR); 1390 sc->sc_mintr = intr; 1391 sc->sc_intr = 0; 1392 sc->sc_arg = arg; 1393 } 1394 1395 /* 1396 * Write a byte to the midi port, when in midi uart mode. 1397 */ 1398 void 1399 sbdsp_midi_output(sc, v) 1400 struct sbdsp_softc *sc; 1401 int v; 1402 { 1403 1404 if (sbdsp_wdsp(sc->sc_iobase, v) < 0) 1405 ++sberr.wmidi; 1406 } 1407 #endif 1408 1409 u_int 1410 sbdsp_get_silence(encoding) 1411 int encoding; 1412 { 1413 #define ULAW_SILENCE 0x7f 1414 #define LINEAR_SILENCE 0 1415 u_int auzero; 1416 1417 switch (encoding) { 1418 case AUDIO_ENCODING_ULAW: 1419 auzero = ULAW_SILENCE; 1420 break; 1421 case AUDIO_ENCODING_PCM16: 1422 default: 1423 auzero = LINEAR_SILENCE; 1424 break; 1425 } 1426 1427 return (auzero); 1428 } 1429 1430 int 1431 sbdsp_setfd(addr, flag) 1432 void *addr; 1433 int flag; 1434 { 1435 /* Can't do full-duplex */ 1436 return(ENOTTY); 1437 } 1438 1439 int 1440 sbdsp_mixer_set_port(addr, cp) 1441 void *addr; 1442 mixer_ctrl_t *cp; 1443 { 1444 register struct sbdsp_softc *sc = addr; 1445 int src, gain; 1446 1447 DPRINTF(("sbdsp_mixer_set_port: port=%d num_channels=%d\n", cp->dev, 1448 cp->un.value.num_channels)); 1449 1450 if (!ISSBPROCLASS(sc)) 1451 return EINVAL; 1452 1453 /* 1454 * Everything is a value except for SBPro BASS/TREBLE and 1455 * RECORD_SOURCE 1456 */ 1457 switch (cp->dev) { 1458 case SB_SPEAKER: 1459 cp->dev = SB_MASTER_VOL; 1460 case SB_MIC_PORT: 1461 case SB_LINE_IN_PORT: 1462 case SB_DAC_PORT: 1463 case SB_FM_PORT: 1464 case SB_CD_PORT: 1465 case SB_MASTER_VOL: 1466 if (cp->type != AUDIO_MIXER_VALUE) 1467 return EINVAL; 1468 1469 /* 1470 * All the mixer ports are stereo except for the microphone. 1471 * If we get a single-channel gain value passed in, then we 1472 * duplicate it to both left and right channels. 1473 */ 1474 1475 switch (cp->dev) { 1476 case SB_MIC_PORT: 1477 if (cp->un.value.num_channels != 1) 1478 return EINVAL; 1479 1480 /* handle funny microphone gain */ 1481 gain = SBP_AGAIN_TO_MICGAIN(cp->un.value.level[AUDIO_MIXER_LEVEL_MONO]); 1482 break; 1483 case SB_LINE_IN_PORT: 1484 case SB_DAC_PORT: 1485 case SB_FM_PORT: 1486 case SB_CD_PORT: 1487 case SB_MASTER_VOL: 1488 switch (cp->un.value.num_channels) { 1489 case 1: 1490 gain = sbdsp_mono_vol(SBP_AGAIN_TO_SBGAIN(cp->un.value.level[AUDIO_MIXER_LEVEL_MONO])); 1491 break; 1492 case 2: 1493 gain = sbdsp_stereo_vol(SBP_AGAIN_TO_SBGAIN(cp->un.value.level[AUDIO_MIXER_LEVEL_LEFT]), 1494 SBP_AGAIN_TO_SBGAIN(cp->un.value.level[AUDIO_MIXER_LEVEL_RIGHT])); 1495 break; 1496 default: 1497 return EINVAL; 1498 } 1499 break; 1500 default: 1501 return EINVAL; 1502 } 1503 1504 switch (cp->dev) { 1505 case SB_MIC_PORT: 1506 src = SBP_MIC_VOL; 1507 break; 1508 case SB_MASTER_VOL: 1509 src = SBP_MASTER_VOL; 1510 break; 1511 case SB_LINE_IN_PORT: 1512 src = SBP_LINE_VOL; 1513 break; 1514 case SB_DAC_PORT: 1515 src = SBP_DAC_VOL; 1516 break; 1517 case SB_FM_PORT: 1518 src = SBP_FM_VOL; 1519 break; 1520 case SB_CD_PORT: 1521 src = SBP_CD_VOL; 1522 break; 1523 default: 1524 return EINVAL; 1525 } 1526 1527 sbdsp_mix_write(sc, src, gain); 1528 sc->gain[cp->dev] = gain; 1529 break; 1530 1531 case SB_TREBLE: 1532 case SB_BASS: 1533 case SB_RECORD_SOURCE: 1534 if (cp->type != AUDIO_MIXER_ENUM) 1535 return EINVAL; 1536 1537 switch (cp->dev) { 1538 case SB_TREBLE: 1539 return sbdsp_set_ifilter(addr, cp->un.ord ? SBP_TREBLE_EQ : 0); 1540 case SB_BASS: 1541 return sbdsp_set_ifilter(addr, cp->un.ord ? SBP_BASS_EQ : 0); 1542 case SB_RECORD_SOURCE: 1543 return sbdsp_set_in_port(addr, cp->un.ord); 1544 } 1545 1546 break; 1547 1548 default: 1549 return EINVAL; 1550 } 1551 1552 return (0); 1553 } 1554 1555 int 1556 sbdsp_mixer_get_port(addr, cp) 1557 void *addr; 1558 mixer_ctrl_t *cp; 1559 { 1560 register struct sbdsp_softc *sc = addr; 1561 int gain; 1562 1563 DPRINTF(("sbdsp_mixer_get_port: port=%d", cp->dev)); 1564 1565 if (!ISSBPROCLASS(sc)) 1566 return EINVAL; 1567 1568 switch (cp->dev) { 1569 case SB_SPEAKER: 1570 cp->dev = SB_MASTER_VOL; 1571 case SB_MIC_PORT: 1572 case SB_LINE_IN_PORT: 1573 case SB_DAC_PORT: 1574 case SB_FM_PORT: 1575 case SB_CD_PORT: 1576 case SB_MASTER_VOL: 1577 gain = sc->gain[cp->dev]; 1578 1579 switch (cp->dev) { 1580 case SB_MIC_PORT: 1581 if (cp->un.value.num_channels != 1) 1582 return EINVAL; 1583 1584 cp->un.value.level[AUDIO_MIXER_LEVEL_MONO] = SBP_MICGAIN_TO_AGAIN(gain); 1585 break; 1586 case SB_LINE_IN_PORT: 1587 case SB_DAC_PORT: 1588 case SB_FM_PORT: 1589 case SB_CD_PORT: 1590 case SB_MASTER_VOL: 1591 switch (cp->un.value.num_channels) { 1592 case 1: 1593 cp->un.value.level[AUDIO_MIXER_LEVEL_MONO] = SBP_SBGAIN_TO_AGAIN(gain); 1594 break; 1595 case 2: 1596 cp->un.value.level[AUDIO_MIXER_LEVEL_LEFT] = SBP_LEFTGAIN(gain); 1597 cp->un.value.level[AUDIO_MIXER_LEVEL_RIGHT] = SBP_RIGHTGAIN(gain); 1598 break; 1599 default: 1600 return EINVAL; 1601 } 1602 break; 1603 } 1604 1605 break; 1606 1607 case SB_TREBLE: 1608 case SB_BASS: 1609 case SB_RECORD_SOURCE: 1610 switch (cp->dev) { 1611 case SB_TREBLE: 1612 cp->un.ord = sbdsp_get_ifilter(addr) == SBP_TREBLE_EQ; 1613 return 0; 1614 case SB_BASS: 1615 cp->un.ord = sbdsp_get_ifilter(addr) == SBP_BASS_EQ; 1616 return 0; 1617 case SB_RECORD_SOURCE: 1618 cp->un.ord = sbdsp_get_in_port(addr); 1619 return 0; 1620 } 1621 1622 break; 1623 1624 default: 1625 return EINVAL; 1626 } 1627 1628 return (0); 1629 } 1630 1631 int 1632 sbdsp_mixer_query_devinfo(addr, dip) 1633 void *addr; 1634 register mixer_devinfo_t *dip; 1635 { 1636 register struct sbdsp_softc *sc = addr; 1637 1638 DPRINTF(("sbdsp_mixer_query_devinfo: index=%d\n", dip->index)); 1639 1640 switch (dip->index) { 1641 case SB_MIC_PORT: 1642 dip->type = AUDIO_MIXER_VALUE; 1643 dip->mixer_class = SB_INPUT_CLASS; 1644 dip->prev = AUDIO_MIXER_LAST; 1645 dip->next = AUDIO_MIXER_LAST; 1646 strcpy(dip->label.name, AudioNmicrophone); 1647 dip->un.v.num_channels = 1; 1648 strcpy(dip->un.v.units.name, AudioNvolume); 1649 return 0; 1650 1651 case SB_SPEAKER: 1652 dip->type = AUDIO_MIXER_VALUE; 1653 dip->mixer_class = SB_OUTPUT_CLASS; 1654 dip->prev = AUDIO_MIXER_LAST; 1655 dip->next = AUDIO_MIXER_LAST; 1656 strcpy(dip->label.name, AudioNspeaker); 1657 dip->un.v.num_channels = 1; 1658 strcpy(dip->un.v.units.name, AudioNvolume); 1659 return 0; 1660 1661 case SB_INPUT_CLASS: 1662 dip->type = AUDIO_MIXER_CLASS; 1663 dip->mixer_class = SB_INPUT_CLASS; 1664 dip->next = dip->prev = AUDIO_MIXER_LAST; 1665 strcpy(dip->label.name, AudioCInputs); 1666 return 0; 1667 1668 case SB_OUTPUT_CLASS: 1669 dip->type = AUDIO_MIXER_CLASS; 1670 dip->mixer_class = SB_OUTPUT_CLASS; 1671 dip->next = dip->prev = AUDIO_MIXER_LAST; 1672 strcpy(dip->label.name, AudioCOutputs); 1673 return 0; 1674 } 1675 1676 if (ISSBPROCLASS(sc)) { 1677 switch (dip->index) { 1678 case SB_LINE_IN_PORT: 1679 dip->type = AUDIO_MIXER_VALUE; 1680 dip->mixer_class = SB_INPUT_CLASS; 1681 dip->prev = AUDIO_MIXER_LAST; 1682 dip->next = AUDIO_MIXER_LAST; 1683 strcpy(dip->label.name, AudioNline); 1684 dip->un.v.num_channels = 2; 1685 strcpy(dip->un.v.units.name, AudioNvolume); 1686 return 0; 1687 1688 case SB_DAC_PORT: 1689 dip->type = AUDIO_MIXER_VALUE; 1690 dip->mixer_class = SB_INPUT_CLASS; 1691 dip->prev = AUDIO_MIXER_LAST; 1692 dip->next = AUDIO_MIXER_LAST; 1693 strcpy(dip->label.name, AudioNdac); 1694 dip->un.v.num_channels = 2; 1695 strcpy(dip->un.v.units.name, AudioNvolume); 1696 return 0; 1697 1698 case SB_CD_PORT: 1699 dip->type = AUDIO_MIXER_VALUE; 1700 dip->mixer_class = SB_INPUT_CLASS; 1701 dip->prev = AUDIO_MIXER_LAST; 1702 dip->next = AUDIO_MIXER_LAST; 1703 strcpy(dip->label.name, AudioNcd); 1704 dip->un.v.num_channels = 2; 1705 strcpy(dip->un.v.units.name, AudioNvolume); 1706 return 0; 1707 1708 case SB_FM_PORT: 1709 dip->type = AUDIO_MIXER_VALUE; 1710 dip->mixer_class = SB_INPUT_CLASS; 1711 dip->prev = AUDIO_MIXER_LAST; 1712 dip->next = AUDIO_MIXER_LAST; 1713 strcpy(dip->label.name, AudioNfmsynth); 1714 dip->un.v.num_channels = 2; 1715 strcpy(dip->un.v.units.name, AudioNvolume); 1716 return 0; 1717 1718 case SB_MASTER_VOL: 1719 dip->type = AUDIO_MIXER_VALUE; 1720 dip->mixer_class = SB_OUTPUT_CLASS; 1721 dip->prev = AUDIO_MIXER_LAST; 1722 dip->next = AUDIO_MIXER_LAST; 1723 strcpy(dip->label.name, AudioNvolume); 1724 dip->un.v.num_channels = 2; 1725 strcpy(dip->un.v.units.name, AudioNvolume); 1726 return 0; 1727 1728 case SB_RECORD_SOURCE: 1729 dip->mixer_class = SB_RECORD_CLASS; 1730 dip->type = AUDIO_MIXER_ENUM; 1731 dip->prev = AUDIO_MIXER_LAST; 1732 dip->next = AUDIO_MIXER_LAST; 1733 strcpy(dip->label.name, AudioNsource); 1734 dip->un.e.num_mem = 3; 1735 strcpy(dip->un.e.member[0].label.name, AudioNmicrophone); 1736 dip->un.e.member[0].ord = SB_MIC_PORT; 1737 strcpy(dip->un.e.member[1].label.name, AudioNcd); 1738 dip->un.e.member[1].ord = SB_CD_PORT; 1739 strcpy(dip->un.e.member[2].label.name, AudioNline); 1740 dip->un.e.member[2].ord = SB_LINE_IN_PORT; 1741 return 0; 1742 1743 case SB_BASS: 1744 dip->type = AUDIO_MIXER_ENUM; 1745 dip->mixer_class = SB_INPUT_CLASS; 1746 dip->prev = AUDIO_MIXER_LAST; 1747 dip->next = AUDIO_MIXER_LAST; 1748 strcpy(dip->label.name, AudioNbass); 1749 dip->un.e.num_mem = 2; 1750 strcpy(dip->un.e.member[0].label.name, AudioNoff); 1751 dip->un.e.member[0].ord = 0; 1752 strcpy(dip->un.e.member[1].label.name, AudioNon); 1753 dip->un.e.member[1].ord = 1; 1754 return 0; 1755 1756 case SB_TREBLE: 1757 dip->type = AUDIO_MIXER_ENUM; 1758 dip->mixer_class = SB_INPUT_CLASS; 1759 dip->prev = AUDIO_MIXER_LAST; 1760 dip->next = AUDIO_MIXER_LAST; 1761 strcpy(dip->label.name, AudioNtreble); 1762 dip->un.e.num_mem = 2; 1763 strcpy(dip->un.e.member[0].label.name, AudioNoff); 1764 dip->un.e.member[0].ord = 0; 1765 strcpy(dip->un.e.member[1].label.name, AudioNon); 1766 dip->un.e.member[1].ord = 1; 1767 return 0; 1768 1769 case SB_RECORD_CLASS: /* record source class */ 1770 dip->type = AUDIO_MIXER_CLASS; 1771 dip->mixer_class = SB_RECORD_CLASS; 1772 dip->next = dip->prev = AUDIO_MIXER_LAST; 1773 strcpy(dip->label.name, AudioCRecord); 1774 return 0; 1775 } 1776 } 1777 1778 return ENXIO; 1779 } 1780
Indexes created Sun Sep 28 12:09:53 GMT 2025