/* $NetBSD: sbdsp.c,v 1.76 1997/10/19 07:42:39 augustss 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 * . * Lots of rewrites by Lennart Augustsson * with information from SB "Hardware Programming Guide" and the * Linux drivers. */ #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include #ifdef AUDIO_DEBUG #define DPRINTF(x) if (sbdspdebug) printf 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; /* * 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 */ /* XXX Should we round the tc? #define SB_RATE_TO_TC(x) (((65536 - 256 * 1000000 / (x)) + 128) >> 8) */ #define SB_RATE_TO_TC(x) (256 - 1000000 / (x)) #define SB_TC_TO_RATE(tc) (1000000 / (256 - (tc))) struct sbmode { short model; u_char channels; u_char precision; u_short lowrate, highrate; u_char cmd; u_char cmdchan; }; static struct sbmode sbpmodes[] = { { SB_1, 1, 8, 4000, 22727, SB_DSP_WDMA }, { SB_20, 1, 8, 4000, 22727, SB_DSP_WDMA_LOOP }, { SB_2x, 1, 8, 4000, 22727, SB_DSP_WDMA_LOOP }, { SB_2x, 1, 8, 22727, 45454, SB_DSP_HS_OUTPUT }, { SB_PRO, 1, 8, 4000, 22727, SB_DSP_WDMA_LOOP }, { SB_PRO, 1, 8, 22727, 45454, SB_DSP_HS_OUTPUT }, { SB_PRO, 2, 8, 11025, 22727, SB_DSP_HS_OUTPUT }, /* Yes, we write the record mode to set 16-bit playback mode. weird, huh? */ { SB_JAZZ, 1, 8, 4000, 22727, SB_DSP_WDMA_LOOP, SB_DSP_RECORD_MONO }, { SB_JAZZ, 1, 8, 22727, 45454, SB_DSP_HS_OUTPUT, SB_DSP_RECORD_MONO }, { SB_JAZZ, 2, 8, 11025, 22727, SB_DSP_HS_OUTPUT, SB_DSP_RECORD_STEREO }, { SB_JAZZ, 1, 16, 4000, 22727, SB_DSP_WDMA_LOOP, JAZZ16_RECORD_MONO }, { SB_JAZZ, 1, 16, 22727, 45454, SB_DSP_HS_OUTPUT, JAZZ16_RECORD_MONO }, { SB_JAZZ, 2, 16, 11025, 22727, SB_DSP_HS_OUTPUT, JAZZ16_RECORD_STEREO }, { SB_16, 1, 8, 5000, 45000, SB_DSP16_WDMA_8 }, { SB_16, 2, 8, 5000, 45000, SB_DSP16_WDMA_8 }, #define PLAY16 15 /* must be the index of the next entry in the table */ { SB_16, 1, 16, 5000, 45000, SB_DSP16_WDMA_16 }, { SB_16, 2, 16, 5000, 45000, SB_DSP16_WDMA_16 }, { -1 } }; static struct sbmode sbrmodes[] = { { SB_1, 1, 8, 4000, 12987, SB_DSP_RDMA }, { SB_20, 1, 8, 4000, 12987, SB_DSP_RDMA_LOOP }, { SB_2x, 1, 8, 4000, 12987, SB_DSP_RDMA_LOOP }, { SB_2x, 1, 8, 12987, 14925, SB_DSP_HS_INPUT }, { SB_PRO, 1, 8, 4000, 22727, SB_DSP_RDMA_LOOP, SB_DSP_RECORD_MONO }, { SB_PRO, 1, 8, 22727, 45454, SB_DSP_HS_INPUT, SB_DSP_RECORD_MONO }, { SB_PRO, 2, 8, 11025, 22727, SB_DSP_HS_INPUT, SB_DSP_RECORD_STEREO }, { SB_JAZZ, 1, 8, 4000, 22727, SB_DSP_RDMA_LOOP, SB_DSP_RECORD_MONO }, { SB_JAZZ, 1, 8, 22727, 45454, SB_DSP_HS_INPUT, SB_DSP_RECORD_MONO }, { SB_JAZZ, 2, 8, 11025, 22727, SB_DSP_HS_INPUT, SB_DSP_RECORD_STEREO }, { SB_JAZZ, 1, 16, 4000, 22727, SB_DSP_RDMA_LOOP, JAZZ16_RECORD_MONO }, { SB_JAZZ, 1, 16, 22727, 45454, SB_DSP_HS_INPUT, JAZZ16_RECORD_MONO }, { SB_JAZZ, 2, 16, 11025, 22727, SB_DSP_HS_INPUT, JAZZ16_RECORD_STEREO }, { SB_16, 1, 8, 5000, 45000, SB_DSP16_RDMA_8 }, { SB_16, 2, 8, 5000, 45000, SB_DSP16_RDMA_8 }, { SB_16, 1, 16, 5000, 45000, SB_DSP16_RDMA_16 }, { SB_16, 2, 16, 5000, 45000, SB_DSP16_RDMA_16 }, { -1 } }; void sbversion __P((struct sbdsp_softc *)); void sbdsp_jazz16_probe __P((struct sbdsp_softc *)); void sbdsp_set_mixer_gain __P((struct sbdsp_softc *sc, int port)); void sbdsp_to __P((void *)); void sbdsp_pause __P((struct sbdsp_softc *)); int sbdsp_set_timeconst __P((struct sbdsp_softc *, int)); int sbdsp16_set_rate __P((struct sbdsp_softc *, int, int)); int sbdsp_set_in_ports __P((struct sbdsp_softc *, int)); void sbdsp_set_ifilter __P((void *, int)); int sbdsp_get_ifilter __P((void *)); static int sbdsp_dma_setup_input __P((struct sbdsp_softc *sc)); static int sbdsp_dma_setup_output __P((struct sbdsp_softc *sc)); static int sbdsp_adjust __P((int, int)); #ifdef AUDIO_DEBUG void sb_printsc __P((struct sbdsp_softc *)); void sb_printsc(sc) struct sbdsp_softc *sc; { int i; printf("open %d dmachan %d/%d %d/%d iobase 0x%x irq %d\n", (int)sc->sc_open, sc->sc_i.run, sc->sc_o.run, sc->sc_drq8, sc->sc_drq16, sc->sc_iobase, sc->sc_irq); printf("irate %d itc %x orate %d otc %x\n", sc->sc_i.rate, sc->sc_i.tc, sc->sc_o.rate, sc->sc_o.tc); printf("spkron %u nintr %lu\n", sc->spkr_state, sc->sc_interrupts); printf("intr8 %p arg8 %p\n", sc->sc_intr8, sc->sc_arg16); printf("intr16 %p arg16 %p\n", sc->sc_intr8, sc->sc_arg16); printf("gain:"); for (i = 0; i < SB_NDEVS; i++) printf(" %u,%u", sc->gain[i][SB_LEFT], sc->gain[i][SB_RIGHT]); printf("\n"); } #endif /* AUDIO_DEBUG */ /* * 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 & 1) sbdsp_jazz16_probe(sc); else sbversion(sc); if (sc->sc_model == SB_UNK) { /* Unknown SB model found. */ DPRINTF(("sbdsp: unknown SB model found\n")); return 0; } return 1; } /* * Try add-on stuff for Jazz16. */ void 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}; bus_space_tag_t iot = sc->sc_iot; bus_space_handle_t ioh; sbversion(sc); DPRINTF(("jazz16 probe\n")); if (bus_space_map(iot, JAZZ16_CONFIG_PORT, 1, 0, &ioh)) { DPRINTF(("bus map failed\n")); return; } if (jazz16_drq_conf[sc->sc_drq8] == (u_char)-1 || jazz16_irq_conf[sc->sc_irq] == (u_char)-1) { DPRINTF(("drq/irq check failed\n")); goto done; /* give up, we can't do it. */ } bus_space_write_1(iot, ioh, 0, JAZZ16_WAKEUP); delay(10000); /* delay 10 ms */ bus_space_write_1(iot, ioh, 0, JAZZ16_SETBASE); bus_space_write_1(iot, ioh, 0, sc->sc_iobase & 0x70); if (sbdsp_reset(sc) < 0) { DPRINTF(("sbdsp_reset check failed\n")); goto done; /* XXX? what else could we do? */ } if (sbdsp_wdsp(sc, JAZZ16_READ_VER)) { DPRINTF(("read16 setup failed\n")); goto done; } if (sbdsp_rdsp(sc) != JAZZ16_VER_JAZZ) { DPRINTF(("read16 failed\n")); goto done; } /* XXX set both 8 & 16-bit drq to same channel, it works fine. */ sc->sc_drq16 = sc->sc_drq8; if (sbdsp_wdsp(sc, JAZZ16_SET_DMAINTR) || sbdsp_wdsp(sc, (jazz16_drq_conf[sc->sc_drq16] << 4) | jazz16_drq_conf[sc->sc_drq8]) || sbdsp_wdsp(sc, jazz16_irq_conf[sc->sc_irq])) { DPRINTF(("sbdsp: can't write jazz16 probe stuff\n")); } else { DPRINTF(("jazz16 detected!\n")); sc->sc_model = SB_JAZZ; sc->sc_mixer_model = SBM_CT1345; /* XXX really? */ } done: bus_space_unmap(iot, ioh, 1); } /* * Attach hardware to driver, attach hardware driver to audio * pseudo-device driver . */ void sbdsp_attach(sc) struct sbdsp_softc *sc; { struct audio_params pparams, rparams; int i; u_int v; /* * Create our DMA maps. */ if (sc->sc_drq8 != -1) { if (isa_dmamap_create(sc->sc_isa, sc->sc_drq8, MAX_ISADMA, BUS_DMA_NOWAIT|BUS_DMA_ALLOCNOW)) { printf("%s: can't create map for drq %d\n", sc->sc_dev.dv_xname, sc->sc_drq8); return; } } if (sc->sc_drq16 != -1 && sc->sc_drq16 != sc->sc_drq8) { if (isa_dmamap_create(sc->sc_isa, sc->sc_drq16, MAX_ISADMA, BUS_DMA_NOWAIT|BUS_DMA_ALLOCNOW)) { printf("%s: can't create map for drq %d\n", sc->sc_dev.dv_xname, sc->sc_drq16); return; } } pparams = audio_default; rparams = audio_default; sbdsp_set_params(sc, AUMODE_RECORD|AUMODE_PLAY, 0, &pparams, &rparams); sbdsp_set_in_ports(sc, 1 << SB_MIC_VOL); if (sc->sc_mixer_model != SBM_NONE) { /* Reset the mixer.*/ sbdsp_mix_write(sc, SBP_MIX_RESET, SBP_MIX_RESET); /* And set our own default values */ for (i = 0; i < SB_NDEVS; i++) { switch(i) { case SB_MIC_VOL: case SB_LINE_IN_VOL: v = 0; break; case SB_BASS: case SB_TREBLE: v = SB_ADJUST_GAIN(sc, AUDIO_MAX_GAIN/2); break; case SB_CD_IN_MUTE: case SB_MIC_IN_MUTE: case SB_LINE_IN_MUTE: case SB_MIDI_IN_MUTE: case SB_CD_SWAP: case SB_MIC_SWAP: case SB_LINE_SWAP: case SB_MIDI_SWAP: case SB_CD_OUT_MUTE: case SB_MIC_OUT_MUTE: case SB_LINE_OUT_MUTE: v = 0; break; default: v = SB_ADJUST_GAIN(sc, AUDIO_MAX_GAIN * 3 / 4); break; } sc->gain[i][SB_LEFT] = sc->gain[i][SB_RIGHT] = v; sbdsp_set_mixer_gain(sc, i); } sc->in_filter = 0; /* no filters turned on, please */ } printf(": dsp v%d.%02d%s\n", SBVER_MAJOR(sc->sc_version), SBVER_MINOR(sc->sc_version), sc->sc_model == SB_JAZZ ? ": " : ""); sc->sc_fullduplex = ISSB16CLASS(sc) && sc->sc_drq8 != -1 && sc->sc_drq16 != -1 && sc->sc_drq8 != sc->sc_drq16; } void sbdsp_mix_write(sc, mixerport, val) struct sbdsp_softc *sc; int mixerport; int val; { bus_space_tag_t iot = sc->sc_iot; bus_space_handle_t ioh = sc->sc_ioh; int s; s = splaudio(); bus_space_write_1(iot, ioh, SBP_MIXER_ADDR, mixerport); delay(20); bus_space_write_1(iot, ioh, SBP_MIXER_DATA, val); delay(30); splx(s); } int sbdsp_mix_read(sc, mixerport) struct sbdsp_softc *sc; int mixerport; { bus_space_tag_t iot = sc->sc_iot; bus_space_handle_t ioh = sc->sc_ioh; int val; int s; s = splaudio(); bus_space_write_1(iot, ioh, SBP_MIXER_ADDR, mixerport); delay(20); val = bus_space_read_1(iot, ioh, SBP_MIXER_DATA); delay(30); splx(s); return val; } /* * Various routines to interface to higher level audio driver */ int sbdsp_query_encoding(addr, fp) void *addr; struct audio_encoding *fp; { struct sbdsp_softc *sc = addr; int emul; emul = ISSB16CLASS(sc) ? 0 : AUDIO_ENCODINGFLAG_EMULATED; switch (fp->index) { case 0: strcpy(fp->name, AudioEulinear); fp->encoding = AUDIO_ENCODING_ULINEAR; fp->precision = 8; fp->flags = 0; return 0; case 1: strcpy(fp->name, AudioEmulaw); fp->encoding = AUDIO_ENCODING_ULAW; fp->precision = 8; fp->flags = AUDIO_ENCODINGFLAG_EMULATED; return 0; case 2: strcpy(fp->name, AudioEalaw); fp->encoding = AUDIO_ENCODING_ALAW; fp->precision = 8; fp->flags = AUDIO_ENCODINGFLAG_EMULATED; return 0; case 3: strcpy(fp->name, AudioEslinear); fp->encoding = AUDIO_ENCODING_SLINEAR; fp->precision = 8; fp->flags = emul; return 0; } if (!ISSB16CLASS(sc) && sc->sc_model != SB_JAZZ) return EINVAL; switch(fp->index) { case 4: strcpy(fp->name, AudioEslinear_le); fp->encoding = AUDIO_ENCODING_SLINEAR_LE; fp->precision = 16; fp->flags = 0; return 0; case 5: strcpy(fp->name, AudioEulinear_le); fp->encoding = AUDIO_ENCODING_ULINEAR_LE; fp->precision = 16; fp->flags = emul; return 0; case 6: strcpy(fp->name, AudioEslinear_be); fp->encoding = AUDIO_ENCODING_SLINEAR_BE; fp->precision = 16; fp->flags = AUDIO_ENCODINGFLAG_EMULATED; return 0; case 7: strcpy(fp->name, AudioEulinear_be); fp->encoding = AUDIO_ENCODING_ULINEAR_BE; fp->precision = 16; fp->flags = AUDIO_ENCODINGFLAG_EMULATED; return 0; default: return EINVAL; } return 0; } int sbdsp_set_params(addr, setmode, usemode, play, rec) void *addr; int setmode, usemode; struct audio_params *play, *rec; { struct sbdsp_softc *sc = addr; struct sbmode *m; u_int rate, tc, bmode; void (*swcode) __P((void *, u_char *buf, int cnt)); int factor; int model; int chan; struct audio_params *p; int mode; model = sc->sc_model; if (model > SB_16) model = SB_16; /* later models work like SB16 */ /* Set first record info, then play info */ for(mode = AUMODE_RECORD; mode != -1; mode = mode == AUMODE_RECORD ? AUMODE_PLAY : -1) { if ((setmode & mode) == 0) continue; p = mode == AUMODE_PLAY ? play : rec; /* Locate proper commands */ for(m = mode == AUMODE_PLAY ? sbpmodes : sbrmodes; m->model != -1; m++) { if (model == m->model && p->channels == m->channels && p->precision == m->precision && p->sample_rate >= m->lowrate && p->sample_rate < m->highrate) break; } if (m->model == -1) return EINVAL; rate = p->sample_rate; swcode = 0; factor = 1; tc = 1; bmode = -1; if (model == SB_16) { switch (p->encoding) { case AUDIO_ENCODING_SLINEAR_BE: if (p->precision == 16) swcode = swap_bytes; /* fall into */ case AUDIO_ENCODING_SLINEAR_LE: bmode = SB_BMODE_SIGNED; break; case AUDIO_ENCODING_ULINEAR_BE: if (p->precision == 16) swcode = swap_bytes; /* fall into */ case AUDIO_ENCODING_ULINEAR_LE: bmode = SB_BMODE_UNSIGNED; break; case AUDIO_ENCODING_ULAW: if (mode == AUMODE_PLAY) { swcode = mulaw_to_ulinear16; factor = 2; m = &sbpmodes[PLAY16]; } else swcode = ulinear8_to_mulaw; bmode = SB_BMODE_UNSIGNED; break; case AUDIO_ENCODING_ALAW: if (mode == AUMODE_PLAY) { swcode = alaw_to_ulinear16; factor = 2; m = &sbpmodes[PLAY16]; } else swcode = ulinear8_to_alaw; bmode = SB_BMODE_UNSIGNED; break; default: return EINVAL; } if (p->channels == 2) bmode |= SB_BMODE_STEREO; } else if (m->model == SB_JAZZ && m->precision == 16) { switch (p->encoding) { case AUDIO_ENCODING_SLINEAR_LE: break; case AUDIO_ENCODING_ULINEAR_LE: swcode = change_sign16; break; case AUDIO_ENCODING_SLINEAR_BE: swcode = swap_bytes; break; case AUDIO_ENCODING_ULINEAR_BE: swcode = mode == AUMODE_PLAY ? swap_bytes_change_sign16 : change_sign16_swap_bytes; break; case AUDIO_ENCODING_ULAW: swcode = mode == AUMODE_PLAY ? mulaw_to_ulinear8 : ulinear8_to_mulaw; break; case AUDIO_ENCODING_ALAW: swcode = mode == AUMODE_PLAY ? alaw_to_ulinear8 : ulinear8_to_alaw; break; default: return EINVAL; } tc = SB_RATE_TO_TC(p->sample_rate * p->channels); p->sample_rate = SB_TC_TO_RATE(tc) / p->channels; } else { switch (p->encoding) { case AUDIO_ENCODING_SLINEAR_BE: case AUDIO_ENCODING_SLINEAR_LE: swcode = change_sign8; break; case AUDIO_ENCODING_ULINEAR_BE: case AUDIO_ENCODING_ULINEAR_LE: break; case AUDIO_ENCODING_ULAW: swcode = mode == AUMODE_PLAY ? mulaw_to_ulinear8 : ulinear8_to_mulaw; break; case AUDIO_ENCODING_ALAW: swcode = mode == AUMODE_PLAY ? alaw_to_ulinear8 : ulinear8_to_alaw; break; default: return EINVAL; } tc = SB_RATE_TO_TC(p->sample_rate * p->channels); p->sample_rate = SB_TC_TO_RATE(tc) / p->channels; } chan = m->precision == 16 ? sc->sc_drq16 : sc->sc_drq8; if (mode == AUMODE_PLAY) { sc->sc_o.rate = rate; sc->sc_o.tc = tc; sc->sc_o.modep = m; sc->sc_o.bmode = bmode; sc->sc_o.dmachan = chan; } else { sc->sc_i.rate = rate; sc->sc_i.tc = tc; sc->sc_i.modep = m; sc->sc_i.bmode = bmode; sc->sc_i.dmachan = chan; } p->sw_code = swcode; p->factor = factor; DPRINTF(("sbdsp_set_params: model=%d, mode=%d, rate=%ld, prec=%d, chan=%d, enc=%d -> tc=%02x, cmd=%02x, bmode=%02x, cmdchan=%02x, swcode=%p, factor=%d\n", sc->sc_model, mode, p->sample_rate, p->precision, p->channels, p->encoding, tc, m->cmd, bmode, m->cmdchan, swcode, factor)); } /* * XXX * Should wait for chip to be idle. */ sc->sc_i.run = SB_NOTRUNNING; sc->sc_o.run = SB_NOTRUNNING; if (sc->sc_fullduplex && (usemode & (AUMODE_PLAY | AUMODE_RECORD)) == (AUMODE_PLAY | AUMODE_RECORD) && sc->sc_i.dmachan == sc->sc_o.dmachan) { DPRINTF(("sbdsp_commit: fd=%d, usemode=%d, idma=%d, odma=%d\n", sc->sc_fullduplex, usemode, sc->sc_i.dmachan, sc->sc_o.dmachan)); if (sc->sc_o.dmachan == sc->sc_drq8) { /* Use 16 bit DMA for playing by expanding the samples. */ play->sw_code = linear8_to_linear16; play->factor = 2; sc->sc_o.modep = &sbpmodes[PLAY16]; sc->sc_o.dmachan = sc->sc_drq16; } else { return EINVAL; } } DPRINTF(("sbdsp_set_params ichan=%d, ochan=%d\n", sc->sc_i.dmachan, sc->sc_o.dmachan)); return 0; } void sbdsp_set_ifilter(addr, which) void *addr; int which; { struct sbdsp_softc *sc = addr; int mixval; mixval = sbdsp_mix_read(sc, SBP_INFILTER) & ~SBP_IFILTER_MASK; switch (which) { case 0: mixval |= SBP_FILTER_OFF; break; case SB_TREBLE: mixval |= SBP_FILTER_ON | SBP_IFILTER_HIGH; break; case SB_BASS: mixval |= SBP_FILTER_ON | SBP_IFILTER_LOW; break; default: return; } sc->in_filter = mixval & SBP_IFILTER_MASK; sbdsp_mix_write(sc, SBP_INFILTER, mixval); } int sbdsp_get_ifilter(addr) void *addr; { struct sbdsp_softc *sc = addr; sc->in_filter = sbdsp_mix_read(sc, SBP_INFILTER) & SBP_IFILTER_MASK; switch (sc->in_filter) { case SBP_FILTER_ON|SBP_IFILTER_HIGH: return SB_TREBLE; case SBP_FILTER_ON|SBP_IFILTER_LOW: return SB_BASS; default: return 0; } } int sbdsp_set_in_ports(sc, mask) struct sbdsp_softc *sc; int mask; { int bitsl, bitsr; int sbport; DPRINTF(("sbdsp_set_in_ports: model=%d, mask=%x\n", sc->sc_mixer_model, mask)); switch(sc->sc_mixer_model) { case SBM_NONE: return EINVAL; case SBM_CT1335: if (mask != (1 << SB_MIC_VOL)) return EINVAL; break; case SBM_CT1345: switch (mask) { case 1 << SB_MIC_VOL: sbport = SBP_FROM_MIC; break; case 1 << SB_LINE_IN_VOL: sbport = SBP_FROM_LINE; break; case 1 << SB_CD_VOL: sbport = SBP_FROM_CD; break; default: return (EINVAL); } sbdsp_mix_write(sc, SBP_RECORD_SOURCE, sbport | sc->in_filter); break; case SBM_CT1XX5: case SBM_CT1745: if (mask & ~((1<in_mask = mask; return 0; } int sbdsp_speaker_ctl(addr, newstate) void *addr; int newstate; { 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; { blk &= -4; /* round to biggest sample size */ return blk; } int sbdsp_open(addr, flags) void *addr; int flags; { struct sbdsp_softc *sc = addr; DPRINTF(("sbdsp_open: sc=%p\n", sc)); if (sc->sc_open != 0 || sbdsp_reset(sc) != 0) return ENXIO; sc->sc_open = 1; sc->sc_openflags = flags; sc->sc_mintr = 0; if (ISSBPRO(sc) && sbdsp_wdsp(sc, 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=%p\n", sc)); sc->sc_open = 0; sbdsp_spkroff(sc); sc->spkr_state = SPKR_OFF; sc->sc_intr8 = 0; sc->sc_intr16 = 0; 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) struct sbdsp_softc *sc; { bus_space_tag_t iot = sc->sc_iot; bus_space_handle_t ioh = sc->sc_ioh; sc->sc_intr8 = 0; sc->sc_intr16 = 0; if (sc->sc_i.run != SB_NOTRUNNING) { isa_dmaabort(sc->sc_isa, sc->sc_i.dmachan); sc->sc_i.run = SB_NOTRUNNING; } if (sc->sc_o.run != SB_NOTRUNNING) { isa_dmaabort(sc->sc_isa, sc->sc_o.dmachan); sc->sc_o.run = SB_NOTRUNNING; } /* * See SBK, section 11.3. * We pulse a reset signal into the card. * Gee, what a brilliant hardware design. */ bus_space_write_1(iot, ioh, SBP_DSP_RESET, 1); delay(10); bus_space_write_1(iot, ioh, SBP_DSP_RESET, 0); delay(30); if (sbdsp_rdsp(sc) != SB_MAGIC) return -1; return 0; } /* * Write a byte to the dsp. * 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(sc, v) struct sbdsp_softc *sc; int v; { bus_space_tag_t iot = sc->sc_iot; bus_space_handle_t ioh = sc->sc_ioh; int i; u_char x; for (i = SBDSP_NPOLL; --i >= 0; ) { x = bus_space_read_1(iot, ioh, SBP_DSP_WSTAT); delay(10); if ((x & SB_DSP_BUSY) == 0) { bus_space_write_1(iot, ioh, SBP_DSP_WRITE, v); delay(10); return 0; } } ++sberr.wdsp; return -1; } /* * Read a byte from the DSP, using polling. */ int sbdsp_rdsp(sc) struct sbdsp_softc *sc; { bus_space_tag_t iot = sc->sc_iot; bus_space_handle_t ioh = sc->sc_ioh; int i; u_char x; for (i = SBDSP_NPOLL; --i >= 0; ) { x = bus_space_read_1(iot, ioh, SBP_DSP_RSTAT); delay(10); if (x & SB_DSP_READY) { x = bus_space_read_1(iot, ioh, 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, 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, SB_DSP_SPKR_OFF); sbdsp_pause(sc); } /* * Read the version number out of the card. * Store version information in the softc. */ void sbversion(sc) struct sbdsp_softc *sc; { int v; sc->sc_model = SB_UNK; sc->sc_version = 0; if (sbdsp_wdsp(sc, SB_DSP_VERSION) < 0) return; v = sbdsp_rdsp(sc) << 8; v |= sbdsp_rdsp(sc); if (v < 0) return; sc->sc_version = v; switch(SBVER_MAJOR(v)) { case 1: sc->sc_mixer_model = SBM_NONE; sc->sc_model = SB_1; break; case 2: /* Some SB2 have a mixer, some don't. */ sbdsp_mix_write(sc, SBP_1335_MASTER_VOL, 0x04); sbdsp_mix_write(sc, SBP_1335_MIDI_VOL, 0x06); /* Check if we can read back the mixer values. */ if ((sbdsp_mix_read(sc, SBP_1335_MASTER_VOL) & 0x0e) == 0x04 && (sbdsp_mix_read(sc, SBP_1335_MIDI_VOL) & 0x0e) == 0x06) sc->sc_mixer_model = SBM_CT1335; else sc->sc_mixer_model = SBM_NONE; if (SBVER_MINOR(v) == 0) sc->sc_model = SB_20; else sc->sc_model = SB_2x; break; case 3: sc->sc_mixer_model = SBM_CT1345; sc->sc_model = SB_PRO; break; case 4: #if 0 /* XXX This does not work */ /* Most SB16 have a tone controls, but some don't. */ sbdsp_mix_write(sc, SB16P_TREBLE_L, 0x80); /* Check if we can read back the mixer value. */ if ((sbdsp_mix_read(sc, SB16P_TREBLE_L) & 0xf0) == 0x80) sc->sc_mixer_model = SBM_CT1745; else sc->sc_mixer_model = SBM_CT1XX5; #else sc->sc_mixer_model = SBM_CT1745; #endif /* XXX what about SB_32 */ if (SBVER_MINOR(v) == 16) sc->sc_model = SB_64; else sc->sc_model = SB_16; break; } } /* * Halt a DMA in progress. */ int sbdsp_haltdma(addr) void *addr; { struct sbdsp_softc *sc = addr; DPRINTF(("sbdsp_haltdma: sc=%p\n", sc)); sbdsp_reset(sc); return 0; } int sbdsp_set_timeconst(sc, tc) struct sbdsp_softc *sc; int tc; { DPRINTF(("sbdsp_set_timeconst: sc=%p tc=%d\n", sc, tc)); if (sbdsp_wdsp(sc, SB_DSP_TIMECONST) < 0 || sbdsp_wdsp(sc, tc) < 0) return EIO; return 0; } int sbdsp16_set_rate(sc, cmd, rate) struct sbdsp_softc *sc; int cmd, rate; { DPRINTF(("sbdsp16_set_rate: sc=%p cmd=0x%02x rate=%d\n", sc, cmd, rate)); if (sbdsp_wdsp(sc, cmd) < 0 || sbdsp_wdsp(sc, rate >> 8) < 0 || sbdsp_wdsp(sc, rate) < 0) return EIO; return 0; } int sbdsp_dma_init_input(addr, buf, cc) void *addr; void *buf; int cc; { struct sbdsp_softc *sc = addr; if (sc->sc_model == SB_1) return 0; sc->sc_i.run = SB_DMARUNNING; DPRINTF(("sbdsp: dma start loop input addr=%p cc=%d chan=%d\n", buf, cc, sc->sc_i.dmachan)); isa_dmastart(sc->sc_isa, sc->sc_i.dmachan, buf, cc, NULL, DMAMODE_READ | DMAMODE_LOOP, BUS_DMA_NOWAIT); return 0; } static int sbdsp_dma_setup_input(sc) struct sbdsp_softc *sc; { int stereo = sc->sc_i.modep->channels == 2; int filter; /* Initialize the PCM */ if (ISSBPRO(sc)) { if (sbdsp_wdsp(sc, sc->sc_i.modep->cmdchan) < 0) return 0; filter = stereo ? SBP_FILTER_OFF : sc->in_filter; sbdsp_mix_write(sc, SBP_INFILTER, (sbdsp_mix_read(sc, SBP_INFILTER) & ~SBP_IFILTER_MASK) | filter); } if (ISSB16CLASS(sc)) { if (sbdsp16_set_rate(sc, SB_DSP16_INPUTRATE, sc->sc_i.rate)) { DPRINTF(("sbdsp_dma_setup_input: rate=%d set failed\n", sc->sc_i.rate)); return 0; } } else { if (sbdsp_set_timeconst(sc, sc->sc_i.tc)) { DPRINTF(("sbdsp_dma_setup_input: tc=%d set failed\n", sc->sc_i.rate)); return 0; } } return 1; } int sbdsp_dma_input(addr, p, cc, intr, arg) void *addr; void *p; int cc; void (*intr) __P((void *)); void *arg; { struct sbdsp_softc *sc = addr; #ifdef AUDIO_DEBUG if (sbdspdebug > 1) printf("sbdsp_dma_input: sc=%p buf=%p cc=%d intr=%p(%p)\n", addr, p, cc, intr, arg); #endif #ifdef DIAGNOSTIC if (sc->sc_i.modep->channels == 2 && (cc & 1)) { DPRINTF(("stereo record odd bytes (%d)\n", cc)); return EIO; } #endif if (sc->sc_i.modep->precision == 8) { #ifdef DIAGNOSTIC if (sc->sc_i.dmachan != sc->sc_drq8) { printf("sbdsp_dma_input: prec=%d bad chan %d\n", sc->sc_i.modep->precision, sc->sc_i.dmachan); return EIO; } #endif sc->sc_intr8 = intr; sc->sc_arg8 = arg; } else { #ifdef DIAGNOSTIC if (sc->sc_i.dmachan != sc->sc_drq16) { printf("sbdsp_dma_input: prec=%d bad chan %d\n", sc->sc_i.modep->precision, sc->sc_i.dmachan); return EIO; } #endif sc->sc_intr16 = intr; sc->sc_arg16 = arg; } switch(sc->sc_i.run) { case SB_NOTRUNNING: /* Non-looping mode, not initialized */ sc->sc_i.run = SB_RUNNING; if (!sbdsp_dma_setup_input(sc)) goto giveup; /* fall into */ case SB_RUNNING: /* Non-looping mode, start DMA */ #ifdef AUDIO_DEBUG if (sbdspdebug > 2) printf("sbdsp_dma_input: dmastart buf=%p cc=%d chan=%d\n", p, cc, sc->sc_i.dmachan); #endif isa_dmastart(sc->sc_isa, sc->sc_i.dmachan, p, cc, NULL, DMAMODE_READ, BUS_DMA_NOWAIT); /* Start PCM in non-looping mode */ if ((sc->sc_model == SB_JAZZ && sc->sc_i.dmachan > 3) || (sc->sc_model != SB_JAZZ && sc->sc_i.modep->precision == 16)) cc >>= 1; --cc; if (sbdsp_wdsp(sc, sc->sc_i.modep->cmd) < 0 || sbdsp_wdsp(sc, cc) < 0 || sbdsp_wdsp(sc, cc >> 8) < 0) { DPRINTF(("sbdsp_dma_input: SB1 DMA start failed\n")); goto giveup; } break; case SB_DMARUNNING: /* Looping mode, not initialized */ sc->sc_i.run = SB_PCMRUNNING; if (!sbdsp_dma_setup_input(sc)) goto giveup; if ((sc->sc_model == SB_JAZZ && sc->sc_i.dmachan > 3) || (sc->sc_model != SB_JAZZ && sc->sc_i.modep->precision == 16)) cc >>= 1; --cc; /* Initialize looping PCM */ if (ISSB16CLASS(sc)) { #ifdef AUDIO_DEBUG if (sbdspdebug > 2) printf("sbdsp16 input command cmd=0x%02x bmode=0x%02x cc=%d\n", sc->sc_i.modep->cmd, sc->sc_i.bmode, cc); #endif if (sbdsp_wdsp(sc, sc->sc_i.modep->cmd) < 0 || sbdsp_wdsp(sc, sc->sc_i.bmode) < 0 || sbdsp_wdsp(sc, cc) < 0 || sbdsp_wdsp(sc, cc >> 8) < 0) { DPRINTF(("sbdsp_dma_input: SB16 DMA start failed\n")); DPRINTF(("sbdsp16 input command cmd=0x%02x bmode=0x%02x cc=%d\n", sc->sc_i.modep->cmd, sc->sc_i.bmode, cc)); goto giveup; } } else { DPRINTF(("sbdsp_dma_input: set blocksize=%d\n", cc)); if (sbdsp_wdsp(sc, SB_DSP_BLOCKSIZE) < 0 || sbdsp_wdsp(sc, cc) < 0 || sbdsp_wdsp(sc, cc >> 8) < 0) { DPRINTF(("sbdsp_dma_input: SB2 DMA blocksize failed\n")); goto giveup; } if (sbdsp_wdsp(sc, sc->sc_i.modep->cmd) < 0) { DPRINTF(("sbdsp_dma_input: SB2 DMA start failed\n")); goto giveup; } } break; case SB_PCMRUNNING: /* Looping mode, nothing to do */ break; } return 0; giveup: sbdsp_reset(sc); return EIO; } int sbdsp_dma_init_output(addr, buf, cc) void *addr; void *buf; int cc; { struct sbdsp_softc *sc = addr; if (sc->sc_model == SB_1) return 0; sc->sc_o.run = SB_DMARUNNING; DPRINTF(("sbdsp: dma start loop output buf=%p cc=%d chan=%d\n", buf, cc, sc->sc_o.dmachan)); isa_dmastart(sc->sc_isa, sc->sc_o.dmachan, buf, cc, NULL, DMAMODE_WRITE | DMAMODE_LOOP, BUS_DMA_NOWAIT); return 0; } static int sbdsp_dma_setup_output(sc) struct sbdsp_softc *sc; { int stereo = sc->sc_o.modep->channels == 2; int cmd; 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) | (stereo ? SBP_PLAYMODE_STEREO : SBP_PLAYMODE_MONO)); cmd = sc->sc_o.modep->cmdchan; if (cmd && sbdsp_wdsp(sc, cmd) < 0) return 0; } if (ISSB16CLASS(sc)) { if (sbdsp16_set_rate(sc, SB_DSP16_OUTPUTRATE, sc->sc_o.rate)) { DPRINTF(("sbdsp_dma_setup_output: rate=%d set failed\n", sc->sc_o.rate)); return 0; } } else { if (sbdsp_set_timeconst(sc, sc->sc_o.tc)) { DPRINTF(("sbdsp_dma_setup_output: tc=%d set failed\n", sc->sc_o.rate)); return 0; } } return 1; } int sbdsp_dma_output(addr, p, cc, intr, arg) void *addr; void *p; int cc; void (*intr) __P((void *)); void *arg; { struct sbdsp_softc *sc = addr; #ifdef AUDIO_DEBUG if (sbdspdebug > 1) printf("sbdsp_dma_output: sc=%p buf=%p cc=%d intr=%p(%p)\n", addr, p, cc, intr, arg); #endif #ifdef DIAGNOSTIC if (sc->sc_o.modep->channels == 2 && (cc & 1)) { DPRINTF(("stereo playback odd bytes (%d)\n", cc)); return EIO; } #endif if (sc->sc_o.modep->precision == 8) { #ifdef DIAGNOSTIC if (sc->sc_o.dmachan != sc->sc_drq8) { printf("sbdsp_dma_output: prec=%d bad chan %d\n", sc->sc_o.modep->precision, sc->sc_o.dmachan); return EIO; } #endif sc->sc_intr8 = intr; sc->sc_arg8 = arg; } else { #ifdef DIAGNOSTIC if (sc->sc_o.dmachan != sc->sc_drq16) { printf("sbdsp_dma_output: prec=%d bad chan %d\n", sc->sc_o.modep->precision, sc->sc_o.dmachan); return EIO; } #endif sc->sc_intr16 = intr; sc->sc_arg16 = arg; } switch(sc->sc_o.run) { case SB_NOTRUNNING: /* Non-looping mode, not initialized */ sc->sc_o.run = SB_RUNNING; if (!sbdsp_dma_setup_output(sc)) goto giveup; /* fall into */ case SB_RUNNING: /* Non-looping mode, initialized. Start DMA and PCM */ #ifdef AUDIO_DEBUG if (sbdspdebug > 2) printf("sbdsp: start dma out addr=%p, cc=%d, chan=%d\n", p, cc, sc->sc_o.dmachan); #endif isa_dmastart(sc->sc_isa, sc->sc_o.dmachan, p, cc, NULL, DMAMODE_WRITE, BUS_DMA_NOWAIT); if ((sc->sc_model == SB_JAZZ && sc->sc_o.dmachan > 3) || (sc->sc_model != SB_JAZZ && sc->sc_o.modep->precision == 16)) cc >>= 1; --cc; if (sbdsp_wdsp(sc, sc->sc_o.modep->cmd) < 0 || sbdsp_wdsp(sc, cc) < 0 || sbdsp_wdsp(sc, cc >> 8) < 0) { DPRINTF(("sbdsp_dma_output: SB1 DMA start failed\n")); goto giveup; } break; case SB_DMARUNNING: /* Looping mode, not initialized */ sc->sc_o.run = SB_PCMRUNNING; if (!sbdsp_dma_setup_output(sc)) goto giveup; if ((sc->sc_model == SB_JAZZ && sc->sc_o.dmachan > 3) || (sc->sc_model != SB_JAZZ && sc->sc_o.modep->precision == 16)) cc >>= 1; --cc; /* Initialize looping PCM */ if (ISSB16CLASS(sc)) { DPRINTF(("sbdsp_dma_output: SB16 cmd=0x%02x bmode=0x%02x cc=%d\n", sc->sc_o.modep->cmd,sc->sc_o.bmode, cc)); if (sbdsp_wdsp(sc, sc->sc_o.modep->cmd) < 0 || sbdsp_wdsp(sc, sc->sc_o.bmode) < 0 || sbdsp_wdsp(sc, cc) < 0 || sbdsp_wdsp(sc, cc >> 8) < 0) { DPRINTF(("sbdsp_dma_output: SB16 DMA start failed\n")); goto giveup; } } else { DPRINTF(("sbdsp_dma_output: set blocksize=%d\n", cc)); if (sbdsp_wdsp(sc, SB_DSP_BLOCKSIZE) < 0 || sbdsp_wdsp(sc, cc) < 0 || sbdsp_wdsp(sc, cc >> 8) < 0) { DPRINTF(("sbdsp_dma_output: SB2 DMA blocksize failed\n")); goto giveup; } if (sbdsp_wdsp(sc, sc->sc_o.modep->cmd) < 0) { DPRINTF(("sbdsp_dma_output: SB2 DMA start failed\n")); goto giveup; } } break; case SB_PCMRUNNING: /* Looping mode, nothing to do */ break; } 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. * * If there is interrupt sharing or a spurious interrupt occurs * there is no way to distinguish this on an SB2. So if you have * an SB2 and experience problems, buy an SB16 (it's only $40). */ int sbdsp_intr(arg) void *arg; { struct sbdsp_softc *sc = arg; int loop = sc->sc_model != SB_1; u_char irq; #ifdef AUDIO_DEBUG if (sbdspdebug > 1) printf("sbdsp_intr: intr8=%p, intr16=%p\n", sc->sc_intr8, sc->sc_intr16); #endif if (ISSB16CLASS(sc)) { irq = sbdsp_mix_read(sc, SBP_IRQ_STATUS); if ((irq & (SBP_IRQ_DMA8 | SBP_IRQ_DMA16)) == 0) { DPRINTF(("sbdsp_intr: Spurious interrupt 0x%x\n", irq)); return 0; } } else { if (!loop && !isa_dmafinished(sc->sc_isa, sc->sc_drq8)) return 0; irq = SBP_IRQ_DMA8; } sc->sc_interrupts++; delay(10); /* XXX why? */ #if 0 if (sc->sc_mintr != 0) { x = sbdsp_rdsp(sc); (*sc->sc_mintr)(sc->sc_arg, x); } else #endif if (sc->sc_intr8 == 0 && sc->sc_intr16 == 0) { DPRINTF(("sbdsp_intr: Unexpected interrupt 0x%x\n", irq)); /* XXX return 0;*/ /* Did not expect an interrupt */ } /* clear interrupt */ if (irq & SBP_IRQ_DMA8) { bus_space_read_1(sc->sc_iot, sc->sc_ioh, SBP_DSP_IRQACK8); if (!loop) isa_dmadone(sc->sc_isa, sc->sc_drq8); if (sc->sc_intr8) (*sc->sc_intr8)(sc->sc_arg8); } if (irq & SBP_IRQ_DMA16) { bus_space_read_1(sc->sc_iot, sc->sc_ioh, SBP_DSP_IRQACK16); if (sc->sc_intr16) (*sc->sc_intr16)(sc->sc_arg16); } 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, 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, v) < 0) ++sberr.wmidi; } #endif /* Mask a value 0-255, but round it first */ #define MAXVAL 256 static int sbdsp_adjust(val, mask) int val, mask; { val += (MAXVAL - mask) >> 1; if (val >= MAXVAL) val = MAXVAL-1; return val & mask; } void sbdsp_set_mixer_gain(sc, port) struct sbdsp_softc *sc; int port; { int src, gain; switch(sc->sc_mixer_model) { case SBM_NONE: return; case SBM_CT1335: gain = SB_1335_GAIN(sc->gain[port][SB_LEFT]); switch(port) { case SB_MASTER_VOL: src = SBP_1335_MASTER_VOL; break; case SB_MIDI_VOL: src = SBP_1335_MIDI_VOL; break; case SB_CD_VOL: src = SBP_1335_CD_VOL; break; case SB_VOICE_VOL: src = SBP_1335_VOICE_VOL; gain = SB_1335_MASTER_GAIN(sc->gain[port][SB_LEFT]); break; default: return; } sbdsp_mix_write(sc, src, gain); break; case SBM_CT1345: gain = SB_STEREO_GAIN(sc->gain[port][SB_LEFT], sc->gain[port][SB_RIGHT]); switch (port) { case SB_MIC_VOL: src = SBP_MIC_VOL; gain = SB_MIC_GAIN(sc->gain[port][SB_LEFT]); break; case SB_MASTER_VOL: src = SBP_MASTER_VOL; break; case SB_LINE_IN_VOL: src = SBP_LINE_VOL; break; case SB_VOICE_VOL: src = SBP_VOICE_VOL; break; case SB_MIDI_VOL: src = SBP_MIDI_VOL; break; case SB_CD_VOL: src = SBP_CD_VOL; break; default: return; } sbdsp_mix_write(sc, src, gain); break; case SBM_CT1XX5: case SBM_CT1745: switch (port) { case SB_MIC_VOL: src = SB16P_MIC_L; break; case SB_MASTER_VOL: src = SB16P_MASTER_L; break; case SB_LINE_IN_VOL: src = SB16P_LINE_L; break; case SB_VOICE_VOL: src = SB16P_VOICE_L; break; case SB_MIDI_VOL: src = SB16P_MIDI_L; break; case SB_CD_VOL: src = SB16P_CD_L; break; case SB_INPUT_GAIN: src = SB16P_INPUT_GAIN_L; break; case SB_OUTPUT_GAIN: src = SB16P_OUTPUT_GAIN_L; break; case SB_TREBLE: src = SB16P_TREBLE_L; break; case SB_BASS: src = SB16P_BASS_L; break; case SB_PCSPEAKER: sbdsp_mix_write(sc, SB16P_PCSPEAKER, sc->gain[port][SB_LEFT]); return; default: return; } sbdsp_mix_write(sc, src, sc->gain[port][SB_LEFT]); sbdsp_mix_write(sc, SB16P_L_TO_R(src), sc->gain[port][SB_RIGHT]); break; } } int sbdsp_mixer_set_port(addr, cp) void *addr; mixer_ctrl_t *cp; { struct sbdsp_softc *sc = addr; int lgain, rgain; int mask, bits; int lmask, rmask, lbits, rbits; int mute, swap; DPRINTF(("sbdsp_mixer_set_port: port=%d num_channels=%d\n", cp->dev, cp->un.value.num_channels)); if (sc->sc_mixer_model == SBM_NONE) return EINVAL; switch (cp->dev) { case SB_TREBLE: case SB_BASS: if (sc->sc_mixer_model == SBM_CT1345 || sc->sc_mixer_model == SBM_CT1XX5) { if (cp->type != AUDIO_MIXER_ENUM) return EINVAL; switch (cp->dev) { case SB_TREBLE: sbdsp_set_ifilter(addr, cp->un.ord ? SB_TREBLE : 0); return 0; case SB_BASS: sbdsp_set_ifilter(addr, cp->un.ord ? SB_BASS : 0); return 0; } } case SB_PCSPEAKER: case SB_INPUT_GAIN: case SB_OUTPUT_GAIN: if (!ISSBM1745(sc)) return EINVAL; case SB_MIC_VOL: case SB_LINE_IN_VOL: if (sc->sc_mixer_model == SBM_CT1335) return EINVAL; case SB_VOICE_VOL: case SB_MIDI_VOL: case SB_CD_VOL: 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_VOL: if (cp->un.value.num_channels != 1) return EINVAL; lgain = rgain = SB_ADJUST_MIC_GAIN(sc, cp->un.value.level[AUDIO_MIXER_LEVEL_MONO]); break; case SB_PCSPEAKER: if (cp->un.value.num_channels != 1) return EINVAL; /* fall into */ case SB_INPUT_GAIN: case SB_OUTPUT_GAIN: lgain = rgain = SB_ADJUST_2_GAIN(sc, cp->un.value.level[AUDIO_MIXER_LEVEL_MONO]); break; default: switch (cp->un.value.num_channels) { case 1: lgain = rgain = SB_ADJUST_GAIN(sc, cp->un.value.level[AUDIO_MIXER_LEVEL_MONO]); break; case 2: if (sc->sc_mixer_model == SBM_CT1335) return EINVAL; lgain = SB_ADJUST_GAIN(sc, cp->un.value.level[AUDIO_MIXER_LEVEL_LEFT]); rgain = SB_ADJUST_GAIN(sc, cp->un.value.level[AUDIO_MIXER_LEVEL_RIGHT]); break; default: return EINVAL; } break; } sc->gain[cp->dev][SB_LEFT] = lgain; sc->gain[cp->dev][SB_RIGHT] = rgain; sbdsp_set_mixer_gain(sc, cp->dev); break; case SB_RECORD_SOURCE: if (ISSBM1745(sc)) { if (cp->type != AUDIO_MIXER_SET) return EINVAL; return sbdsp_set_in_ports(sc, cp->un.mask); } else { if (cp->type != AUDIO_MIXER_ENUM) return EINVAL; sc->in_port = cp->un.ord; return sbdsp_set_in_ports(sc, 1 << cp->un.ord); } break; case SB_AGC: if (!ISSBM1745(sc) || cp->type != AUDIO_MIXER_ENUM) return EINVAL; sbdsp_mix_write(sc, SB16P_AGC, cp->un.ord & 1); break; case SB_CD_OUT_MUTE: mask = SB16P_SW_CD; goto omute; case SB_MIC_OUT_MUTE: mask = SB16P_SW_MIC; goto omute; case SB_LINE_OUT_MUTE: mask = SB16P_SW_LINE; omute: if (cp->type != AUDIO_MIXER_ENUM) return EINVAL; bits = sbdsp_mix_read(sc, SB16P_OSWITCH); sc->gain[cp->dev][SB_LR] = cp->un.ord != 0; if (cp->un.ord) bits = bits & ~mask; else bits = bits | mask; sbdsp_mix_write(sc, SB16P_OSWITCH, bits); break; case SB_MIC_IN_MUTE: case SB_MIC_SWAP: lmask = rmask = SB16P_SW_MIC; goto imute; case SB_CD_IN_MUTE: case SB_CD_SWAP: lmask = SB16P_SW_CD_L; rmask = SB16P_SW_CD_R; goto imute; case SB_LINE_IN_MUTE: case SB_LINE_SWAP: lmask = SB16P_SW_LINE_L; rmask = SB16P_SW_LINE_R; goto imute; case SB_MIDI_IN_MUTE: case SB_MIDI_SWAP: lmask = SB16P_SW_MIDI_L; rmask = SB16P_SW_MIDI_R; imute: if (cp->type != AUDIO_MIXER_ENUM) return EINVAL; mask = lmask | rmask; lbits = sbdsp_mix_read(sc, SB16P_ISWITCH_L) & ~mask; rbits = sbdsp_mix_read(sc, SB16P_ISWITCH_R) & ~mask; sc->gain[cp->dev][SB_LR] = cp->un.ord != 0; if (SB_IS_IN_MUTE(cp->dev)) { mute = cp->dev; swap = mute - SB_CD_IN_MUTE + SB_CD_SWAP; } else { swap = cp->dev; mute = swap + SB_CD_IN_MUTE - SB_CD_SWAP; } if (sc->gain[swap][SB_LR]) { mask = lmask; lmask = rmask; rmask = mask; } if (!sc->gain[mute][SB_LR]) { lbits = lbits | lmask; rbits = rbits | rmask; } sbdsp_mix_write(sc, SB16P_ISWITCH_L, lbits); sbdsp_mix_write(sc, SB16P_ISWITCH_L, rbits); break; default: return EINVAL; } return 0; } int sbdsp_mixer_get_port(addr, cp) void *addr; mixer_ctrl_t *cp; { struct sbdsp_softc *sc = addr; DPRINTF(("sbdsp_mixer_get_port: port=%d\n", cp->dev)); if (sc->sc_mixer_model == SBM_NONE) return EINVAL; switch (cp->dev) { case SB_TREBLE: case SB_BASS: if (sc->sc_mixer_model == SBM_CT1345 || sc->sc_mixer_model == SBM_CT1XX5) { switch (cp->dev) { case SB_TREBLE: cp->un.ord = sbdsp_get_ifilter(addr) == SB_TREBLE; return 0; case SB_BASS: cp->un.ord = sbdsp_get_ifilter(addr) == SB_BASS; return 0; } } case SB_PCSPEAKER: case SB_INPUT_GAIN: case SB_OUTPUT_GAIN: if (!ISSBM1745(sc)) return EINVAL; case SB_MIC_VOL: case SB_LINE_IN_VOL: if (sc->sc_mixer_model == SBM_CT1335) return EINVAL; case SB_VOICE_VOL: case SB_MIDI_VOL: case SB_CD_VOL: case SB_MASTER_VOL: switch (cp->dev) { case SB_MIC_VOL: case SB_PCSPEAKER: if (cp->un.value.num_channels != 1) return EINVAL; /* fall into */ default: switch (cp->un.value.num_channels) { case 1: cp->un.value.level[AUDIO_MIXER_LEVEL_MONO] = sc->gain[cp->dev][SB_LEFT]; break; case 2: cp->un.value.level[AUDIO_MIXER_LEVEL_LEFT] = sc->gain[cp->dev][SB_LEFT]; cp->un.value.level[AUDIO_MIXER_LEVEL_RIGHT] = sc->gain[cp->dev][SB_RIGHT]; break; default: return EINVAL; } break; } break; case SB_RECORD_SOURCE: if (ISSBM1745(sc)) cp->un.mask = sc->in_mask; else cp->un.ord = sc->in_port; break; case SB_AGC: if (!ISSBM1745(sc)) return EINVAL; cp->un.ord = sbdsp_mix_read(sc, SB16P_AGC); break; case SB_CD_IN_MUTE: case SB_MIC_IN_MUTE: case SB_LINE_IN_MUTE: case SB_MIDI_IN_MUTE: case SB_CD_SWAP: case SB_MIC_SWAP: case SB_LINE_SWAP: case SB_MIDI_SWAP: case SB_CD_OUT_MUTE: case SB_MIC_OUT_MUTE: case SB_LINE_OUT_MUTE: cp->un.ord = sc->gain[cp->dev][SB_LR]; break; default: return EINVAL; } return 0; } int sbdsp_mixer_query_devinfo(addr, dip) void *addr; mixer_devinfo_t *dip; { struct sbdsp_softc *sc = addr; int chan, class, is1745; DPRINTF(("sbdsp_mixer_query_devinfo: model=%d index=%d\n", sc->sc_mixer_model, dip->index)); if (sc->sc_mixer_model == SBM_NONE) return ENXIO; chan = sc->sc_mixer_model == SBM_CT1335 ? 1 : 2; is1745 = ISSBM1745(sc); class = is1745 ? SB_INPUT_CLASS : SB_OUTPUT_CLASS; switch (dip->index) { case SB_MASTER_VOL: dip->type = AUDIO_MIXER_VALUE; dip->mixer_class = SB_OUTPUT_CLASS; dip->prev = dip->next = AUDIO_MIXER_LAST; strcpy(dip->label.name, AudioNmaster); dip->un.v.num_channels = chan; strcpy(dip->un.v.units.name, AudioNvolume); return 0; case SB_MIDI_VOL: dip->type = AUDIO_MIXER_VALUE; dip->mixer_class = class; dip->prev = AUDIO_MIXER_LAST; dip->next = is1745 ? SB_MIDI_IN_MUTE : AUDIO_MIXER_LAST; strcpy(dip->label.name, AudioNfmsynth); dip->un.v.num_channels = chan; strcpy(dip->un.v.units.name, AudioNvolume); return 0; case SB_CD_VOL: dip->type = AUDIO_MIXER_VALUE; dip->mixer_class = class; dip->prev = AUDIO_MIXER_LAST; dip->next = is1745 ? SB_CD_IN_MUTE : AUDIO_MIXER_LAST; strcpy(dip->label.name, AudioNcd); dip->un.v.num_channels = chan; strcpy(dip->un.v.units.name, AudioNvolume); return 0; case SB_VOICE_VOL: dip->type = AUDIO_MIXER_VALUE; dip->mixer_class = class; dip->prev = AUDIO_MIXER_LAST; dip->next = AUDIO_MIXER_LAST; strcpy(dip->label.name, AudioNdac); dip->un.v.num_channels = chan; strcpy(dip->un.v.units.name, AudioNvolume); 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 (sc->sc_mixer_model == SBM_CT1335) return ENXIO; switch (dip->index) { case SB_MIC_VOL: dip->type = AUDIO_MIXER_VALUE; dip->mixer_class = class; dip->prev = AUDIO_MIXER_LAST; dip->next = is1745 ? SB_MIC_IN_MUTE : 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_LINE_IN_VOL: dip->type = AUDIO_MIXER_VALUE; dip->mixer_class = class; dip->prev = AUDIO_MIXER_LAST; dip->next = is1745 ? SB_LINE_IN_MUTE : 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_RECORD_SOURCE: dip->mixer_class = SB_RECORD_CLASS; dip->prev = dip->next = AUDIO_MIXER_LAST; strcpy(dip->label.name, AudioNsource); if (ISSBM1745(sc)) { dip->type = AUDIO_MIXER_SET; dip->un.s.num_mem = 4; strcpy(dip->un.s.member[0].label.name, AudioNmicrophone); dip->un.s.member[0].mask = 1 << SB_MIC_VOL; strcpy(dip->un.s.member[1].label.name, AudioNcd); dip->un.s.member[1].mask = 1 << SB_CD_VOL; strcpy(dip->un.s.member[2].label.name, AudioNline); dip->un.s.member[2].mask = 1 << SB_LINE_IN_VOL; strcpy(dip->un.s.member[3].label.name, AudioNfmsynth); dip->un.s.member[3].mask = 1 << SB_MIDI_VOL; } else { dip->type = AUDIO_MIXER_ENUM; dip->un.e.num_mem = 3; strcpy(dip->un.e.member[0].label.name, AudioNmicrophone); dip->un.e.member[0].ord = SB_MIC_VOL; strcpy(dip->un.e.member[1].label.name, AudioNcd); dip->un.e.member[1].ord = SB_CD_VOL; strcpy(dip->un.e.member[2].label.name, AudioNline); dip->un.e.member[2].ord = SB_LINE_IN_VOL; } return 0; case SB_BASS: dip->prev = dip->next = AUDIO_MIXER_LAST; strcpy(dip->label.name, AudioNbass); if (sc->sc_mixer_model == SBM_CT1745) { dip->type = AUDIO_MIXER_VALUE; dip->mixer_class = SB_EQUALIZATION_CLASS; dip->un.v.num_channels = 2; strcpy(dip->un.v.units.name, AudioNbass); } else { dip->type = AUDIO_MIXER_ENUM; dip->mixer_class = SB_INPUT_CLASS; 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->prev = dip->next = AUDIO_MIXER_LAST; strcpy(dip->label.name, AudioNtreble); if (sc->sc_mixer_model == SBM_CT1745) { dip->type = AUDIO_MIXER_VALUE; dip->mixer_class = SB_EQUALIZATION_CLASS; dip->un.v.num_channels = 2; strcpy(dip->un.v.units.name, AudioNtreble); } else { dip->type = AUDIO_MIXER_ENUM; dip->mixer_class = SB_INPUT_CLASS; 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; 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; } if (sc->sc_mixer_model == SBM_CT1345) return ENXIO; switch(dip->index) { case SB_PCSPEAKER: dip->type = AUDIO_MIXER_VALUE; dip->mixer_class = SB_INPUT_CLASS; dip->prev = dip->next = AUDIO_MIXER_LAST; strcpy(dip->label.name, "pc_speaker"); dip->un.v.num_channels = 1; strcpy(dip->un.v.units.name, AudioNvolume); return 0; case SB_INPUT_GAIN: dip->type = AUDIO_MIXER_VALUE; dip->mixer_class = SB_INPUT_CLASS; dip->prev = dip->next = AUDIO_MIXER_LAST; strcpy(dip->label.name, AudioNinput); dip->un.v.num_channels = 2; strcpy(dip->un.v.units.name, AudioNvolume); return 0; case SB_OUTPUT_GAIN: dip->type = AUDIO_MIXER_VALUE; dip->mixer_class = SB_OUTPUT_CLASS; dip->prev = dip->next = AUDIO_MIXER_LAST; strcpy(dip->label.name, AudioNoutput); dip->un.v.num_channels = 2; strcpy(dip->un.v.units.name, AudioNvolume); return 0; case SB_AGC: dip->type = AUDIO_MIXER_ENUM; dip->mixer_class = SB_INPUT_CLASS; dip->prev = dip->next = AUDIO_MIXER_LAST; strcpy(dip->label.name, "AGC"); 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_EQUALIZATION_CLASS: dip->type = AUDIO_MIXER_CLASS; dip->mixer_class = SB_EQUALIZATION_CLASS; dip->next = dip->prev = AUDIO_MIXER_LAST; strcpy(dip->label.name, AudioCequalization); return 0; case SB_CD_IN_MUTE: dip->prev = SB_CD_VOL; dip->next = SB_CD_SWAP; dip->mixer_class = SB_INPUT_CLASS; goto mute; case SB_MIC_IN_MUTE: dip->prev = SB_MIC_VOL; dip->next = SB_MIC_SWAP; dip->mixer_class = SB_INPUT_CLASS; goto mute; case SB_LINE_IN_MUTE: dip->prev = SB_LINE_IN_VOL; dip->next = SB_LINE_SWAP; dip->mixer_class = SB_INPUT_CLASS; goto mute; case SB_MIDI_IN_MUTE: dip->prev = SB_MIDI_VOL; dip->next = SB_MIDI_SWAP; dip->mixer_class = SB_INPUT_CLASS; goto mute; case SB_CD_SWAP: dip->prev = SB_CD_IN_MUTE; dip->next = SB_CD_OUT_MUTE; goto swap; case SB_MIC_SWAP: dip->prev = SB_MIC_IN_MUTE; dip->next = SB_MIC_OUT_MUTE; goto swap; case SB_LINE_SWAP: dip->prev = SB_LINE_IN_MUTE; dip->next = SB_LINE_OUT_MUTE; goto swap; case SB_MIDI_SWAP: dip->prev = SB_MIDI_IN_MUTE; dip->next = AUDIO_MIXER_LAST; swap: dip->mixer_class = SB_INPUT_CLASS; strcpy(dip->label.name, AudioNswap); goto mute1; case SB_CD_OUT_MUTE: dip->prev = SB_CD_SWAP; dip->next = AUDIO_MIXER_LAST; dip->mixer_class = SB_OUTPUT_CLASS; goto mute; case SB_MIC_OUT_MUTE: dip->prev = SB_MIC_SWAP; dip->next = AUDIO_MIXER_LAST; dip->mixer_class = SB_OUTPUT_CLASS; goto mute; case SB_LINE_OUT_MUTE: dip->prev = SB_LINE_SWAP; dip->next = AUDIO_MIXER_LAST; dip->mixer_class = SB_OUTPUT_CLASS; mute: strcpy(dip->label.name, AudioNmute); mute1: dip->type = AUDIO_MIXER_ENUM; 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; } return ENXIO; } void * sb_malloc(addr, size, pool, flags) void *addr; unsigned long size; int pool; int flags; { struct sbdsp_softc *sc = addr; return isa_malloc(sc->sc_isa, 4, size, pool, flags); } void sb_free(addr, ptr, pool) void *addr; void *ptr; int pool; { isa_free(ptr, pool); } unsigned long sb_round(addr, size) void *addr; unsigned long size; { if (size > MAX_ISADMA) size = MAX_ISADMA; return size; } int sb_mappage(addr, mem, off, prot) void *addr; void *mem; int off; int prot; { return isa_mappage(mem, off, prot); } int sbdsp_get_props(addr) void *addr; { struct sbdsp_softc *sc = addr; return AUDIO_PROP_MMAP | (sc->sc_fullduplex ? AUDIO_PROP_FULLDUPLEX : 0); }