/* $OpenBSD: sbdsp.c,v 1.8 1996/11/02 01:09:37 millert Exp $ */ /* $NetBSD: sbdsp.c,v 1.30 1996/10/25 07:25:48 fvdl Exp $ */ /* * Copyright (c) 1991-1993 Regents of the University of California. * All rights reserved. * * Redistribution and use in source and binary forms, with or without * modification, are permitted provided that the following conditions * are met: * 1. Redistributions of source code must retain the above copyright * notice, this list of conditions and the following disclaimer. * 2. Redistributions in binary form must reproduce the above copyright * notice, this list of conditions and the following disclaimer in the * documentation and/or other materials provided with the distribution. * 3. All advertising materials mentioning features or use of this software * must display the following acknowledgement: * This product includes software developed by the Computer Systems * Engineering Group at Lawrence Berkeley Laboratory. * 4. Neither the name of the University nor of the Laboratory may be used * to endorse or promote products derived from this software without * specific prior written permission. * * THIS SOFTWARE IS PROVIDED BY THE REGENTS AND CONTRIBUTORS ``AS IS'' AND * ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE * IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE * ARE DISCLAIMED. IN NO EVENT SHALL THE REGENTS OR CONTRIBUTORS BE LIABLE * FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL * DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS * OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) * HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT * LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY * OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF * SUCH DAMAGE. * */ /* * SoundBlaster Pro code provided by John Kohl, based on lots of * information he gleaned from Steve Haehnichen 's * SBlast driver for 386BSD and DOS driver code from Daniel Sachs * . */ #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include /* XXX BROKEN; WHY? */ #include #include #ifdef AUDIO_DEBUG extern void Dprintf __P((const char *, ...)); #define DPRINTF(x) if (sbdspdebug) Dprintf x int sbdspdebug = 0; #else #define DPRINTF(x) #endif #ifndef SBDSP_NPOLL #define SBDSP_NPOLL 3000 #endif struct { int wdsp; int rdsp; int wmidi; } sberr; int sbdsp_srtotc __P((struct sbdsp_softc *sc, int sr, int isdac, int *tcp, int *modep)); u_int sbdsp_jazz16_probe __P((struct sbdsp_softc *)); /* * Time constant routines follow. See SBK, section 12. * Although they don't come out and say it (in the docs), * the card clearly uses a 1MHz countdown timer, as the * low-speed formula (p. 12-4) is: * tc = 256 - 10^6 / sr * In high-speed mode, the constant is the upper byte of a 16-bit counter, * and a 256MHz clock is used: * tc = 65536 - 256 * 10^ 6 / sr * Since we can only use the upper byte of the HS TC, the two formulae * are equivalent. (Why didn't they say so?) E.g., * (65536 - 256 * 10 ^ 6 / x) >> 8 = 256 - 10^6 / x * * The crossover point (from low- to high-speed modes) is different * for the SBPRO and SB20. The table on p. 12-5 gives the following data: * * SBPRO SB20 * ----- -------- * input ls min 4 KHz 4 KHz * input ls max 23 KHz 13 KHz * input hs max 44.1 KHz 15 KHz * output ls min 4 KHz 4 KHz * output ls max 23 KHz 23 KHz * output hs max 44.1 KHz 44.1 KHz */ #define SB_LS_MIN 0x06 /* 4000 Hz */ #define SB_8K 0x83 /* 8000 Hz */ #define SBPRO_ADC_LS_MAX 0xd4 /* 22727 Hz */ #define SBPRO_ADC_HS_MAX 0xea /* 45454 Hz */ #define SBCLA_ADC_LS_MAX 0xb3 /* 12987 Hz */ #define SBCLA_ADC_HS_MAX 0xbd /* 14925 Hz */ #define SB_DAC_LS_MAX 0xd4 /* 22727 Hz */ #define SB_DAC_HS_MAX 0xea /* 45454 Hz */ int sbdsp16_wait __P((int)); void sbdsp_to __P((void *)); void sbdsp_pause __P((struct sbdsp_softc *)); int sbdsp_setrate __P((struct sbdsp_softc *, int, int, int *)); int sbdsp_tctosr __P((struct sbdsp_softc *, int)); int sbdsp_set_timeconst __P((struct sbdsp_softc *, int)); #ifdef AUDIO_DEBUG void sb_printsc __P((struct sbdsp_softc *)); #endif #ifdef AUDIO_DEBUG void sb_printsc(sc) struct sbdsp_softc *sc; { int i; printf("open %d dmachan %d iobase %x\n", sc->sc_open, sc->sc_drq, sc->sc_iobase); printf("irate %d itc %d imode %d orate %d otc %d omode %d encoding %x\n", sc->sc_irate, sc->sc_itc, sc->sc_imode, sc->sc_orate, sc->sc_otc, sc->sc_omode, sc->encoding); printf("outport %d inport %d spkron %d nintr %lu\n", sc->out_port, sc->in_port, sc->spkr_state, sc->sc_interrupts); printf("precision %d channels %d intr %p arg %p\n", sc->sc_precision, sc->sc_channels, sc->sc_intr, sc->sc_arg); printf("gain: "); for (i = 0; i < SB_NDEVS; i++) printf("%d ", sc->gain[i]); printf("\n"); } #endif /* * Probe / attach routines. */ /* * Probe for the soundblaster hardware. */ int sbdsp_probe(sc) struct sbdsp_softc *sc; { if (sbdsp_reset(sc) < 0) { DPRINTF(("sbdsp: couldn't reset card\n")); return 0; } /* if flags set, go and probe the jazz16 stuff */ if (sc->sc_dev.dv_cfdata->cf_flags != 0) sc->sc_model = sbdsp_jazz16_probe(sc); else sc->sc_model = sbversion(sc); return 1; } /* * Try add-on stuff for Jazz16. */ u_int sbdsp_jazz16_probe(sc) struct sbdsp_softc *sc; { static u_char jazz16_irq_conf[16] = { -1, -1, 0x02, 0x03, -1, 0x01, -1, 0x04, -1, 0x02, 0x05, -1, -1, -1, -1, 0x06}; static u_char jazz16_drq_conf[8] = { -1, 0x01, -1, 0x02, -1, 0x03, -1, 0x04}; u_int rval = sbversion(sc); register int iobase = sc->sc_iobase; if (jazz16_drq_conf[sc->sc_drq] == (u_char)-1 || jazz16_irq_conf[sc->sc_irq] == (u_char)-1) return rval; /* give up, we can't do it. */ outb(JAZZ16_CONFIG_PORT, JAZZ16_WAKEUP); delay(10000); /* delay 10 ms */ outb(JAZZ16_CONFIG_PORT, JAZZ16_SETBASE); outb(JAZZ16_CONFIG_PORT, iobase & 0x70); if (sbdsp_reset(sc) < 0) return rval; /* XXX? what else could we do? */ if (sbdsp_wdsp(iobase, JAZZ16_READ_VER)) return rval; if (sbdsp_rdsp(iobase) != JAZZ16_VER_JAZZ) return rval; if (sbdsp_wdsp(iobase, JAZZ16_SET_DMAINTR) || /* set both 8 & 16-bit drq to same channel, it works fine. */ sbdsp_wdsp(iobase, (jazz16_drq_conf[sc->sc_drq] << 4) | jazz16_drq_conf[sc->sc_drq]) || sbdsp_wdsp(iobase, jazz16_irq_conf[sc->sc_irq])) { DPRINTF(("sbdsp: can't write jazz16 probe stuff")); return rval; } return (rval | MODEL_JAZZ16); } /* * Attach hardware to driver, attach hardware driver to audio * pseudo-device driver . */ void sbdsp_attach(sc) struct sbdsp_softc *sc; { /* Set defaults */ if (ISSB16CLASS(sc)) sc->sc_irate = sc->sc_orate = 8000; else if (ISSBPROCLASS(sc)) sc->sc_itc = sc->sc_otc = SB_8K; else sc->sc_itc = sc->sc_otc = SB_8K; sc->sc_precision = 8; sc->sc_channels = 1; sc->encoding = AUDIO_ENCODING_ULAW; (void) sbdsp_set_in_port(sc, SB_MIC_PORT); (void) sbdsp_set_out_port(sc, SB_SPEAKER); if (ISSBPROCLASS(sc)) { int i; /* set mixer to default levels, by sending a mixer reset command. */ sbdsp_mix_write(sc, SBP_MIX_RESET, SBP_MIX_RESET); /* then some adjustments :) */ sbdsp_mix_write(sc, SBP_CD_VOL, sbdsp_stereo_vol(SBP_MAXVOL, SBP_MAXVOL)); sbdsp_mix_write(sc, SBP_DAC_VOL, sbdsp_stereo_vol(SBP_MAXVOL, SBP_MAXVOL)); sbdsp_mix_write(sc, SBP_MASTER_VOL, sbdsp_stereo_vol(SBP_MAXVOL/2, SBP_MAXVOL/2)); sbdsp_mix_write(sc, SBP_LINE_VOL, sbdsp_stereo_vol(SBP_MAXVOL, SBP_MAXVOL)); for (i = 0; i < SB_NDEVS; i++) sc->gain[i] = sbdsp_stereo_vol(SBP_MAXVOL, SBP_MAXVOL); sc->in_filter = 0; /* no filters turned on, please */ } printf(": dsp v%d.%02d%s\n", SBVER_MAJOR(sc->sc_model), SBVER_MINOR(sc->sc_model), ISJAZZ16(sc) ? ": " : ""); #ifdef notyet sbdsp_mix_write(sc, SBP_SET_IRQ, 0x04); sbdsp_mix_write(sc, SBP_SET_DRQ, 0x22); printf("sbdsp_attach: irq=%02x, drq=%02x\n", sbdsp_mix_read(sc, SBP_SET_IRQ), sbdsp_mix_read(sc, SBP_SET_DRQ)); #else if (ISSB16CLASS(sc)) sc->sc_model = 0x0300; #endif } /* * Various routines to interface to higher level audio driver */ void sbdsp_mix_write(sc, mixerport, val) struct sbdsp_softc *sc; int mixerport; int val; { int iobase = sc->sc_iobase; outb(iobase + SBP_MIXER_ADDR, mixerport); delay(10); outb(iobase + SBP_MIXER_DATA, val); delay(30); } int sbdsp_mix_read(sc, mixerport) struct sbdsp_softc *sc; int mixerport; { int iobase = sc->sc_iobase; outb(iobase + SBP_MIXER_ADDR, mixerport); delay(10); return inb(iobase + SBP_MIXER_DATA); } int sbdsp_set_in_sr(addr, sr) void *addr; u_long sr; { register struct sbdsp_softc *sc = addr; if (ISSB16CLASS(sc)) return (sbdsp_setrate(sc, sr, SB_INPUT_RATE, &sc->sc_irate)); else return (sbdsp_srtotc(sc, sr, SB_INPUT_RATE, &sc->sc_itc, &sc->sc_imode)); } u_long sbdsp_get_in_sr(addr) void *addr; { register struct sbdsp_softc *sc = addr; if (ISSB16CLASS(sc)) return (sc->sc_irate); else return (sbdsp_tctosr(sc, sc->sc_itc)); } int sbdsp_set_out_sr(addr, sr) void *addr; u_long sr; { register struct sbdsp_softc *sc = addr; if (ISSB16CLASS(sc)) return (sbdsp_setrate(sc, sr, SB_OUTPUT_RATE, &sc->sc_orate)); else return (sbdsp_srtotc(sc, sr, SB_OUTPUT_RATE, &sc->sc_otc, &sc->sc_omode)); } u_long sbdsp_get_out_sr(addr) void *addr; { register struct sbdsp_softc *sc = addr; if (ISSB16CLASS(sc)) return (sc->sc_orate); else return (sbdsp_tctosr(sc, sc->sc_otc)); } int sbdsp_query_encoding(addr, fp) void *addr; struct audio_encoding *fp; { switch (fp->index) { case 0: strcpy(fp->name, AudioEmulaw); fp->format_id = AUDIO_ENCODING_ULAW; break; case 1: strcpy(fp->name, AudioEpcm16); fp->format_id = AUDIO_ENCODING_PCM16; break; default: return (EINVAL); } return (0); } int sbdsp_set_encoding(addr, encoding) void *addr; u_int encoding; { register struct sbdsp_softc *sc = addr; switch (encoding) { case AUDIO_ENCODING_ULAW: sc->encoding = AUDIO_ENCODING_ULAW; break; case AUDIO_ENCODING_LINEAR: sc->encoding = AUDIO_ENCODING_LINEAR; break; default: return (EINVAL); } return (0); } int sbdsp_get_encoding(addr) void *addr; { register struct sbdsp_softc *sc = addr; return (sc->encoding); } int sbdsp_set_precision(addr, precision) void *addr; u_int precision; { register struct sbdsp_softc *sc = addr; if (ISSB16CLASS(sc) || ISJAZZ16(sc)) { if (precision != 16 && precision != 8) return (EINVAL); sc->sc_precision = precision; } else { if (precision != 8) return (EINVAL); sc->sc_precision = precision; } return (0); } int sbdsp_get_precision(addr) void *addr; { register struct sbdsp_softc *sc = addr; return (sc->sc_precision); } int sbdsp_set_channels(addr, channels) void *addr; int channels; { register struct sbdsp_softc *sc = addr; if (ISSBPROCLASS(sc)) { if (channels != 1 && channels != 2) return (EINVAL); sc->sc_channels = channels; sc->sc_dmadir = SB_DMA_NONE; /* * XXXX * With 2 channels, SBPro can't do more than 22kHz. * No framework to check this. */ } else { if (channels != 1) return (EINVAL); sc->sc_channels = channels; } return (0); } int sbdsp_get_channels(addr) void *addr; { register struct sbdsp_softc *sc = addr; return (sc->sc_channels); } int sbdsp_set_ifilter(addr, which) void *addr; int which; { register struct sbdsp_softc *sc = addr; int mixval; if (ISSBPROCLASS(sc)) { mixval = sbdsp_mix_read(sc, SBP_INFILTER) & ~SBP_IFILTER_MASK; switch (which) { case 0: mixval |= SBP_FILTER_OFF; break; case SBP_TREBLE_EQ: mixval |= SBP_FILTER_ON | SBP_IFILTER_HIGH; break; case SBP_BASS_EQ: mixval |= SBP_FILTER_ON | SBP_IFILTER_LOW; break; default: return (EINVAL); } sc->in_filter = mixval & SBP_IFILTER_MASK; sbdsp_mix_write(sc, SBP_INFILTER, mixval); return (0); } else return (EINVAL); } int sbdsp_get_ifilter(addr) void *addr; { register struct sbdsp_softc *sc = addr; if (ISSBPROCLASS(sc)) { sc->in_filter = sbdsp_mix_read(sc, SBP_INFILTER) & SBP_IFILTER_MASK; switch (sc->in_filter) { case SBP_FILTER_ON|SBP_IFILTER_HIGH: return (SBP_TREBLE_EQ); case SBP_FILTER_ON|SBP_IFILTER_LOW: return (SBP_BASS_EQ); case SBP_FILTER_OFF: default: return (0); } } else return (0); } int sbdsp_set_out_port(addr, port) void *addr; int port; { register struct sbdsp_softc *sc = addr; sc->out_port = port; /* Just record it */ return (0); } int sbdsp_get_out_port(addr) void *addr; { register struct sbdsp_softc *sc = addr; return (sc->out_port); } int sbdsp_set_in_port(addr, port) void *addr; int port; { register struct sbdsp_softc *sc = addr; int mixport, sbport; if (ISSBPROCLASS(sc)) { switch (port) { case SB_MIC_PORT: sbport = SBP_FROM_MIC; mixport = SBP_MIC_VOL; break; case SB_LINE_IN_PORT: sbport = SBP_FROM_LINE; mixport = SBP_LINE_VOL; break; case SB_CD_PORT: sbport = SBP_FROM_CD; mixport = SBP_CD_VOL; break; case SB_DAC_PORT: case SB_FM_PORT: default: return (EINVAL); } } else { switch (port) { case SB_MIC_PORT: sbport = SBP_FROM_MIC; mixport = SBP_MIC_VOL; break; default: return (EINVAL); } } sc->in_port = port; /* Just record it */ if (ISSBPROCLASS(sc)) { /* record from that port */ sbdsp_mix_write(sc, SBP_RECORD_SOURCE, SBP_RECORD_FROM(sbport, SBP_FILTER_OFF, SBP_IFILTER_HIGH)); /* fetch gain from that port */ sc->gain[port] = sbdsp_mix_read(sc, mixport); } return (0); } int sbdsp_get_in_port(addr) void *addr; { register struct sbdsp_softc *sc = addr; return (sc->in_port); } int sbdsp_speaker_ctl(addr, newstate) void *addr; int newstate; { register struct sbdsp_softc *sc = addr; if ((newstate == SPKR_ON) && (sc->spkr_state == SPKR_OFF)) { sbdsp_spkron(sc); sc->spkr_state = SPKR_ON; } if ((newstate == SPKR_OFF) && (sc->spkr_state == SPKR_ON)) { sbdsp_spkroff(sc); sc->spkr_state = SPKR_OFF; } return(0); } int sbdsp_round_blocksize(addr, blk) void *addr; int blk; { register struct sbdsp_softc *sc = addr; sc->sc_last_hs_size = 0; /* Higher speeds need bigger blocks to avoid popping and silence gaps. */ if (blk < NBPG/4 || blk > NBPG/2) { if (ISSB16CLASS(sc)) { if (sc->sc_orate > 8000 || sc->sc_irate > 8000) blk = NBPG/2; } else { if (sc->sc_otc > SB_8K || sc->sc_itc > SB_8K) blk = NBPG/2; } } /* don't try to DMA too much at once, though. */ if (blk > NBPG) blk = NBPG; if (sc->sc_channels == 2) return (blk & ~1); /* must be even to preserve stereo separation */ else return (blk); /* Anything goes :-) */ } int sbdsp_commit_settings(addr) void *addr; { register struct sbdsp_softc *sc = addr; /* due to potentially unfortunate ordering in the above layers, re-do a few sets which may be important--input gains (adjust the proper channels), number of input channels (hit the record rate and set mode) */ if (ISSBPRO(sc)) { /* * With 2 channels, SBPro can't do more than 22kHz. * Whack the rates down to speed if necessary. * Reset the time constant anyway * because it may have been adjusted with a different number * of channels, which means it might have computed the wrong * mode (low/high speed). */ if (sc->sc_channels == 2 && sbdsp_tctosr(sc, sc->sc_itc) > 22727) { sbdsp_srtotc(sc, 22727, SB_INPUT_RATE, &sc->sc_itc, &sc->sc_imode); } else sbdsp_srtotc(sc, sbdsp_tctosr(sc, sc->sc_itc), SB_INPUT_RATE, &sc->sc_itc, &sc->sc_imode); if (sc->sc_channels == 2 && sbdsp_tctosr(sc, sc->sc_otc) > 22727) { sbdsp_srtotc(sc, 22727, SB_OUTPUT_RATE, &sc->sc_otc, &sc->sc_omode); } else sbdsp_srtotc(sc, sbdsp_tctosr(sc, sc->sc_otc), SB_OUTPUT_RATE, &sc->sc_otc, &sc->sc_omode); } if (ISSB16CLASS(sc) || ISJAZZ16(sc)) { if (sc->encoding == AUDIO_ENCODING_ULAW && sc->sc_precision == 16) { sc->sc_precision = 8; return EINVAL; /* XXX what should we really do? */ } } /* * XXX * Should wait for chip to be idle. */ sc->sc_dmadir = SB_DMA_NONE; return 0; } int sbdsp_open(sc, dev, flags) register struct sbdsp_softc *sc; dev_t dev; int flags; { DPRINTF(("sbdsp_open: sc=0x%x\n", sc)); if (sc->sc_open != 0 || sbdsp_reset(sc) != 0) return ENXIO; sc->sc_open = 1; sc->sc_mintr = 0; if (ISSBPROCLASS(sc) && sbdsp_wdsp(sc->sc_iobase, SB_DSP_RECORD_MONO) < 0) { DPRINTF(("sbdsp_open: can't set mono mode\n")); /* we'll readjust when it's time for DMA. */ } /* * Leave most things as they were; users must change things if * the previous process didn't leave it they way they wanted. * Looked at another way, it's easy to set up a configuration * in one program and leave it for another to inherit. */ DPRINTF(("sbdsp_open: opened\n")); return 0; } void sbdsp_close(addr) void *addr; { struct sbdsp_softc *sc = addr; DPRINTF(("sbdsp_close: sc=0x%x\n", sc)); sc->sc_open = 0; sbdsp_spkroff(sc); sc->spkr_state = SPKR_OFF; sc->sc_mintr = 0; sbdsp_haltdma(sc); DPRINTF(("sbdsp_close: closed\n")); } /* * Lower-level routines */ /* * Reset the card. * Return non-zero if the card isn't detected. */ int sbdsp_reset(sc) register struct sbdsp_softc *sc; { register int iobase = sc->sc_iobase; sc->sc_intr = 0; if (sc->sc_dmadir != SB_DMA_NONE) { isa_dmaabort(sc->sc_drq); sc->sc_dmadir = SB_DMA_NONE; } sc->sc_last_hs_size = 0; /* * See SBK, section 11.3. * We pulse a reset signal into the card. * Gee, what a brilliant hardware design. */ outb(iobase + SBP_DSP_RESET, 1); delay(10); outb(iobase + SBP_DSP_RESET, 0); delay(30); if (sbdsp_rdsp(iobase) != SB_MAGIC) return -1; return 0; } int sbdsp16_wait(iobase) int iobase; { register int i; for (i = SBDSP_NPOLL; --i >= 0; ) { register u_char x; x = inb(iobase + SBP_DSP_WSTAT); delay(10); if ((x & SB_DSP_BUSY) == 0) continue; return 0; } ++sberr.wdsp; return -1; } /* * Write a byte to the dsp. * XXX We are at the mercy of the card as we use a * polling loop and wait until it can take the byte. */ int sbdsp_wdsp(int iobase, int v) { register int i; for (i = SBDSP_NPOLL; --i >= 0; ) { register u_char x; x = inb(iobase + SBP_DSP_WSTAT); delay(10); if ((x & SB_DSP_BUSY) != 0) continue; outb(iobase + SBP_DSP_WRITE, v); delay(10); return 0; } ++sberr.wdsp; return -1; } /* * Read a byte from the DSP, using polling. */ int sbdsp_rdsp(int iobase) { register int i; for (i = SBDSP_NPOLL; --i >= 0; ) { register u_char x; x = inb(iobase + SBP_DSP_RSTAT); delay(10); if ((x & SB_DSP_READY) == 0) continue; x = inb(iobase + SBP_DSP_READ); delay(10); return x; } ++sberr.rdsp; return -1; } /* * Doing certain things (like toggling the speaker) make * the SB hardware go away for a while, so pause a little. */ void sbdsp_to(arg) void *arg; { wakeup(arg); } void sbdsp_pause(sc) struct sbdsp_softc *sc; { extern int hz; timeout(sbdsp_to, sbdsp_to, hz/8); (void)tsleep(sbdsp_to, PWAIT, "sbpause", 0); } /* * Turn on the speaker. The SBK documention says this operation * can take up to 1/10 of a second. Higher level layers should * probably let the task sleep for this amount of time after * calling here. Otherwise, things might not work (because * sbdsp_wdsp() and sbdsp_rdsp() will probably timeout.) * * These engineers had their heads up their ass when * they designed this card. */ void sbdsp_spkron(sc) struct sbdsp_softc *sc; { (void)sbdsp_wdsp(sc->sc_iobase, SB_DSP_SPKR_ON); sbdsp_pause(sc); } /* * Turn off the speaker; see comment above. */ void sbdsp_spkroff(sc) struct sbdsp_softc *sc; { (void)sbdsp_wdsp(sc->sc_iobase, SB_DSP_SPKR_OFF); sbdsp_pause(sc); } /* * Read the version number out of the card. Return major code * in high byte, and minor code in low byte. */ short sbversion(sc) struct sbdsp_softc *sc; { register int iobase = sc->sc_iobase; short v; if (sbdsp_wdsp(iobase, SB_DSP_VERSION) < 0) return 0; v = sbdsp_rdsp(iobase) << 8; v |= sbdsp_rdsp(iobase); return ((v >= 0) ? v : 0); } /* * Halt a DMA in progress. A low-speed transfer can be * resumed with sbdsp_contdma(). */ int sbdsp_haltdma(addr) void *addr; { register struct sbdsp_softc *sc = addr; DPRINTF(("sbdsp_haltdma: sc=0x%x\n", sc)); sbdsp_reset(sc); return 0; } int sbdsp_contdma(addr) void *addr; { register struct sbdsp_softc *sc = addr; DPRINTF(("sbdsp_contdma: sc=0x%x\n", sc)); /* XXX how do we reinitialize the DMA controller state? do we care? */ (void)sbdsp_wdsp(sc->sc_iobase, SB_DSP_CONT); return(0); } int sbdsp_setrate(sc, sr, isdac, ratep) register struct sbdsp_softc *sc; int sr; int isdac; int *ratep; { /* * XXXX * More checks here? */ if (sr < 5000 || sr > 44100) return (EINVAL); *ratep = sr; return (0); } /* * Convert a linear sampling rate into the DAC time constant. * Set *mode to indicate the high/low-speed DMA operation. * Because of limitations of the card, not all rates are possible. * We return the time constant of the closest possible rate. * The sampling rate limits are different for the DAC and ADC, * so isdac indicates output, and !isdac indicates input. */ int sbdsp_srtotc(sc, sr, isdac, tcp, modep) register struct sbdsp_softc *sc; int sr; int isdac; int *tcp, *modep; { int tc, realtc, mode; /* * Don't forget to compute which mode we'll be in based on whether * we need to double the rate for stereo on SBPRO. */ if (sr == 0) { tc = SB_LS_MIN; mode = SB_ADAC_LS; goto out; } tc = 256 - (1000000 / sr); if (sc->sc_channels == 2 && ISSBPRO(sc)) /* compute based on 2x sample rate when needed */ realtc = 256 - ( 500000 / sr); else realtc = tc; if (tc < SB_LS_MIN) { tc = SB_LS_MIN; mode = SB_ADAC_LS; /* NB: 2x minimum speed is still low * speed mode. */ goto out; } else if (isdac) { if (realtc <= SB_DAC_LS_MAX) mode = SB_ADAC_LS; else { mode = SB_ADAC_HS; if (tc > SB_DAC_HS_MAX) tc = SB_DAC_HS_MAX; } } else { int adc_ls_max, adc_hs_max; /* XXX use better rounding--compare distance to nearest tc on both sides of requested speed */ if (ISSBPROCLASS(sc)) { adc_ls_max = SBPRO_ADC_LS_MAX; adc_hs_max = SBPRO_ADC_HS_MAX; } else { adc_ls_max = SBCLA_ADC_LS_MAX; adc_hs_max = SBCLA_ADC_HS_MAX; } if (realtc <= adc_ls_max) mode = SB_ADAC_LS; else { mode = SB_ADAC_HS; if (tc > adc_hs_max) tc = adc_hs_max; } } out: *tcp = tc; *modep = mode; return (0); } /* * Convert a DAC time constant to a sampling rate. * See SBK, section 12. */ int sbdsp_tctosr(sc, tc) register struct sbdsp_softc *sc; int tc; { int adc; if (ISSBPROCLASS(sc)) adc = SBPRO_ADC_HS_MAX; else adc = SBCLA_ADC_HS_MAX; if (tc > adc) tc = adc; return (1000000 / (256 - tc)); } int sbdsp_set_timeconst(sc, tc) register struct sbdsp_softc *sc; int tc; { register int iobase; /* * A SBPro in stereo mode uses time constants at double the * actual rate. */ if (ISSBPRO(sc) && sc->sc_channels == 2) tc = 256 - ((256 - tc) / 2); DPRINTF(("sbdsp_set_timeconst: sc=%p tc=%d\n", sc, tc)); iobase = sc->sc_iobase; if (sbdsp_wdsp(iobase, SB_DSP_TIMECONST) < 0 || sbdsp_wdsp(iobase, tc) < 0) return (EIO); return (0); } int sbdsp_dma_input(addr, p, cc, intr, arg) void *addr; void *p; int cc; void (*intr) __P((void *)); void *arg; { register struct sbdsp_softc *sc = addr; register int iobase; #ifdef AUDIO_DEBUG if (sbdspdebug > 1) Dprintf("sbdsp_dma_input: cc=%d 0x%x (0x%x)\n", cc, intr, arg); #endif if (sc->sc_channels == 2 && (cc & 1)) { DPRINTF(("sbdsp_dma_input: stereo input, odd bytecnt\n")); return EIO; } iobase = sc->sc_iobase; if (sc->sc_dmadir != SB_DMA_IN) { if (ISSBPRO(sc)) { if (sc->sc_channels == 2) { if (ISJAZZ16(sc) && sc->sc_precision == 16) { if (sbdsp_wdsp(iobase, JAZZ16_RECORD_STEREO) < 0) { goto badmode; } } else if (sbdsp_wdsp(iobase, SB_DSP_RECORD_STEREO) < 0) goto badmode; sbdsp_mix_write(sc, SBP_INFILTER, (sbdsp_mix_read(sc, SBP_INFILTER) & ~SBP_IFILTER_MASK) | SBP_FILTER_OFF); } else { if (ISJAZZ16(sc) && sc->sc_precision == 16) { if (sbdsp_wdsp(iobase, JAZZ16_RECORD_MONO) < 0) { goto badmode; } } else if (sbdsp_wdsp(iobase, SB_DSP_RECORD_MONO) < 0) goto badmode; sbdsp_mix_write(sc, SBP_INFILTER, (sbdsp_mix_read(sc, SBP_INFILTER) & ~SBP_IFILTER_MASK) | sc->in_filter); } } if (ISSB16CLASS(sc)) { if (sbdsp_wdsp(iobase, SB_DSP16_INPUTRATE) < 0 || sbdsp_wdsp(iobase, sc->sc_irate >> 8) < 0 || sbdsp_wdsp(iobase, sc->sc_irate) < 0) goto giveup; } else sbdsp_set_timeconst(sc, sc->sc_itc); sc->sc_dmadir = SB_DMA_IN; } isa_dmastart(DMAMODE_READ, p, cc, sc->sc_drq); sc->sc_intr = intr; sc->sc_arg = arg; sc->dmaflags = DMAMODE_READ; sc->dmaaddr = p; sc->dmacnt = cc; /* DMA controller is strange...? */ if ((ISSB16CLASS(sc) && sc->sc_precision == 16) || (ISJAZZ16(sc) && sc->sc_drq > 3)) cc >>= 1; --cc; if (ISSB16CLASS(sc)) { if (sbdsp_wdsp(iobase, sc->sc_precision == 16 ? SB_DSP16_RDMA_16 : SB_DSP16_RDMA_8) < 0 || sbdsp_wdsp(iobase, (sc->sc_precision == 16 ? 0x10 : 0x00) | (sc->sc_channels == 2 ? 0x20 : 0x00)) < 0 || sbdsp16_wait(iobase) || sbdsp_wdsp(iobase, cc) < 0 || sbdsp_wdsp(iobase, cc >> 8) < 0) { DPRINTF(("sbdsp_dma_input: SB16 DMA start failed\n")); goto giveup; } } else if (sc->sc_imode == SB_ADAC_LS) { if (sbdsp_wdsp(iobase, SB_DSP_RDMA) < 0 || sbdsp_wdsp(iobase, cc) < 0 || sbdsp_wdsp(iobase, cc >> 8) < 0) { DPRINTF(("sbdsp_dma_input: LS DMA start failed\n")); goto giveup; } } else { if (cc != sc->sc_last_hs_size) { if (sbdsp_wdsp(iobase, SB_DSP_BLOCKSIZE) < 0 || sbdsp_wdsp(iobase, cc) < 0 || sbdsp_wdsp(iobase, cc >> 8) < 0) { DPRINTF(("sbdsp_dma_input: HS DMA start failed\n")); goto giveup; } sc->sc_last_hs_size = cc; } if (sbdsp_wdsp(iobase, SB_DSP_HS_INPUT) < 0) { DPRINTF(("sbdsp_dma_input: HS DMA restart failed\n")); goto giveup; } } return 0; giveup: sbdsp_reset(sc); return EIO; badmode: DPRINTF(("sbdsp_dma_input: can't set %s mode\n", sc->sc_channels == 2 ? "stereo" : "mono")); return EIO; } int sbdsp_dma_output(addr, p, cc, intr, arg) void *addr; void *p; int cc; void (*intr) __P((void *)); void *arg; { register struct sbdsp_softc *sc = addr; register int iobase; #ifdef AUDIO_DEBUG if (sbdspdebug > 1) Dprintf("sbdsp_dma_output: cc=%d 0x%x (0x%x)\n", cc, intr, arg); #endif if (sc->sc_channels == 2 && (cc & 1)) { DPRINTF(("stereo playback odd bytes (%d)\n", cc)); return EIO; } iobase = sc->sc_iobase; if (sc->sc_dmadir != SB_DMA_OUT) { if (ISSBPRO(sc)) { /* make sure we re-set stereo mixer bit when we start output. */ sbdsp_mix_write(sc, SBP_STEREO, (sbdsp_mix_read(sc, SBP_STEREO) & ~SBP_PLAYMODE_MASK) | (sc->sc_channels == 2 ? SBP_PLAYMODE_STEREO : SBP_PLAYMODE_MONO)); if (ISJAZZ16(sc)) { /* Yes, we write the record mode to set 16-bit playback mode. weird, huh? */ if (sc->sc_precision == 16) { sbdsp_wdsp(iobase, sc->sc_channels == 2 ? JAZZ16_RECORD_STEREO : JAZZ16_RECORD_MONO); } else { sbdsp_wdsp(iobase, sc->sc_channels == 2 ? SB_DSP_RECORD_STEREO : SB_DSP_RECORD_MONO); } } } if (ISSB16CLASS(sc)) { if (sbdsp_wdsp(iobase, SB_DSP16_OUTPUTRATE) < 0 || sbdsp_wdsp(iobase, sc->sc_orate >> 8) < 0 || sbdsp_wdsp(iobase, sc->sc_orate) < 0) goto giveup; } else sbdsp_set_timeconst(sc, sc->sc_otc); sc->sc_dmadir = SB_DMA_OUT; } isa_dmastart(DMAMODE_WRITE, p, cc, sc->sc_drq); sc->sc_intr = intr; sc->sc_arg = arg; sc->dmaflags = DMAMODE_WRITE; sc->dmaaddr = p; sc->dmacnt = cc; /* a vagary of how DMA works, apparently. */ if ((ISSB16CLASS(sc) && sc->sc_precision == 16) || (ISJAZZ16(sc) && sc->sc_drq > 3)) cc >>= 1; --cc; if (ISSB16CLASS(sc)) { if (sbdsp_wdsp(iobase, sc->sc_precision == 16 ? SB_DSP16_WDMA_16 : SB_DSP16_WDMA_8) < 0 || sbdsp_wdsp(iobase, (sc->sc_precision == 16 ? 0x10 : 0x00) | (sc->sc_channels == 2 ? 0x20 : 0x00)) < 0 || sbdsp16_wait(iobase) || sbdsp_wdsp(iobase, cc) < 0 || sbdsp_wdsp(iobase, cc >> 8) < 0) { DPRINTF(("sbdsp_dma_output: SB16 DMA start failed\n")); goto giveup; } } else if (sc->sc_omode == SB_ADAC_LS) { if (sbdsp_wdsp(iobase, SB_DSP_WDMA) < 0 || sbdsp_wdsp(iobase, cc) < 0 || sbdsp_wdsp(iobase, cc >> 8) < 0) { DPRINTF(("sbdsp_dma_output: LS DMA start failed\n")); goto giveup; } } else { if (cc != sc->sc_last_hs_size) { if (sbdsp_wdsp(iobase, SB_DSP_BLOCKSIZE) < 0 || sbdsp_wdsp(iobase, cc) < 0 || sbdsp_wdsp(iobase, cc >> 8) < 0) { DPRINTF(("sbdsp_dma_output: HS DMA start failed\n")); goto giveup; } sc->sc_last_hs_size = cc; } if (sbdsp_wdsp(iobase, SB_DSP_HS_OUTPUT) < 0) { DPRINTF(("sbdsp_dma_output: HS DMA restart failed\n")); goto giveup; } } return 0; giveup: sbdsp_reset(sc); return EIO; } /* * Only the DSP unit on the sound blaster generates interrupts. * There are three cases of interrupt: reception of a midi byte * (when mode is enabled), completion of dma transmission, or * completion of a dma reception. The three modes are mutually * exclusive so we know a priori which event has occurred. */ int sbdsp_intr(arg) void *arg; { register struct sbdsp_softc *sc = arg; u_char x; #ifdef AUDIO_DEBUG if (sbdspdebug > 1) Dprintf("sbdsp_intr: intr=0x%x\n", sc->sc_intr); #endif if (!isa_dmafinished(sc->sc_drq)) { #ifdef AUDIO_DEBUG printf("sbdsp_intr: not finished\n"); #endif return 0; } sc->sc_interrupts++; /* clear interrupt */ #ifdef notyet x = sbdsp_mix_read(sc, 0x82); x = inb(sc->sc_iobase + 15); #endif x = inb(sc->sc_iobase + SBP_DSP_RSTAT); delay(10); #if 0 if (sc->sc_mintr != 0) { x = sbdsp_rdsp(sc->sc_iobase); (*sc->sc_mintr)(sc->sc_arg, x); } else #endif if (sc->sc_intr != 0) { isa_dmadone(sc->dmaflags, sc->dmaaddr, sc->dmacnt, sc->sc_drq); (*sc->sc_intr)(sc->sc_arg); } else return 0; return 1; } #if 0 /* * Enter midi uart mode and arrange for read interrupts * to vector to `intr'. This puts the card in a mode * which allows only midi I/O; the card must be reset * to leave this mode. Unfortunately, the card does not * use transmit interrupts, so bytes must be output * using polling. To keep the polling overhead to a * minimum, output should be driven off a timer. * This is a little tricky since only 320us separate * consecutive midi bytes. */ void sbdsp_set_midi_mode(sc, intr, arg) struct sbdsp_softc *sc; void (*intr)(); void *arg; { sbdsp_wdsp(sc->sc_iobase, SB_MIDI_UART_INTR); sc->sc_mintr = intr; sc->sc_intr = 0; sc->sc_arg = arg; } /* * Write a byte to the midi port, when in midi uart mode. */ void sbdsp_midi_output(sc, v) struct sbdsp_softc *sc; int v; { if (sbdsp_wdsp(sc->sc_iobase, v) < 0) ++sberr.wmidi; } #endif u_int sbdsp_get_silence(encoding) int encoding; { #define ULAW_SILENCE 0x7f #define LINEAR_SILENCE 0 u_int auzero; switch (encoding) { case AUDIO_ENCODING_ULAW: auzero = ULAW_SILENCE; break; case AUDIO_ENCODING_PCM16: default: auzero = LINEAR_SILENCE; break; } return (auzero); } int sbdsp_setfd(addr, flag) void *addr; int flag; { /* Can't do full-duplex */ return(ENOTTY); } int sbdsp_mixer_set_port(addr, cp) void *addr; mixer_ctrl_t *cp; { register struct sbdsp_softc *sc = addr; int src, gain; DPRINTF(("sbdsp_mixer_set_port: port=%d num_channels=%d\n", cp->dev, cp->un.value.num_channels)); if (!ISSBPROCLASS(sc)) return EINVAL; /* * Everything is a value except for SBPro BASS/TREBLE and * RECORD_SOURCE */ switch (cp->dev) { case SB_SPEAKER: cp->dev = SB_MASTER_VOL; case SB_MIC_PORT: case SB_LINE_IN_PORT: case SB_DAC_PORT: case SB_FM_PORT: case SB_CD_PORT: case SB_MASTER_VOL: if (cp->type != AUDIO_MIXER_VALUE) return EINVAL; /* * All the mixer ports are stereo except for the microphone. * If we get a single-channel gain value passed in, then we * duplicate it to both left and right channels. */ switch (cp->dev) { case SB_MIC_PORT: if (cp->un.value.num_channels != 1) return EINVAL; /* handle funny microphone gain */ gain = SBP_AGAIN_TO_MICGAIN(cp->un.value.level[AUDIO_MIXER_LEVEL_MONO]); break; case SB_LINE_IN_PORT: case SB_DAC_PORT: case SB_FM_PORT: case SB_CD_PORT: case SB_MASTER_VOL: switch (cp->un.value.num_channels) { case 1: gain = sbdsp_mono_vol(SBP_AGAIN_TO_SBGAIN(cp->un.value.level[AUDIO_MIXER_LEVEL_MONO])); break; case 2: gain = sbdsp_stereo_vol(SBP_AGAIN_TO_SBGAIN(cp->un.value.level[AUDIO_MIXER_LEVEL_LEFT]), SBP_AGAIN_TO_SBGAIN(cp->un.value.level[AUDIO_MIXER_LEVEL_RIGHT])); break; default: return EINVAL; } break; default: return EINVAL; } switch (cp->dev) { case SB_MIC_PORT: src = SBP_MIC_VOL; break; case SB_MASTER_VOL: src = SBP_MASTER_VOL; break; case SB_LINE_IN_PORT: src = SBP_LINE_VOL; break; case SB_DAC_PORT: src = SBP_DAC_VOL; break; case SB_FM_PORT: src = SBP_FM_VOL; break; case SB_CD_PORT: src = SBP_CD_VOL; break; default: return EINVAL; } sbdsp_mix_write(sc, src, gain); sc->gain[cp->dev] = gain; break; case SB_TREBLE: case SB_BASS: case SB_RECORD_SOURCE: if (cp->type != AUDIO_MIXER_ENUM) return EINVAL; switch (cp->dev) { case SB_TREBLE: return sbdsp_set_ifilter(addr, cp->un.ord ? SBP_TREBLE_EQ : 0); case SB_BASS: return sbdsp_set_ifilter(addr, cp->un.ord ? SBP_BASS_EQ : 0); case SB_RECORD_SOURCE: return sbdsp_set_in_port(addr, cp->un.ord); } break; default: return EINVAL; } return (0); } int sbdsp_mixer_get_port(addr, cp) void *addr; mixer_ctrl_t *cp; { register struct sbdsp_softc *sc = addr; int gain; DPRINTF(("sbdsp_mixer_get_port: port=%d", cp->dev)); if (!ISSBPROCLASS(sc)) return EINVAL; switch (cp->dev) { case SB_SPEAKER: cp->dev = SB_MASTER_VOL; case SB_MIC_PORT: case SB_LINE_IN_PORT: case SB_DAC_PORT: case SB_FM_PORT: case SB_CD_PORT: case SB_MASTER_VOL: gain = sc->gain[cp->dev]; switch (cp->dev) { case SB_MIC_PORT: if (cp->un.value.num_channels != 1) return EINVAL; cp->un.value.level[AUDIO_MIXER_LEVEL_MONO] = SBP_MICGAIN_TO_AGAIN(gain); break; case SB_LINE_IN_PORT: case SB_DAC_PORT: case SB_FM_PORT: case SB_CD_PORT: case SB_MASTER_VOL: switch (cp->un.value.num_channels) { case 1: cp->un.value.level[AUDIO_MIXER_LEVEL_MONO] = SBP_SBGAIN_TO_AGAIN(gain); break; case 2: cp->un.value.level[AUDIO_MIXER_LEVEL_LEFT] = SBP_LEFTGAIN(gain); cp->un.value.level[AUDIO_MIXER_LEVEL_RIGHT] = SBP_RIGHTGAIN(gain); break; default: return EINVAL; } break; } break; case SB_TREBLE: case SB_BASS: case SB_RECORD_SOURCE: switch (cp->dev) { case SB_TREBLE: cp->un.ord = sbdsp_get_ifilter(addr) == SBP_TREBLE_EQ; return 0; case SB_BASS: cp->un.ord = sbdsp_get_ifilter(addr) == SBP_BASS_EQ; return 0; case SB_RECORD_SOURCE: cp->un.ord = sbdsp_get_in_port(addr); return 0; } break; default: return EINVAL; } return (0); } int sbdsp_mixer_query_devinfo(addr, dip) void *addr; register mixer_devinfo_t *dip; { register struct sbdsp_softc *sc = addr; DPRINTF(("sbdsp_mixer_query_devinfo: index=%d\n", dip->index)); switch (dip->index) { case SB_MIC_PORT: dip->type = AUDIO_MIXER_VALUE; dip->mixer_class = SB_INPUT_CLASS; dip->prev = AUDIO_MIXER_LAST; dip->next = AUDIO_MIXER_LAST; strcpy(dip->label.name, AudioNmicrophone); dip->un.v.num_channels = 1; strcpy(dip->un.v.units.name, AudioNvolume); return 0; case SB_SPEAKER: dip->type = AUDIO_MIXER_VALUE; dip->mixer_class = SB_OUTPUT_CLASS; dip->prev = AUDIO_MIXER_LAST; dip->next = AUDIO_MIXER_LAST; strcpy(dip->label.name, AudioNspeaker); dip->un.v.num_channels = 1; strcpy(dip->un.v.units.name, AudioNvolume); return 0; case SB_INPUT_CLASS: dip->type = AUDIO_MIXER_CLASS; dip->mixer_class = SB_INPUT_CLASS; dip->next = dip->prev = AUDIO_MIXER_LAST; strcpy(dip->label.name, AudioCInputs); return 0; case SB_OUTPUT_CLASS: dip->type = AUDIO_MIXER_CLASS; dip->mixer_class = SB_OUTPUT_CLASS; dip->next = dip->prev = AUDIO_MIXER_LAST; strcpy(dip->label.name, AudioCOutputs); return 0; } if (ISSBPROCLASS(sc)) { switch (dip->index) { case SB_LINE_IN_PORT: dip->type = AUDIO_MIXER_VALUE; dip->mixer_class = SB_INPUT_CLASS; dip->prev = AUDIO_MIXER_LAST; dip->next = AUDIO_MIXER_LAST; strcpy(dip->label.name, AudioNline); dip->un.v.num_channels = 2; strcpy(dip->un.v.units.name, AudioNvolume); return 0; case SB_DAC_PORT: dip->type = AUDIO_MIXER_VALUE; dip->mixer_class = SB_INPUT_CLASS; dip->prev = AUDIO_MIXER_LAST; dip->next = AUDIO_MIXER_LAST; strcpy(dip->label.name, AudioNdac); dip->un.v.num_channels = 2; strcpy(dip->un.v.units.name, AudioNvolume); return 0; case SB_CD_PORT: dip->type = AUDIO_MIXER_VALUE; dip->mixer_class = SB_INPUT_CLASS; dip->prev = AUDIO_MIXER_LAST; dip->next = AUDIO_MIXER_LAST; strcpy(dip->label.name, AudioNcd); dip->un.v.num_channels = 2; strcpy(dip->un.v.units.name, AudioNvolume); return 0; case SB_FM_PORT: dip->type = AUDIO_MIXER_VALUE; dip->mixer_class = SB_INPUT_CLASS; dip->prev = AUDIO_MIXER_LAST; dip->next = AUDIO_MIXER_LAST; strcpy(dip->label.name, AudioNfmsynth); dip->un.v.num_channels = 2; strcpy(dip->un.v.units.name, AudioNvolume); return 0; case SB_MASTER_VOL: dip->type = AUDIO_MIXER_VALUE; dip->mixer_class = SB_OUTPUT_CLASS; dip->prev = AUDIO_MIXER_LAST; dip->next = AUDIO_MIXER_LAST; strcpy(dip->label.name, AudioNvolume); dip->un.v.num_channels = 2; strcpy(dip->un.v.units.name, AudioNvolume); return 0; case SB_RECORD_SOURCE: dip->mixer_class = SB_RECORD_CLASS; dip->type = AUDIO_MIXER_ENUM; dip->prev = AUDIO_MIXER_LAST; dip->next = AUDIO_MIXER_LAST; strcpy(dip->label.name, AudioNsource); dip->un.e.num_mem = 3; strcpy(dip->un.e.member[0].label.name, AudioNmicrophone); dip->un.e.member[0].ord = SB_MIC_PORT; strcpy(dip->un.e.member[1].label.name, AudioNcd); dip->un.e.member[1].ord = SB_CD_PORT; strcpy(dip->un.e.member[2].label.name, AudioNline); dip->un.e.member[2].ord = SB_LINE_IN_PORT; return 0; case SB_BASS: dip->type = AUDIO_MIXER_ENUM; dip->mixer_class = SB_INPUT_CLASS; dip->prev = AUDIO_MIXER_LAST; dip->next = AUDIO_MIXER_LAST; strcpy(dip->label.name, AudioNbass); dip->un.e.num_mem = 2; strcpy(dip->un.e.member[0].label.name, AudioNoff); dip->un.e.member[0].ord = 0; strcpy(dip->un.e.member[1].label.name, AudioNon); dip->un.e.member[1].ord = 1; return 0; case SB_TREBLE: dip->type = AUDIO_MIXER_ENUM; dip->mixer_class = SB_INPUT_CLASS; dip->prev = AUDIO_MIXER_LAST; dip->next = AUDIO_MIXER_LAST; strcpy(dip->label.name, AudioNtreble); dip->un.e.num_mem = 2; strcpy(dip->un.e.member[0].label.name, AudioNoff); dip->un.e.member[0].ord = 0; strcpy(dip->un.e.member[1].label.name, AudioNon); dip->un.e.member[1].ord = 1; return 0; case SB_RECORD_CLASS: /* record source class */ dip->type = AUDIO_MIXER_CLASS; dip->mixer_class = SB_RECORD_CLASS; dip->next = dip->prev = AUDIO_MIXER_LAST; strcpy(dip->label.name, AudioCRecord); return 0; } } return ENXIO; }