/* $NetBSD: sbdsp.c,v 1.117 2005/02/27 00:27:17 perry Exp $ */ /*- * Copyright (c) 1999 The NetBSD Foundation, Inc. * All rights reserved. * * This code is derived from software contributed to The NetBSD Foundation * by Charles M. Hannum. * * 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 NetBSD * Foundation, Inc. and its contributors. * 4. Neither the name of The NetBSD Foundation nor the names of its * contributors may be used to endorse or promote products derived * from this software without specific prior written permission. * * THIS SOFTWARE IS PROVIDED BY THE NETBSD FOUNDATION, INC. 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 FOUNDATION 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. */ /* * 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 __KERNEL_RCSID(0, "$NetBSD: sbdsp.c,v 1.117 2005/02/27 00:27:17 perry Exp $"); #include "midi.h" #include "mpu.h" #include #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 #define DPRINTFN(n,x) if (sbdspdebug >= (n)) printf x int sbdspdebug = 0; #else #define DPRINTF(x) #define DPRINTFN(n,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 halt, cont; u_char cmdchan; }; static struct sbmode sbpmodes[] = { { SB_1, 1, 8, 4000,22727,SB_DSP_WDMA ,SB_DSP_HALT ,SB_DSP_CONT }, { SB_20, 1, 8, 4000,22727,SB_DSP_WDMA_LOOP,SB_DSP_HALT ,SB_DSP_CONT }, { SB_2x, 1, 8,22727,45454,SB_DSP_HS_OUTPUT,SB_DSP_HALT ,SB_DSP_CONT }, { SB_2x, 1, 8, 4000,22727,SB_DSP_WDMA_LOOP,SB_DSP_HALT ,SB_DSP_CONT }, { SB_PRO, 1, 8,22727,45454,SB_DSP_HS_OUTPUT,SB_DSP_HALT ,SB_DSP_CONT }, { SB_PRO, 1, 8, 4000,22727,SB_DSP_WDMA_LOOP,SB_DSP_HALT ,SB_DSP_CONT }, { SB_PRO, 2, 8,11025,22727,SB_DSP_HS_OUTPUT,SB_DSP_HALT ,SB_DSP_CONT }, /* Yes, we write the record mode to set 16-bit playback mode. weird, huh? */ { SB_JAZZ,1, 8,22727,45454,SB_DSP_HS_OUTPUT,SB_DSP_HALT ,SB_DSP_CONT ,SB_DSP_RECORD_MONO }, { SB_JAZZ,1, 8, 4000,22727,SB_DSP_WDMA_LOOP,SB_DSP_HALT ,SB_DSP_CONT ,SB_DSP_RECORD_MONO }, { SB_JAZZ,2, 8,11025,22727,SB_DSP_HS_OUTPUT,SB_DSP_HALT ,SB_DSP_CONT ,SB_DSP_RECORD_STEREO }, { SB_JAZZ,1,16,22727,45454,SB_DSP_HS_OUTPUT,SB_DSP_HALT ,SB_DSP_CONT ,JAZZ16_RECORD_MONO }, { SB_JAZZ,1,16, 4000,22727,SB_DSP_WDMA_LOOP,SB_DSP_HALT ,SB_DSP_CONT ,JAZZ16_RECORD_MONO }, { SB_JAZZ,2,16,11025,22727,SB_DSP_HS_OUTPUT,SB_DSP_HALT ,SB_DSP_CONT ,JAZZ16_RECORD_STEREO }, { SB_16, 1, 8, 5000,49000,SB_DSP16_WDMA_8 ,SB_DSP_HALT ,SB_DSP_CONT }, { SB_16, 2, 8, 5000,49000,SB_DSP16_WDMA_8 ,SB_DSP_HALT ,SB_DSP_CONT }, #define PLAY16 15 /* must be the index of the next entry in the table */ { SB_16, 1,16, 5000,49000,SB_DSP16_WDMA_16,SB_DSP16_HALT,SB_DSP16_CONT}, { SB_16, 2,16, 5000,49000,SB_DSP16_WDMA_16,SB_DSP16_HALT,SB_DSP16_CONT}, { -1 } }; static struct sbmode sbrmodes[] = { { SB_1, 1, 8, 4000,12987,SB_DSP_RDMA ,SB_DSP_HALT ,SB_DSP_CONT }, { SB_20, 1, 8, 4000,12987,SB_DSP_RDMA_LOOP,SB_DSP_HALT ,SB_DSP_CONT }, { SB_2x, 1, 8,12987,14925,SB_DSP_HS_INPUT ,SB_DSP_HALT ,SB_DSP_CONT }, { SB_2x, 1, 8, 4000,12987,SB_DSP_RDMA_LOOP,SB_DSP_HALT ,SB_DSP_CONT }, { SB_PRO, 1, 8,22727,45454,SB_DSP_HS_INPUT ,SB_DSP_HALT ,SB_DSP_CONT ,SB_DSP_RECORD_MONO }, { SB_PRO, 1, 8, 4000,22727,SB_DSP_RDMA_LOOP,SB_DSP_HALT ,SB_DSP_CONT ,SB_DSP_RECORD_MONO }, { SB_PRO, 2, 8,11025,22727,SB_DSP_HS_INPUT ,SB_DSP_HALT ,SB_DSP_CONT ,SB_DSP_RECORD_STEREO }, { SB_JAZZ,1, 8,22727,45454,SB_DSP_HS_INPUT ,SB_DSP_HALT ,SB_DSP_CONT ,SB_DSP_RECORD_MONO }, { SB_JAZZ,1, 8, 4000,22727,SB_DSP_RDMA_LOOP,SB_DSP_HALT ,SB_DSP_CONT ,SB_DSP_RECORD_MONO }, { SB_JAZZ,2, 8,11025,22727,SB_DSP_HS_INPUT ,SB_DSP_HALT ,SB_DSP_CONT ,SB_DSP_RECORD_STEREO }, { SB_JAZZ,1,16,22727,45454,SB_DSP_HS_INPUT ,SB_DSP_HALT ,SB_DSP_CONT ,JAZZ16_RECORD_MONO }, { SB_JAZZ,1,16, 4000,22727,SB_DSP_RDMA_LOOP,SB_DSP_HALT ,SB_DSP_CONT ,JAZZ16_RECORD_MONO }, { SB_JAZZ,2,16,11025,22727,SB_DSP_HS_INPUT ,SB_DSP_HALT ,SB_DSP_CONT ,JAZZ16_RECORD_STEREO }, { SB_16, 1, 8, 5000,49000,SB_DSP16_RDMA_8 ,SB_DSP_HALT ,SB_DSP_CONT }, { SB_16, 2, 8, 5000,49000,SB_DSP16_RDMA_8 ,SB_DSP_HALT ,SB_DSP_CONT }, { SB_16, 1,16, 5000,49000,SB_DSP16_RDMA_16,SB_DSP16_HALT,SB_DSP16_CONT}, { SB_16, 2,16, 5000,49000,SB_DSP16_RDMA_16,SB_DSP16_HALT,SB_DSP16_CONT}, { -1 } }; void sbversion(struct sbdsp_softc *); void sbdsp_jazz16_probe(struct sbdsp_softc *); void sbdsp_set_mixer_gain(struct sbdsp_softc *, int); void sbdsp_pause(struct sbdsp_softc *); int sbdsp_set_timeconst(struct sbdsp_softc *, int); int sbdsp16_set_rate(struct sbdsp_softc *, int, int); int sbdsp_set_in_ports(struct sbdsp_softc *, int); void sbdsp_set_ifilter(void *, int); int sbdsp_get_ifilter(void *); int sbdsp_block_output(void *); int sbdsp_block_input(void *); static int sbdsp_adjust(int, int); int sbdsp_midi_intr(void *); static void sbdsp_powerhook(int, void*); #ifdef AUDIO_DEBUG void sb_printsc(struct sbdsp_softc *); void sb_printsc(struct sbdsp_softc *sc) { int i; printf("open %d DMA chan %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 intr16 %p\n", sc->sc_intr8, sc->sc_intr16); 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(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(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; bus_space_handle_t ioh; iot = sc->sc_iot; 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(struct sbdsp_softc *sc) { int i, error; u_int v; 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 / 2); 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; if (sc->sc_drq8 != -1) { sc->sc_drq8_maxsize = isa_dmamaxsize(sc->sc_ic, sc->sc_drq8); error = isa_dmamap_create(sc->sc_ic, sc->sc_drq8, sc->sc_drq8_maxsize, BUS_DMA_NOWAIT|BUS_DMA_ALLOCNOW); if (error) { 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) { sc->sc_drq16_maxsize = isa_dmamaxsize(sc->sc_ic, sc->sc_drq16); error = isa_dmamap_create(sc->sc_ic, sc->sc_drq16, sc->sc_drq16_maxsize, BUS_DMA_NOWAIT|BUS_DMA_ALLOCNOW); if (error) { printf("%s: can't create map for drq %d\n", sc->sc_dev.dv_xname, sc->sc_drq16); isa_dmamap_destroy(sc->sc_ic, sc->sc_drq8); return; } } powerhook_establish (sbdsp_powerhook, sc); } static void sbdsp_powerhook(int why, void *arg) { struct sbdsp_softc *sc; int i; sc = arg; if (!sc || why != PWR_RESUME) return; /* Reset the mixer. */ sbdsp_mix_write(sc, SBP_MIX_RESET, SBP_MIX_RESET); for (i = 0; i < SB_NDEVS; i++) sbdsp_set_mixer_gain (sc, i); } void sbdsp_mix_write(struct sbdsp_softc *sc, int mixerport, int val) { bus_space_tag_t iot; bus_space_handle_t ioh; int s; iot = sc->sc_iot; ioh = sc->sc_ioh; 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(struct sbdsp_softc *sc, int mixerport) { bus_space_tag_t iot; bus_space_handle_t ioh; int val; int s; iot = sc->sc_iot; ioh = sc->sc_ioh; 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(void *addr, struct audio_encoding *fp) { struct sbdsp_softc *sc; int emul; sc = addr; 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( void *addr, int setmode, int usemode, audio_params_t *play, audio_params_t *rec, stream_filter_list_t *pfil, stream_filter_list_t *rfil) { struct sbdsp_softc *sc; struct sbmode *m; u_int rate, tc, bmode; stream_filter_factory_t *swcode; int model; int chan; struct audio_params *p; audio_params_t hw; stream_filter_list_t *fil; int mode; sc = addr; if (sc->sc_open == SB_OPEN_MIDI) return EBUSY; /* Later models work like SB16. */ model = min(sc->sc_model, SB_16); /* * Prior to the SB16, we have only one clock, so make the sample * rates match. */ if (!ISSB16CLASS(sc) && play->sample_rate != rec->sample_rate && usemode == (AUMODE_PLAY | AUMODE_RECORD)) { if (setmode == AUMODE_PLAY) { rec->sample_rate = play->sample_rate; setmode |= AUMODE_RECORD; } else if (setmode == AUMODE_RECORD) { play->sample_rate = rec->sample_rate; setmode |= AUMODE_PLAY; } else return EINVAL; } /* 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 = NULL; fil = mode == AUMODE_PLAY ? pfil : rfil; hw = *p; tc = 1; bmode = -1; if (model == SB_16) { switch (p->encoding) { case AUDIO_ENCODING_SLINEAR_BE: if (p->precision == 16) { hw.encoding = AUDIO_ENCODING_SLINEAR_LE; swcode = swap_bytes; } /* fall into */ case AUDIO_ENCODING_SLINEAR_LE: bmode = SB_BMODE_SIGNED; break; case AUDIO_ENCODING_ULINEAR_BE: if (p->precision == 16) { hw.encoding = AUDIO_ENCODING_ULINEAR_LE; swcode = swap_bytes; } /* fall into */ case AUDIO_ENCODING_ULINEAR_LE: bmode = SB_BMODE_UNSIGNED; break; case AUDIO_ENCODING_ULAW: hw.encoding = AUDIO_ENCODING_ULINEAR_LE; if (mode == AUMODE_PLAY) { hw.precision = hw.validbits = 16; swcode = mulaw_to_linear16; m = &sbpmodes[PLAY16]; } else swcode = linear8_to_mulaw; bmode = SB_BMODE_UNSIGNED; break; case AUDIO_ENCODING_ALAW: hw.encoding = AUDIO_ENCODING_ULINEAR_LE; if (mode == AUMODE_PLAY) { hw.precision = hw.validbits = 16; swcode = alaw_to_linear16; m = &sbpmodes[PLAY16]; } else swcode = linear8_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: hw.encoding = AUDIO_ENCODING_SLINEAR_LE; swcode = change_sign16; break; case AUDIO_ENCODING_SLINEAR_BE: hw.encoding = AUDIO_ENCODING_SLINEAR_LE; swcode = swap_bytes; break; case AUDIO_ENCODING_ULINEAR_BE: hw.encoding = AUDIO_ENCODING_SLINEAR_LE; swcode = swap_bytes_change_sign16; break; case AUDIO_ENCODING_ULAW: hw.encoding = AUDIO_ENCODING_ULINEAR_LE; swcode = mode == AUMODE_PLAY ? mulaw_to_linear8 : linear8_to_mulaw; break; case AUDIO_ENCODING_ALAW: hw.encoding = AUDIO_ENCODING_ULINEAR_LE; swcode = mode == AUMODE_PLAY ? alaw_to_linear8 : linear8_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; hw.sample_rate = p->sample_rate; } else { switch (p->encoding) { case AUDIO_ENCODING_SLINEAR_BE: case AUDIO_ENCODING_SLINEAR_LE: hw.encoding = AUDIO_ENCODING_ULINEAR_LE; swcode = change_sign8; break; case AUDIO_ENCODING_ULINEAR_BE: case AUDIO_ENCODING_ULINEAR_LE: break; case AUDIO_ENCODING_ULAW: hw.encoding = AUDIO_ENCODING_ULINEAR_LE; swcode = mode == AUMODE_PLAY ? mulaw_to_linear8 : linear8_to_mulaw; break; case AUDIO_ENCODING_ALAW: hw.encoding = AUDIO_ENCODING_ULINEAR_LE; swcode = mode == AUMODE_PLAY ? alaw_to_linear8 : linear8_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; hw.sample_rate = p->sample_rate; } 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; } if (swcode != NULL) fil->append(fil, swcode, &hw); DPRINTF(("sbdsp_set_params: model=%d, mode=%d, rate=%u, " "prec=%d, chan=%d, enc=%d -> tc=%02x, cmd=%02x, " "bmode=%02x, cmdchan=%02x\n", sc->sc_model, mode, p->sample_rate, p->precision, p->channels, p->encoding, tc, m->cmd, bmode, m->cmdchan)); } if (sc->sc_fullduplex && usemode == (AUMODE_PLAY | AUMODE_RECORD) && sc->sc_i.dmachan == sc->sc_o.dmachan) { DPRINTF(("sbdsp_set_params: 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. */ hw.precision = hw.validbits = 16; pfil->append(pfil, linear8_to_linear16, &hw); 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(void *addr, int which) { struct sbdsp_softc *sc; int mixval; sc = addr; 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(void *addr) { struct sbdsp_softc *sc; 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(struct sbdsp_softc *sc, int mask) { int bitsl, bitsr; int sbport; if (sc->sc_open == SB_OPEN_MIDI) return EBUSY; 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(void *addr, int newstate) { struct sbdsp_softc *sc; sc = addr; if (sc->sc_open == SB_OPEN_MIDI) return EBUSY; 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( void *addr, int blk, int mode, const audio_params_t *param) { return blk & -4; /* round to biggest sample size */ } int sbdsp_open(void *addr, int flags) { struct sbdsp_softc *sc; int error, state; sc = addr; DPRINTF(("sbdsp_open: sc=%p\n", sc)); if (sc->sc_open != SB_CLOSED) return EBUSY; sc->sc_open = SB_OPEN_AUDIO; state = 0; if (sc->sc_drq8 != -1) { error = isa_drq_alloc(sc->sc_ic, sc->sc_drq8); if (error != 0) goto bad; state |= 1; } if (sc->sc_drq16 != -1 && sc->sc_drq16 != sc->sc_drq8) { error = isa_drq_alloc(sc->sc_ic, sc->sc_drq16); if (error != 0) goto bad; state |= 2; } if (sbdsp_reset(sc) != 0) { error = EIO; goto bad; } 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; bad: if (state & 1) isa_drq_free(sc->sc_ic, sc->sc_drq8); if (state & 2) isa_drq_free(sc->sc_ic, sc->sc_drq16); sc->sc_open = SB_CLOSED; return error; } void sbdsp_close(void *addr) { struct sbdsp_softc *sc; sc = addr; DPRINTF(("sbdsp_close: sc=%p\n", sc)); sbdsp_spkroff(sc); sc->spkr_state = SPKR_OFF; sc->sc_intr8 = 0; sc->sc_intr16 = 0; if (sc->sc_drq8 != -1) isa_drq_free(sc->sc_ic, sc->sc_drq8); if (sc->sc_drq16 != -1 && sc->sc_drq16 != sc->sc_drq8) isa_drq_free(sc->sc_ic, sc->sc_drq16); sc->sc_open = SB_CLOSED; DPRINTF(("sbdsp_close: closed\n")); } /* * Lower-level routines */ /* * Reset the card. * Return non-zero if the card isn't detected. */ int sbdsp_reset(struct sbdsp_softc *sc) { bus_space_tag_t iot; bus_space_handle_t ioh; iot = sc->sc_iot; ioh = sc->sc_ioh; sc->sc_intr8 = 0; sc->sc_intr16 = 0; sc->sc_intrm = 0; /* * 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(struct sbdsp_softc *sc, int v) { bus_space_tag_t iot; bus_space_handle_t ioh; int i; u_char x; iot = sc->sc_iot; ioh = sc->sc_ioh; 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(struct sbdsp_softc *sc) { bus_space_tag_t iot; bus_space_handle_t ioh; int i; u_char x; iot = sc->sc_iot; ioh = sc->sc_ioh; 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; } void sbdsp_pause(struct sbdsp_softc *sc) { (void) tsleep(sbdsp_pause, PWAIT, "sbpause", hz / 8); } /* * 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(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(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(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 #if 0 /* XXX figure out a good way of determining the model */ /* XXX what about SB_32 */ if (SBVER_MINOR(v) == 16) sc->sc_model = SB_64; else #endif sc->sc_model = SB_16; break; } } int sbdsp_set_timeconst(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(struct sbdsp_softc *sc, int cmd, int 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_trigger_input( void *addr, void *start, void *end, int blksize, void (*intr)(void *), void *arg, const audio_params_t *param) { struct sbdsp_softc *sc; int stereo; int width; int filter; sc = addr; stereo = param->channels == 2; width = param->precision; #ifdef DIAGNOSTIC if (stereo && (blksize & 1)) { DPRINTF(("stereo record odd bytes (%d)\n", blksize)); return EIO; } if (sc->sc_i.run != SB_NOTRUNNING) printf("sbdsp_trigger_input: already running\n"); #endif sc->sc_intrr = intr; sc->sc_argr = arg; if (width == 8) { #ifdef DIAGNOSTIC if (sc->sc_i.dmachan != sc->sc_drq8) { printf("sbdsp_trigger_input: width=%d bad chan %d\n", width, sc->sc_i.dmachan); return EIO; } #endif sc->sc_intr8 = sbdsp_block_input; } else { #ifdef DIAGNOSTIC if (sc->sc_i.dmachan != sc->sc_drq16) { printf("sbdsp_trigger_input: width=%d bad chan %d\n", width, sc->sc_i.dmachan); return EIO; } #endif sc->sc_intr16 = sbdsp_block_input; } if ((sc->sc_model == SB_JAZZ) ? (sc->sc_i.dmachan > 3) : (width == 16)) blksize >>= 1; --blksize; sc->sc_i.blksize = blksize; if (ISSBPRO(sc)) { if (sbdsp_wdsp(sc, sc->sc_i.modep->cmdchan) < 0) return EIO; 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_trigger_input: rate=%d set failed\n", sc->sc_i.rate)); return EIO; } } else { if (sbdsp_set_timeconst(sc, sc->sc_i.tc)) { DPRINTF(("sbdsp_trigger_input: tc=%d set failed\n", sc->sc_i.rate)); return EIO; } } DPRINTF(("sbdsp: DMA start loop input start=%p end=%p chan=%d\n", start, end, sc->sc_i.dmachan)); isa_dmastart(sc->sc_ic, sc->sc_i.dmachan, start, (char *)end - (char *)start, NULL, DMAMODE_READ | DMAMODE_LOOPDEMAND, BUS_DMA_NOWAIT); return sbdsp_block_input(addr); } int sbdsp_block_input(void *addr) { struct sbdsp_softc *sc; int cc; sc = addr; cc = sc->sc_i.blksize; DPRINTFN(2, ("sbdsp_block_input: sc=%p cc=%d\n", addr, cc)); if (sc->sc_i.run != SB_NOTRUNNING) sc->sc_intrr(sc->sc_argr); if (sc->sc_model == SB_1) { /* Non-looping mode, start DMA */ if (sbdsp_wdsp(sc, sc->sc_i.modep->cmd) < 0 || sbdsp_wdsp(sc, cc) < 0 || sbdsp_wdsp(sc, cc >> 8) < 0) { DPRINTF(("sbdsp_block_input: SB1 DMA start failed\n")); return EIO; } sc->sc_i.run = SB_RUNNING; } else if (sc->sc_i.run == SB_NOTRUNNING) { /* Initialize looping PCM */ if (ISSB16CLASS(sc)) { DPRINTFN(3, ("sbdsp16 input command cmd=0x%02x bmode=0x%02x cc=%d\n", sc->sc_i.modep->cmd, sc->sc_i.bmode, cc)); 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_block_input: SB16 DMA start failed\n")); return EIO; } } else { DPRINTF(("sbdsp_block_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_block_input: SB2 DMA blocksize failed\n")); return EIO; } if (sbdsp_wdsp(sc, sc->sc_i.modep->cmd) < 0) { DPRINTF(("sbdsp_block_input: SB2 DMA start failed\n")); return EIO; } } sc->sc_i.run = SB_LOOPING; } return 0; } int sbdsp_trigger_output( void *addr, void *start, void *end, int blksize, void (*intr)(void *), void *arg, const audio_params_t *param) { struct sbdsp_softc *sc; int stereo; int width; int cmd; sc = addr; stereo = param->channels == 2; width = param->precision; #ifdef DIAGNOSTIC if (stereo && (blksize & 1)) { DPRINTF(("stereo playback odd bytes (%d)\n", blksize)); return EIO; } if (sc->sc_o.run != SB_NOTRUNNING) printf("sbdsp_trigger_output: already running\n"); #endif sc->sc_intrp = intr; sc->sc_argp = arg; if (width == 8) { #ifdef DIAGNOSTIC if (sc->sc_o.dmachan != sc->sc_drq8) { printf("sbdsp_trigger_output: width=%d bad chan %d\n", width, sc->sc_o.dmachan); return EIO; } #endif sc->sc_intr8 = sbdsp_block_output; } else { #ifdef DIAGNOSTIC if (sc->sc_o.dmachan != sc->sc_drq16) { printf("sbdsp_trigger_output: width=%d bad chan %d\n", width, sc->sc_o.dmachan); return EIO; } #endif sc->sc_intr16 = sbdsp_block_output; } if ((sc->sc_model == SB_JAZZ) ? (sc->sc_o.dmachan > 3) : (width == 16)) blksize >>= 1; --blksize; sc->sc_o.blksize = blksize; 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 EIO; } if (ISSB16CLASS(sc)) { if (sbdsp16_set_rate(sc, SB_DSP16_OUTPUTRATE, sc->sc_o.rate)) { DPRINTF(("sbdsp_trigger_output: rate=%d set failed\n", sc->sc_o.rate)); return EIO; } } else { if (sbdsp_set_timeconst(sc, sc->sc_o.tc)) { DPRINTF(("sbdsp_trigger_output: tc=%d set failed\n", sc->sc_o.rate)); return EIO; } } DPRINTF(("sbdsp: DMA start loop output start=%p end=%p chan=%d\n", start, end, sc->sc_o.dmachan)); isa_dmastart(sc->sc_ic, sc->sc_o.dmachan, start, (char *)end - (char *)start, NULL, DMAMODE_WRITE | DMAMODE_LOOPDEMAND, BUS_DMA_NOWAIT); return sbdsp_block_output(addr); } int sbdsp_block_output(void *addr) { struct sbdsp_softc *sc; int cc; sc = addr; cc = sc->sc_o.blksize; DPRINTFN(2, ("sbdsp_block_output: sc=%p cc=%d\n", addr, cc)); if (sc->sc_o.run != SB_NOTRUNNING) sc->sc_intrp(sc->sc_argp); if (sc->sc_model == SB_1) { /* Non-looping mode, initialized. Start DMA and PCM */ if (sbdsp_wdsp(sc, sc->sc_o.modep->cmd) < 0 || sbdsp_wdsp(sc, cc) < 0 || sbdsp_wdsp(sc, cc >> 8) < 0) { DPRINTF(("sbdsp_block_output: SB1 DMA start failed\n")); return EIO; } sc->sc_o.run = SB_RUNNING; } else if (sc->sc_o.run == SB_NOTRUNNING) { /* Initialize looping PCM */ if (ISSB16CLASS(sc)) { DPRINTF(("sbdsp_block_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_block_output: SB16 DMA start failed\n")); return EIO; } } else { DPRINTF(("sbdsp_block_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_block_output: SB2 DMA blocksize failed\n")); return EIO; } if (sbdsp_wdsp(sc, sc->sc_o.modep->cmd) < 0) { DPRINTF(("sbdsp_block_output: SB2 DMA start failed\n")); return EIO; } } sc->sc_o.run = SB_LOOPING; } return 0; } int sbdsp_halt_output(void *addr) { struct sbdsp_softc *sc; sc = addr; if (sc->sc_o.run != SB_NOTRUNNING) { if (sbdsp_wdsp(sc, sc->sc_o.modep->halt) < 0) printf("sbdsp_halt_output: failed to halt\n"); isa_dmaabort(sc->sc_ic, sc->sc_o.dmachan); sc->sc_o.run = SB_NOTRUNNING; } return 0; } int sbdsp_halt_input(void *addr) { struct sbdsp_softc *sc; sc = addr; if (sc->sc_i.run != SB_NOTRUNNING) { if (sbdsp_wdsp(sc, sc->sc_i.modep->halt) < 0) printf("sbdsp_halt_input: failed to halt\n"); isa_dmaabort(sc->sc_ic, sc->sc_i.dmachan); sc->sc_i.run = SB_NOTRUNNING; } return 0; } /* * 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(void *arg) { struct sbdsp_softc *sc; u_char irq; sc = arg; DPRINTFN(2, ("sbdsp_intr: intr8=%p, intr16=%p\n", sc->sc_intr8, sc->sc_intr16)); if (ISSB16CLASS(sc)) { irq = sbdsp_mix_read(sc, SBP_IRQ_STATUS); if ((irq & (SBP_IRQ_DMA8 | SBP_IRQ_DMA16 | SBP_IRQ_MPU401)) == 0) { DPRINTF(("sbdsp_intr: Spurious interrupt 0x%x\n", irq)); return 0; } } else { /* XXXX CHECK FOR INTERRUPT */ irq = SBP_IRQ_DMA8; } sc->sc_interrupts++; delay(10); /* XXX why? */ /* clear interrupt */ if (irq & SBP_IRQ_DMA8) { bus_space_read_1(sc->sc_iot, sc->sc_ioh, SBP_DSP_IRQACK8); if (sc->sc_intr8) sc->sc_intr8(arg); } 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(arg); } #if NMPU > 0 if ((irq & SBP_IRQ_MPU401) && sc->sc_mpudev) { mpu_intr(sc->sc_mpudev); } #endif return 1; } /* Like val & mask, but make sure the result is correctly rounded. */ #define MAXVAL 256 static int sbdsp_adjust(int val, int mask) { val += (MAXVAL - mask) >> 1; if (val >= MAXVAL) val = MAXVAL-1; return val & mask; } void sbdsp_set_mixer_gain(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(void *addr, mixer_ctrl_t *cp) { struct sbdsp_softc *sc; int lgain, rgain; int mask, bits; int lmask, rmask, lbits, rbits; int mute, swap; sc = addr; if (sc->sc_open == SB_OPEN_MIDI) return EBUSY; 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(void *addr, mixer_ctrl_t *cp) { struct sbdsp_softc *sc; sc = addr; if (sc->sc_open == SB_OPEN_MIDI) return EBUSY; 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(void *addr, mixer_devinfo_t *dip) { struct sbdsp_softc *sc = addr; int chan, class, is1745; sc = addr; 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(void *addr, int direction, size_t size, struct malloc_type *pool, int flags) { struct sbdsp_softc *sc; int drq; sc = addr; if (sc->sc_drq8 != -1) drq = sc->sc_drq8; else drq = sc->sc_drq16; return isa_malloc(sc->sc_ic, drq, size, pool, flags); } void sb_free(void *addr, void *ptr, struct malloc_type *pool) { isa_free(ptr, pool); } size_t sb_round_buffersize(void *addr, int direction, size_t size) { struct sbdsp_softc *sc; bus_size_t maxsize; sc = addr; if (sc->sc_drq8 != -1) maxsize = sc->sc_drq8_maxsize; else maxsize = sc->sc_drq16_maxsize; if (size > maxsize) size = maxsize; return size; } paddr_t sb_mappage(void *addr, void *mem, off_t off, int prot) { return isa_mappage(mem, off, prot); } int sbdsp_get_props(void *addr) { struct sbdsp_softc *sc; sc = addr; return AUDIO_PROP_MMAP | AUDIO_PROP_INDEPENDENT | (sc->sc_fullduplex ? AUDIO_PROP_FULLDUPLEX : 0); } #if NMPU > 0 /* * MIDI related routines. */ int sbdsp_midi_open(void *addr, int flags, void (*iintr)(void *, int), void (*ointr)(void *), void *arg) { struct sbdsp_softc *sc; sc = addr; DPRINTF(("sbdsp_midi_open: sc=%p\n", sc)); if (sc->sc_open != SB_CLOSED) return EBUSY; if (sbdsp_reset(sc) != 0) return EIO; sc->sc_open = SB_OPEN_MIDI; if (sc->sc_model >= SB_20) if (sbdsp_wdsp(sc, SB_MIDI_UART_INTR)) /* enter UART mode */ return EIO; sc->sc_intr8 = sbdsp_midi_intr; sc->sc_intrm = iintr; sc->sc_argm = arg; return 0; } void sbdsp_midi_close(void *addr) { struct sbdsp_softc *sc; sc = addr; DPRINTF(("sbdsp_midi_close: sc=%p\n", sc)); if (sc->sc_model >= SB_20) sbdsp_reset(sc); /* exit UART mode */ sc->sc_intrm = 0; sc->sc_open = SB_CLOSED; } int sbdsp_midi_output(void *addr, int d) { struct sbdsp_softc *sc; sc = addr; if (sc->sc_model < SB_20 && sbdsp_wdsp(sc, SB_MIDI_WRITE)) return EIO; if (sbdsp_wdsp(sc, d)) return EIO; return 0; } void sbdsp_midi_getinfo(void *addr, struct midi_info *mi) { struct sbdsp_softc *sc; sc = addr; mi->name = sc->sc_model < SB_20 ? "SB MIDI cmd" : "SB MIDI UART"; mi->props = MIDI_PROP_CAN_INPUT; } int sbdsp_midi_intr(void *addr) { struct sbdsp_softc *sc; sc = addr; sc->sc_intrm(sc->sc_argm, sbdsp_rdsp(sc)); return (0); } #endif