/* $NetBSD: sbdsp.c,v 1.5 1995/03/25 00:01:00 mycroft 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. * * $Id: sbdsp.c,v 1.5 1995/03/25 00:01:00 mycroft Exp $ */ /* * SoundBlaster Pro code provided by John Kohl, based on lots of * information he gleaned from Steve Haehnichen 's * SBlast driver for 386BSD and DOS driver code from Daniel Sachs * . */ #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include #ifdef DEBUG extern void Dprintf __P((const char *, ...)); #define DPRINTF(x) if (sbdspdebug) printf x int sbdspdebug = 0; #else #define DPRINTF(x) #endif #ifndef NEWCONFIG #define at_dma(flags, ptr, cc, chan) isa_dmastart(flags, ptr, cc, chan) #endif #ifndef SBDSP_NPOLL #define SBDSP_NPOLL 3000 #endif struct { int wdsp; int rdsp; int wmidi; } sberr; #ifdef DEBUG void sb_printsc(struct sbdsp_softc *sc) { int i; printf("open %d dmachan %d iobase %x locked %d\n", sc->sc_open, sc->sc_drq, sc->sc_iobase, sc->sc_locked); printf("hispeed %d irate %d orate %d encoding %x\n", sc->sc_adacmode, sc->sc_irate, sc->sc_orate, sc->encoding); printf("outport %d inport %d spkron %d nintr %d\n", sc->out_port, sc->in_port, sc->spkr_state, sc->sc_interrupts); printf("tc %x chans %x scintr %x arg %x\n", sc->sc_adactc, sc->sc_chans, sc->sc_intr, sc->sc_arg); printf("gain: "); for (i = 0; i < SB_NDEVS; i++) printf("%d ", sc->gain[i]); printf("\n"); } #endif /* * Probe / attach routines. */ /* * Probe for the soundblaster hardware. */ int sbdsp_probe(sc) struct sbdsp_softc *sc; { register u_short iobase = sc->sc_iobase; if (sbdsp_reset(sc) < 0) { DPRINTF(("sbdsp: couldn't reset card\n")); return 0; } sc->sc_model = sbversion(sc); return 1; } /* * Attach hardware to driver, attach hardware driver to audio * pseudo-device driver . */ void sbdsp_attach(sc) struct sbdsp_softc *sc; { register u_short iobase = sc->sc_iobase; sc->sc_locked = 0; #ifdef NEWCONFIG /* * We limit DMA transfers to a page, and use the generic DMA handling * code in isa.c. This code can end up copying a buffer, but since * the audio driver uses relative small buffers this isn't likely. * * This allocation scheme means that the maximum transfer is limited * by the page size (rather than 64k). This is reasonable. For 4K * pages, the transfer time at 48KHz is 4096 / 48000 = 85ms. This * is plenty long enough to amortize any fixed time overhead. */ at_setup_dmachan(sc->sc_drq, NBPG); #endif /* Set defaults */ if (ISSBPROCLASS(sc)) sc->sc_irate = sc->sc_orate = 45454; else sc->sc_irate = sc->sc_orate = 14925; sc->sc_chans = 1; sc->encoding = AUDIO_ENCODING_LINEAR; (void) sbdsp_set_in_sr_real(sc, sc->sc_irate); (void) sbdsp_set_out_sr_real(sc, sc->sc_orate); (void) sbdsp_set_in_port(sc, SB_MIC_PORT); (void) sbdsp_set_out_port(sc, SB_SPEAKER); if (ISSBPROCLASS(sc)) { int i; /* set mixer to default levels, by sending a mixer reset command. */ sbdsp_mix_write(sc, SBP_MIX_RESET, SBP_MIX_RESET); /* then some adjustments :) */ sbdsp_mix_write(sc, SBP_CD_VOL, sbdsp_stereo_vol(SBP_MAXVOL, SBP_MAXVOL)); sbdsp_mix_write(sc, SBP_DAC_VOL, sbdsp_stereo_vol(SBP_MAXVOL, SBP_MAXVOL)); sbdsp_mix_write(sc, SBP_MASTER_VOL, sbdsp_stereo_vol(SBP_MAXVOL, SBP_MAXVOL)); sbdsp_mix_write(sc, SBP_LINE_VOL, sbdsp_stereo_vol(SBP_MAXVOL, SBP_MAXVOL)); for (i = 0; i < SB_NDEVS; i++) sc->gain[i] = sbdsp_stereo_vol(SBP_MAXVOL, SBP_MAXVOL); } printf(": dsp v%d.%d\n", SBVER_MAJOR(sc->sc_model), SBVER_MINOR(sc->sc_model)); } /* * Various routines to interface to higher level audio driver */ void sbdsp_mix_write(sc, mixerport, val) struct sbdsp_softc *sc; int mixerport; int val; { int iobase = sc->sc_iobase; outb(iobase + SBP_MIXER_ADDR, mixerport); delay(10); outb(iobase + SBP_MIXER_DATA, val); delay(30); } int sbdsp_mix_read(sc, mixerport) struct sbdsp_softc *sc; int mixerport; { int iobase = sc->sc_iobase; outb(iobase + SBP_MIXER_ADDR, mixerport); delay(10); return inb(iobase + SBP_MIXER_DATA); } int sbdsp_set_in_sr(addr, sr) void *addr; u_long sr; { register struct sbdsp_softc *sc = addr; sc->sc_irate = sr; return 0; } int sbdsp_set_in_sr_real(addr, sr) void *addr; u_long sr; { register struct sbdsp_softc *sc = addr; int rval; if (rval = sbdsp_set_sr(sc, &sr, SB_INPUT_RATE)) return rval; sc->sc_irate = sr; sc->sc_dmain_inprogress = 0; /* do it again on next DMA out */ sc->sc_dmaout_inprogress = 0; return(0); } u_long sbdsp_get_in_sr(addr) void *addr; { register struct sbdsp_softc *sc = addr; return(sc->sc_irate); } int sbdsp_set_out_sr(addr, sr) void *addr; u_long sr; { register struct sbdsp_softc *sc = addr; sc->sc_orate = sr; return(0); } int sbdsp_set_out_sr_real(addr, sr) void *addr; u_long sr; { register struct sbdsp_softc *sc = addr; int rval; if (rval = sbdsp_set_sr(sc, &sr, SB_OUTPUT_RATE)) return rval; sc->sc_orate = sr; sc->sc_dmain_inprogress = 0; /* do it again on next DMA out */ return(0); } u_long sbdsp_get_out_sr(addr) void *addr; { register struct sbdsp_softc *sc = addr; return(sc->sc_orate); } int sbdsp_query_encoding(addr, fp) void *addr; struct audio_encoding *fp; { register struct sbdsp_softc *sc = addr; switch (fp->index) { case 0: strcpy(fp->name, "MU-Law"); fp->format_id = AUDIO_ENCODING_ULAW; break; case 2: strcpy(fp->name, "pcm16"); fp->format_id = AUDIO_ENCODING_PCM16; break; default: return(EINVAL); /*NOTREACHED*/ } return (0); } int sbdsp_set_encoding(addr, enc) void *addr; u_int enc; { register struct sbdsp_softc *sc = addr; switch(enc){ case AUDIO_ENCODING_ULAW: sc->encoding = AUDIO_ENCODING_ULAW; break; case AUDIO_ENCODING_LINEAR: sc->encoding = AUDIO_ENCODING_LINEAR; break; default: return (EINVAL); } return (0); } int sbdsp_get_encoding(addr) void *addr; { register struct sbdsp_softc *sc = addr; return(sc->encoding); } int sbdsp_set_precision(addr, prec) void *addr; u_int prec; { if (prec != 8) return(EINVAL); return(0); } int sbdsp_get_precision(addr) void *addr; { return(8); } int sbdsp_set_channels(addr, chans) void *addr; int chans; { register struct sbdsp_softc *sc = addr; int rval; if (ISSBPROCLASS(sc)) { if (chans != 1 && chans != 2) return(EINVAL); sc->sc_chans = chans; if (rval = sbdsp_set_in_sr_real(addr, sc->sc_irate)) return rval; sbdsp_mix_write(sc, SBP_STEREO, (sbdsp_mix_read(sc, SBP_STEREO) & ~SBP_PLAYMODE_MASK) | (chans == 2 ? SBP_PLAYMODE_STEREO : SBP_PLAYMODE_MONO)); /* recording channels needs to be done right when we start DMA recording. Just record number of channels for now and set stereo when ready. */ } else { if (chans != 1) return(EINVAL); sc->sc_chans = 1; } return(0); } int sbdsp_get_channels(addr) void *addr; { register struct sbdsp_softc *sc = addr; #if 0 /* recording stereo may frob the mixer output */ if (ISSBPROCLASS(sc)) { if ((sbdsp_mix_read(sc, SBP_STEREO) & SBP_PLAYMODE_MASK) == SBP_PLAYMODE_STEREO) { sc->sc_chans = 2; } else { sc->sc_chans = 1; } } else { sc->sc_chans = 1; } #endif return(sc->sc_chans); } int sbdsp_set_out_port(addr, port) void *addr; int port; { register struct sbdsp_softc *sc = addr; sc->out_port = port; /* Just record it */ return(0); } int sbdsp_get_out_port(addr) void *addr; { register struct sbdsp_softc *sc = addr; return(sc->out_port); } int sbdsp_set_in_port(addr, port) void *addr; int port; { register struct sbdsp_softc *sc = addr; int mixport, sbport; switch (port) { case SB_MIC_PORT: sbport = SBP_FROM_MIC; mixport = SBP_MIC_VOL; break; } if (ISSBPROCLASS(sc)) { switch (port) { case SB_LINE_IN_PORT: sbport = SBP_FROM_LINE; mixport = SBP_LINE_VOL; break; case SB_CD_PORT: sbport = SBP_FROM_CD; mixport = SBP_CD_VOL; break; case SB_DAC_PORT: case SB_FM_PORT: default: return(EINVAL); /*NOTREACHED*/ } } else { return(EINVAL); /*NOTREACHED*/ } sc->in_port = port; /* Just record it */ if (ISSBPROCLASS(sc)) { /* record from that port */ sbdsp_mix_write(sc, SBP_RECORD_SOURCE, SBP_RECORD_FROM(sbport, SBP_FILTER_OFF, SBP_FILTER_HIGH)); /* fetch gain from that port */ sc->gain[port] = sbdsp_mix_read(sc, mixport); } return(0); } int sbdsp_get_in_port(addr) void *addr; { register struct sbdsp_softc *sc = addr; return(sc->in_port); } int sbdsp_speaker_ctl(addr, newstate) void *addr; int newstate; { register struct sbdsp_softc *sc = addr; if ((newstate == SPKR_ON) && (sc->spkr_state == SPKR_OFF)) { sbdsp_spkron(sc); sc->spkr_state = SPKR_ON; } if ((newstate == SPKR_OFF) && (sc->spkr_state == SPKR_ON)) { sbdsp_spkroff(sc); sc->spkr_state = SPKR_OFF; } return(0); } int sbdsp_round_blocksize(addr, blk) void *addr; int blk; { register struct sbdsp_softc *sc = addr; sc->sc_last_hsr_size = sc->sc_last_hsw_size = 0; /* Higher speeds need bigger blocks to avoid popping and silence gaps. */ if ((sc->sc_orate > 8000 || sc->sc_irate > 8000) && (blk > NBPG/2 || blk < NBPG/4)) blk = NBPG/2; /* don't try to DMA too much at once, though. */ if (blk > NBPG) blk = NBPG; if (sc->sc_chans == 2) return (blk & ~1); /* must be even to preserve stereo separation */ else return(blk); /* Anything goes :-) */ } int sbdsp_commit_settings(addr) void *addr; { /* due to potentially unfortunate ordering in the above layers, re-do a few sets which may be important--input gains (adjust the proper channels), number of input channels (hit the record rate and set mode) */ register struct sbdsp_softc *sc = addr; sbdsp_set_out_sr_real(addr, sc->sc_orate); sbdsp_set_in_sr_real(addr, sc->sc_irate); sc->sc_last_hsw_size = sc->sc_last_hsr_size = 0; return(0); } int sbdsp_open(sc, dev, flags) register struct sbdsp_softc *sc; dev_t dev; int flags; { DPRINTF(("sbdsp_open: sc=0x%x\n", sc)); if (sc->sc_open != 0 || sbdsp_reset(sc) != 0) return ENXIO; sc->sc_open = 1; sc->sc_mintr = 0; sc->sc_intr = 0; sc->sc_arg = 0; sc->sc_locked = 0; if (ISSBPROCLASS(sc) && sbdsp_wdsp(sc->sc_iobase, SB_DSP_RECORD_MONO) < 0) { DPRINTF(("sbdsp_open: can't set mono mode\n")); /* we'll readjust when it's time for DMA. */ } sc->sc_dmain_inprogress = 0; sc->sc_dmaout_inprogress = 0; /* * Leave most things as they were; users must change things if * the previous process didn't leave it they way they wanted. * Looked at another way, it's easy to set up a configuration * in one program and leave it for another to inherit. */ DPRINTF(("sbdsp_open: opened\n")); return 0; } void sbdsp_close(addr) void *addr; { struct sbdsp_softc *sc = addr; DPRINTF(("sbdsp_close: sc=0x%x\n", sc)); sc->sc_open = 0; sbdsp_spkroff(sc); sc->spkr_state = SPKR_OFF; sc->sc_intr = 0; sc->sc_mintr = 0; /* XXX this will turn off any dma */ sbdsp_reset(sc); DPRINTF(("sbdsp_close: closed\n")); } /* * Lower-level routines */ /* * Reset the card. * Return non-zero if the card isn't detected. */ int sbdsp_reset(sc) register struct sbdsp_softc *sc; { register u_short iobase = sc->sc_iobase; /* * erase any memory of last transfer size. */ sc->sc_last_hsr_size = sc->sc_last_hsw_size = 0; /* * See SBK, section 11.3. * We pulse a reset signal into the card. * Gee, what a brilliant hardware design. */ outb(iobase + SBP_DSP_RESET, 1); delay(3); outb(iobase + SBP_DSP_RESET, 0); if (sbdsp_rdsp(iobase) != SB_MAGIC) return -1; return 0; } /* * Write a byte to the dsp. * XXX We are at the mercy of the card as we use a * polling loop and wait until it can take the byte. */ int sbdsp_wdsp(u_short iobase, int v) { register int i; for (i = SBDSP_NPOLL; --i >= 0; ) { if ((inb(iobase + SBP_DSP_WSTAT) & SB_DSP_BUSY) != 0) { delay(10); continue; } outb(iobase + SBP_DSP_WRITE, v); return 0; } ++sberr.wdsp; return -1; } /* * Read a byte from the DSP, using polling. */ int sbdsp_rdsp(u_short iobase) { register int i; for (i = SBDSP_NPOLL; --i >= 0; ) { if ((inb(iobase + SBP_DSP_RSTAT) & SB_DSP_READY) == 0) continue; return inb(iobase + SBP_DSP_READ); } ++sberr.rdsp; return -1; } /* * Doing certain things (like toggling the speaker) make * the SB hardware go away for a while, so pause a little. */ void sbdsp_to(arg) void *arg; { wakeup(arg); } void sbdsp_pause(sc) struct sbdsp_softc *sc; { extern int hz; timeout(sbdsp_to, sbdsp_to, hz/8); (void)tsleep(sbdsp_to, PWAIT, "sbpause", 0); } /* * Turn on the speaker. The SBK documention says this operation * can take up to 1/10 of a second. Higher level layers should * probably let the task sleep for this amount of time after * calling here. Otherwise, things might not work (because * sbdsp_wdsp() and sbdsp_rdsp() will probably timeout.) * * These engineers had their heads up their ass when * they designed this card. */ void sbdsp_spkron(sc) struct sbdsp_softc *sc; { (void)sbdsp_wdsp(sc->sc_iobase, SB_DSP_SPKR_ON); sbdsp_pause(sc); } /* * Turn off the speaker; see comment above. */ void sbdsp_spkroff(sc) struct sbdsp_softc *sc; { (void)sbdsp_wdsp(sc->sc_iobase, SB_DSP_SPKR_OFF); sbdsp_pause(sc); } /* * Read the version number out of the card. Return major code * in high byte, and minor code in low byte. */ short sbversion(sc) struct sbdsp_softc *sc; { register u_short iobase = sc->sc_iobase; short v; if (sbdsp_wdsp(iobase, SB_DSP_VERSION) < 0) return 0; v = sbdsp_rdsp(iobase) << 8; v |= sbdsp_rdsp(iobase); return ((v >= 0) ? v : 0); } /* * Halt a DMA in progress. A low-speed transfer can be * resumed with sbdsp_contdma(). */ int sbdsp_haltdma(addr) void *addr; { register struct sbdsp_softc *sc = addr; DPRINTF(("sbdsp_haltdma: sc=0x%x\n", sc)); if (sc->sc_locked) sbdsp_reset(sc); else (void)sbdsp_wdsp(sc->sc_iobase, SB_DSP_HALT); isa_dmaabort(sc->sc_drq); sc->dmaaddr = 0; sc->dmacnt = 0; sc->sc_locked = 0; sc->dmaflags = 0; sc->sc_dmain_inprogress = sc->sc_dmaout_inprogress = 0; return(0); } int sbdsp_contdma(addr) void *addr; { register struct sbdsp_softc *sc = addr; DPRINTF(("sbdsp_contdma: sc=0x%x\n", sc)); /* XXX how do we reinitialize the DMA controller state? do we care? */ (void)sbdsp_wdsp(sc->sc_iobase, SB_DSP_CONT); return(0); } /* * Time constant routines follow. See SBK, section 12. * Although they don't come out and say it (in the docs), * the card clearly uses a 1MHz countdown timer, as the * low-speed formula (p. 12-4) is: * tc = 256 - 10^6 / sr * In high-speed mode, the constant is the upper byte of a 16-bit counter, * and a 256MHz clock is used: * tc = 65536 - 256 * 10^ 6 / sr * Since we can only use the upper byte of the HS TC, the two formulae * are equivalent. (Why didn't they say so?) E.g., * (65536 - 256 * 10 ^ 6 / x) >> 8 = 256 - 10^6 / x * * The crossover point (from low- to high-speed modes) is different * for the SBPRO and SB20. The table on p. 12-5 gives the following data: * * SBPRO SB20 * ----- -------- * input ls min 4 KHz 4 KHz * input ls max 23 KHz 13 KHz * input hs max 44.1 KHz 15 KHz * output ls min 4 KHz 4 KHz * output ls max 23 KHz 23 KHz * output hs max 44.1 KHz 44.1 KHz */ #define SB_LS_MIN 0x06 /* 4000 Hz */ #define SBPRO_ADC_LS_MAX 0xd4 /* 22727 Hz */ #define SBPRO_ADC_HS_MAX 0xea /* 45454 Hz */ #define SBCLA_ADC_LS_MAX 0xb3 /* 12987 Hz */ #define SBCLA_ADC_HS_MAX 0xbd /* 14925 Hz */ #define SB_DAC_LS_MAX 0xd4 /* 22727 Hz */ #define SB_DAC_HS_MAX 0xea /* 45454 Hz */ /* * Convert a linear sampling rate into the DAC time constant. * Set *mode to indicate the high/low-speed DMA operation. * Because of limitations of the card, not all rates are possible. * We return the time constant of the closest possible rate. * The sampling rate limits are different for the DAC and ADC, * so isdac indicates output, and !isdac indicates input. */ int sbdsp_srtotc(sc, sr, mode, isdac) register struct sbdsp_softc *sc; int sr; int *mode; int isdac; { int adc_ls_max, adc_hs_max; register int tc; if (sr == 0) { *mode = SB_ADAC_LS; return SB_LS_MIN; } tc = 256 - 1000000 / sr; /* XXX use better rounding--compare distance to nearest tc on both sides of requested speed */ if (ISSBPROCLASS(sc)) { adc_ls_max = SBPRO_ADC_LS_MAX; adc_hs_max = SBPRO_ADC_HS_MAX; } else { adc_ls_max = SBCLA_ADC_LS_MAX; adc_hs_max = SBCLA_ADC_HS_MAX; } if (tc < SB_LS_MIN) { tc = SB_LS_MIN; *mode = SB_ADAC_LS; } else if (isdac) { if (tc <= SB_DAC_LS_MAX) *mode = SB_ADAC_LS; else { *mode = SB_ADAC_HS; if (tc > SB_DAC_HS_MAX) tc = SB_DAC_HS_MAX; } } else { if (tc <= adc_ls_max) *mode = SB_ADAC_LS; else { *mode = SB_ADAC_HS; if (tc > adc_hs_max) tc = adc_hs_max; } } return tc; } /* * Convert a DAC time constant to a sampling rate. * See SBK, section 12. */ int sbdsp_tctosr(sc, tc) register struct sbdsp_softc *sc; int tc; { int adc; if (ISSBPROCLASS(sc)) adc = SBPRO_ADC_HS_MAX; else adc = SBCLA_ADC_HS_MAX; if (tc > adc) tc = adc; return (1000000 / (256 - tc)); } int sbdsp_set_sr(sc, srp, isdac) register struct sbdsp_softc *sc; u_long *srp; int isdac; { register int tc; int mode; int sr = *srp; register u_short iobase; /* * A SBPro in stereo mode uses time constants at double the * actual rate. */ if (ISSBPRO(sc) && sc->sc_chans == 2) { if (sr > 22727) sr = 22727; /* Can't bounce it...order of operations may yield bogus sr here. */ sr *= 2; } else if (!ISSBPROCLASS(sc) && sc->sc_chans != 1) return EINVAL; tc = sbdsp_srtotc(sc, sr, &mode, isdac); DPRINTF(("sbdsp_set_sr: sc=0x%x sr=%d mode=0x%x\n", sc, sr, mode)); iobase = sc->sc_iobase; if (sbdsp_wdsp(iobase, SB_DSP_TIMECONST) < 0 || sbdsp_wdsp(iobase, tc) < 0) return EIO; sr = sbdsp_tctosr(sc, tc); if (ISSBPRO(sc) && sc->sc_chans == 2) *srp = sr / 2; else *srp = sr; sc->sc_adacmode = mode; sc->sc_adactc = tc; return 0; } int sbdsp_dma_input(addr, p, cc, intr, arg) void *addr; void *p; int cc; void (*intr)(); void *arg; { register struct sbdsp_softc *sc = addr; register u_short iobase; u_int phys; #ifdef DEBUG if (sbdspdebug > 1) Dprintf("sbdsp_dma_input: cc=%d 0x%x (0x%x)\n", cc, intr, arg); #endif if (sc->sc_chans == 2 && (cc & 1)) { DPRINTF(("sbdsp_dma_input: stereo input, odd bytecnt\n")); return EIO; } iobase = sc->sc_iobase; if (ISSBPROCLASS(sc) && !sc->sc_dmain_inprogress) { if (sc->sc_chans == 2) { if (sbdsp_wdsp(iobase, SB_DSP_RECORD_STEREO) < 0) goto badmode; sbdsp_mix_write(sc, SBP_STEREO, sbdsp_mix_read(sc, SBP_STEREO) & ~SBP_PLAYMODE_MASK); sbdsp_mix_write(sc, SBP_INFILTER, sbdsp_mix_read(sc, SBP_INFILTER) | SBP_FILTER_OFF); } else { if (sbdsp_wdsp(iobase, SB_DSP_RECORD_MONO) < 0) goto badmode; sbdsp_mix_write(sc, SBP_STEREO, sbdsp_mix_read(sc, SBP_STEREO) & ~SBP_PLAYMODE_MASK); sbdsp_mix_write(sc, SBP_INFILTER, sc->sc_irate <= 8000 ? sbdsp_mix_read(sc, SBP_INFILTER) & ~SBP_FILTER_MASK : sbdsp_mix_read(sc, SBP_INFILTER) | SBP_FILTER_OFF); } sc->sc_dmain_inprogress = 1; sc->sc_last_hsr_size = 0; /* restarting */ } sc->sc_dmaout_inprogress = 0; at_dma(B_READ, p, cc, sc->sc_drq); sc->sc_intr = intr; sc->sc_arg = arg; sc->dmaflags = B_READ; sc->dmaaddr = p; sc->dmacnt = --cc; /* DMA controller is strange...? */ if (sc->sc_adacmode == SB_ADAC_LS) { if (sbdsp_wdsp(iobase, SB_DSP_RDMA) < 0 || sbdsp_wdsp(iobase, cc) < 0 || sbdsp_wdsp(iobase, cc >> 8) < 0) { goto giveup; } } else { if (cc != sc->sc_last_hsr_size) { if (sbdsp_wdsp(iobase, SB_DSP_BLOCKSIZE) < 0 || sbdsp_wdsp(iobase, cc) < 0 || sbdsp_wdsp(iobase, cc >> 8) < 0) goto giveup; } if (sbdsp_wdsp(iobase, SB_DSP_HS_INPUT) < 0) goto giveup; sc->sc_last_hsr_size = cc; sc->sc_locked = 1; } return 0; giveup: isa_dmaabort(sc->sc_drq); sbdsp_reset(sc); sc->sc_intr = 0; sc->sc_arg = 0; return EIO; badmode: DPRINTF(("sbdsp_dma_input: can't set %s mode\n", sc->sc_chans == 2 ? "stereo" : "mono")); return EIO; } int sbdsp_dma_output(addr, p, cc, intr, arg) void *addr; void *p; int cc; void (*intr)(); void *arg; { register struct sbdsp_softc *sc = addr; register u_short iobase; #ifdef DEBUG if (sbdspdebug > 1) Dprintf("sbdsp_dma_output: cc=%d 0x%x (0x%x)\n", cc, intr, arg); #endif if (sc->sc_chans == 2 && cc & 1) { DPRINTF(("stereo playback odd bytes (%d)\n", cc)); return EIO; } if (ISSBPROCLASS(sc) && !sc->sc_dmaout_inprogress) { /* make sure we re-set stereo mixer bit when we start output. */ sbdsp_mix_write(sc, SBP_STEREO, (sbdsp_mix_read(sc, SBP_STEREO) & ~SBP_PLAYMODE_MASK) | (sc->sc_chans == 2 ? SBP_PLAYMODE_STEREO : SBP_PLAYMODE_MONO)); sc->sc_dmaout_inprogress = 1; sc->sc_last_hsw_size = 0; /* restarting */ } sc->sc_dmain_inprogress = 0; at_dma(B_WRITE, p, cc, sc->sc_drq); sc->sc_intr = intr; sc->sc_arg = arg; sc->dmaflags = B_WRITE; sc->dmaaddr = p; sc->dmacnt = --cc; /* a vagary of how DMA works, apparently. */ iobase = sc->sc_iobase; if (sc->sc_adacmode == SB_ADAC_LS) { if (sbdsp_wdsp(iobase, SB_DSP_WDMA) < 0 || sbdsp_wdsp(iobase, cc) < 0 || sbdsp_wdsp(iobase, cc >> 8) < 0) { DPRINTF(("sbdsp_dma_output: LS DMA start failed\n")); goto giveup; } } else { if (cc != sc->sc_last_hsw_size) { if (sbdsp_wdsp(iobase, SB_DSP_BLOCKSIZE) < 0) { /* sometimes fails initial startup?? */ delay(100); if (sbdsp_wdsp(iobase, SB_DSP_BLOCKSIZE) < 0) { DPRINTF(("sbdsp_dma_output: BLOCKSIZE failed\n")); goto giveup; } } if (sbdsp_wdsp(iobase, cc) < 0 || sbdsp_wdsp(iobase, cc >> 8) < 0) { DPRINTF(("sbdsp_dma_output: HS DMA start failed\n")); goto giveup; } sc->sc_last_hsw_size = cc; } if (sbdsp_wdsp(iobase, SB_DSP_HS_OUTPUT) < 0) { delay(100); if (sbdsp_wdsp(iobase, SB_DSP_HS_OUTPUT) < 0) { DPRINTF(("sbdsp_dma_output: HS DMA restart failed\n")); goto giveup; } } sc->sc_locked = 1; } return 0; giveup: isa_dmaabort(sc->sc_drq); sbdsp_reset(sc); sc->sc_intr = 0; sc->sc_arg = 0; return EIO; } /* * Only the DSP unit on the sound blaster generates interrupts. * There are three cases of interrupt: reception of a midi byte * (when mode is enabled), completion of dma transmission, or * completion of a dma reception. The three modes are mutually * exclusive so we know a priori which event has occurred. */ int sbdsp_intr(sc) register struct sbdsp_softc *sc; { #ifdef DEBUG if (sbdspdebug > 1) Dprintf("sbdsp_intr: intr=0x%x\n", sc->sc_intr); #endif sc->sc_interrupts++; sc->sc_locked = 0; /* clear interrupt */ inb(sc->sc_iobase + SBP_DSP_RSTAT); #if 0 if (sc->sc_mintr != 0) { int c = sbdsp_rdsp(sc->sc_iobase); (*sc->sc_mintr)(sc->sc_arg, c); } else #endif if (sc->sc_intr != 0) { /* * The SBPro used to develop and test this driver often * generated dma underruns--it interrupted to signal * completion of the DMA input recording block, but the * ISA DMA controller didn't think the channel was * finished. Maybe this is just a bus speed issue, I dunno, * but it seems strange and leads to channel-flipping with stereo * recording. Sigh. */ isa_dmadone(sc->dmaflags, sc->dmaaddr, sc->dmacnt, sc->sc_drq); sc->dmaflags = 0; sc->dmaaddr = 0; sc->dmacnt = 0; (*sc->sc_intr)(sc->sc_arg); } else return 0; return 1; } #if 0 /* * Enter midi uart mode and arrange for read interrupts * to vector to `intr'. This puts the card in a mode * which allows only midi I/O; the card must be reset * to leave this mode. Unfortunately, the card does not * use transmit interrupts, so bytes must be output * using polling. To keep the polling overhead to a * minimum, output should be driven off a timer. * This is a little tricky since only 320us separate * consecutive midi bytes. */ void sbdsp_set_midi_mode(sc, intr, arg) struct sbdsp_softc *sc; void (*intr)(); void *arg; { sbdsp_wdsp(sc->sc_iobase, SB_MIDI_UART_INTR); sc->sc_mintr = intr; sc->sc_intr = 0; sc->sc_arg = arg; } /* * Write a byte to the midi port, when in midi uart mode. */ void sbdsp_midi_output(sc, v) struct sbdsp_softc *sc; int v; { if (sbdsp_wdsp(sc->sc_iobase, v) < 0) ++sberr.wmidi; } #endif u_int sbdsp_get_silence(enc) int enc; { #define ULAW_SILENCE 0x7f #define LINEAR_SILENCE 0 u_int auzero; switch (enc) { case AUDIO_ENCODING_ULAW: auzero = ULAW_SILENCE; break; case AUDIO_ENCODING_PCM16: default: auzero = LINEAR_SILENCE; break; } return(auzero); } static u_char mulawtolin[256] = { 128, 4, 8, 12, 16, 20, 24, 28, 32, 36, 40, 44, 48, 52, 56, 60, 64, 66, 68, 70, 72, 74, 76, 78, 80, 82, 84, 86, 88, 90, 92, 94, 96, 97, 98, 99, 100, 101, 102, 103, 104, 105, 106, 107, 108, 109, 110, 111, 112, 112, 113, 113, 114, 114, 115, 115, 116, 116, 117, 117, 118, 118, 119, 119, 120, 120, 120, 121, 121, 121, 121, 122, 122, 122, 122, 123, 123, 123, 123, 124, 124, 124, 124, 124, 125, 125, 125, 125, 125, 125, 125, 125, 126, 126, 126, 126, 126, 126, 126, 126, 126, 126, 126, 126, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 255, 251, 247, 243, 239, 235, 231, 227, 223, 219, 215, 211, 207, 203, 199, 195, 191, 189, 187, 185, 183, 181, 179, 177, 175, 173, 171, 169, 167, 165, 163, 161, 159, 158, 157, 156, 155, 154, 153, 152, 151, 150, 149, 148, 147, 146, 145, 144, 143, 143, 142, 142, 141, 141, 140, 140, 139, 139, 138, 138, 137, 137, 136, 136, 135, 135, 135, 134, 134, 134, 134, 133, 133, 133, 133, 132, 132, 132, 132, 131, 131, 131, 131, 131, 130, 130, 130, 130, 130, 130, 130, 130, 129, 129, 129, 129, 129, 129, 129, 129, 129, 129, 129, 129, 128, 128, 128, 128, 128, 128, 128, 128, 128, 128, 128, 128, 128, 128, 128, 128, 128, 128, 128, 128, 128, 128, 128, 128, }; static u_char lintomulaw[256] = { 0, 0, 0, 0, 0, 1, 1, 1, 1, 2, 2, 2, 2, 3, 3, 3, 3, 4, 4, 4, 4, 5, 5, 5, 5, 6, 6, 6, 6, 7, 7, 7, 7, 8, 8, 8, 8, 9, 9, 9, 9, 10, 10, 10, 10, 11, 11, 11, 11, 12, 12, 12, 12, 13, 13, 13, 13, 14, 14, 14, 14, 15, 15, 15, 15, 16, 16, 17, 17, 18, 18, 19, 19, 20, 20, 21, 21, 22, 22, 23, 23, 24, 24, 25, 25, 26, 26, 27, 27, 28, 28, 29, 29, 30, 30, 31, 31, 32, 33, 34, 35, 36, 37, 38, 39, 40, 41, 42, 43, 44, 45, 46, 47, 48, 50, 52, 54, 56, 58, 60, 62, 65, 69, 73, 77, 83, 91, 103, 255, 231, 219, 211, 205, 201, 197, 193, 190, 188, 186, 184, 182, 180, 178, 176, 175, 174, 173, 172, 171, 170, 169, 168, 167, 166, 165, 164, 163, 162, 161, 160, 159, 159, 158, 158, 157, 157, 156, 156, 155, 155, 154, 154, 153, 153, 152, 152, 151, 151, 150, 150, 149, 149, 148, 148, 147, 147, 146, 146, 145, 145, 144, 144, 143, 143, 143, 143, 142, 142, 142, 142, 141, 141, 141, 141, 140, 140, 140, 140, 139, 139, 139, 139, 138, 138, 138, 138, 137, 137, 137, 137, 136, 136, 136, 136, 135, 135, 135, 135, 134, 134, 134, 134, 133, 133, 133, 133, 132, 132, 132, 132, 131, 131, 131, 131, 130, 130, 130, 130, 129, 129, 129, 129, 128, 128, 128, 128, }; void sbdsp_compress(e, p, cc) int e; u_char *p; int cc; { u_char *tab; switch (e) { case AUDIO_ENCODING_ULAW: tab = lintomulaw; break; default: return; } while (--cc >= 0) { *p = tab[*p]; ++p; } } void sbdsp_expand(e, p, cc) int e; u_char *p; int cc; { u_char *tab; switch (e) { case AUDIO_ENCODING_ULAW: tab = mulawtolin; break; default: return; } while (--cc >= 0) { *p = tab[*p]; ++p; } } int sbdsp_setfd(addr, flag) void *addr; int flag; { /* Can't do full-duplex */ return(ENOTTY); } int sbdsp_mixer_set_port(addr, cp) void *addr; mixer_ctrl_t *cp; { register struct sbdsp_softc *sc = addr; int error = 0; int src, gain; int left, right; DPRINTF(("sbdsp_mixer_set_port: port=%d num_channels=%d\n", cp->dev, cp->un.value.num_channels)); /* * Everything is a value except for SBPro special OUTPUT_MODE and * RECORD_SOURCE */ if (cp->type != AUDIO_MIXER_VALUE) { if (!ISSBPROCLASS(sc) || (cp->dev != SB_OUTPUT_MODE && cp->dev != SB_RECORD_SOURCE)) return EINVAL; } else { /* * 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. */ if (cp->un.value.num_channels == 2) { left = cp->un.value.level[AUDIO_MIXER_LEVEL_LEFT]; right = cp->un.value.level[AUDIO_MIXER_LEVEL_RIGHT]; } else left = right = cp->un.value.level[AUDIO_MIXER_LEVEL_MONO]; } if (ISSBPROCLASS(sc)) { /* The _PORT things are all signal inputs to the mixer. * Here we are tweaking their mixing level. * * We can also tweak the output stage volume (MASTER_VOL) */ gain = sbdsp_stereo_vol(SBP_AGAIN_TO_SBGAIN(left), SBP_AGAIN_TO_SBGAIN(right)); switch(cp->dev) { case SB_MIC_PORT: src = SBP_MIC_VOL; if (cp->un.value.num_channels != 1) error = EINVAL; else /* handle funny microphone gain */ gain = SBP_AGAIN_TO_MICGAIN(left); break; case SB_LINE_IN_PORT: src = SBP_LINE_VOL; break; case SB_DAC_PORT: src = SBP_DAC_VOL; break; case SB_FM_PORT: src = SBP_FM_VOL; break; case SB_CD_PORT: src = SBP_CD_VOL; break; case SB_SPEAKER: cp->dev = SB_MASTER_VOL; case SB_MASTER_VOL: src = SBP_MASTER_VOL; break; #if 0 case SB_OUTPUT_MODE: if (cp->type == AUDIO_MIXER_ENUM) return sbdsp_set_channels(addr, cp->un.ord); /* fall through...carefully! */ #endif case SB_RECORD_SOURCE: if (cp->type == AUDIO_MIXER_ENUM) return sbdsp_set_in_port(addr, cp->un.ord); /* else fall through: bad input */ case SB_TREBLE: case SB_BASS: default: error = EINVAL; break; } if (!error) sbdsp_mix_write(sc, src, gain); } else if (cp->dev != SB_MIC_PORT && cp->dev != SB_SPEAKER) error = EINVAL; if (!error) sc->gain[cp->dev] = gain; return(error); } int sbdsp_mixer_get_port(addr, cp) void *addr; mixer_ctrl_t *cp; { register struct sbdsp_softc *sc = addr; int error = 0; int done = 0; DPRINTF(("sbdsp_mixer_get_port: port=%d", cp->dev)); if (ISSBPROCLASS(sc)) switch(cp->dev) { case SB_MIC_PORT: if (cp->un.value.num_channels == 1) { cp->un.value.level[AUDIO_MIXER_LEVEL_MONO] = SBP_MICGAIN_TO_AGAIN(sc->gain[cp->dev]); return 0; } else return EINVAL; break; case SB_LINE_IN_PORT: case SB_DAC_PORT: case SB_FM_PORT: case SB_CD_PORT: case SB_MASTER_VOL: break; case SB_SPEAKER: cp->dev = SB_MASTER_VOL; break; default: error = EINVAL; break; } else { if (cp->un.value.num_channels != 1) /* no stereo on SB classic */ error = EINVAL; else switch(cp->dev) { case SB_MIC_PORT: break; case SB_SPEAKER: break; default: error = EINVAL; break; } } if (error == 0) { if (cp->un.value.num_channels == 1) { cp->un.value.level[AUDIO_MIXER_LEVEL_MONO] = SBP_SBGAIN_TO_AGAIN(sc->gain[cp->dev]); } else if (cp->un.value.num_channels == 2) { cp->un.value.level[AUDIO_MIXER_LEVEL_LEFT] = SBP_LEFTGAIN(sc->gain[cp->dev]); cp->un.value.level[AUDIO_MIXER_LEVEL_RIGHT] = SBP_RIGHTGAIN(sc->gain[cp->dev]); } else return EINVAL; } return(error); } int sbdsp_mixer_query_devinfo(addr, dip) void *addr; register mixer_devinfo_t *dip; { register struct sbdsp_softc *sc = addr; int done = 0; DPRINTF(("sbdsp_mixer_query_devinfo: index=%d\n", dip->index)); switch (dip->index) { case SB_MIC_PORT: dip->type = AUDIO_MIXER_VALUE; dip->mixer_class = SB_INPUT_CLASS; dip->prev = AUDIO_MIXER_LAST; dip->next = AUDIO_MIXER_LAST; strcpy(dip->label.name, AudioNmicrophone); dip->un.v.num_channels = 1; strcpy(dip->un.v.units.name, AudioNvolume); done = 1; break; case SB_SPEAKER: dip->type = AUDIO_MIXER_VALUE; dip->mixer_class = SB_OUTPUT_CLASS; dip->prev = AUDIO_MIXER_LAST; dip->next = AUDIO_MIXER_LAST; strcpy(dip->label.name, AudioNspeaker); dip->un.v.num_channels = 1; strcpy(dip->un.v.units.name, AudioNvolume); done = 1; break; 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); done = 1; break; 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); done = 1; break; } if (!done) { if (ISSBPROCLASS(sc)) switch(dip->index) { case SB_LINE_IN_PORT: dip->type = AUDIO_MIXER_VALUE; dip->mixer_class = SB_INPUT_CLASS; dip->prev = AUDIO_MIXER_LAST; dip->next = AUDIO_MIXER_LAST; strcpy(dip->label.name, AudioNline); dip->un.v.num_channels = 2; strcpy(dip->un.v.units.name, AudioNvolume); break; case SB_DAC_PORT: dip->type = AUDIO_MIXER_VALUE; dip->mixer_class = SB_INPUT_CLASS; dip->prev = AUDIO_MIXER_LAST; dip->next = AUDIO_MIXER_LAST; strcpy(dip->label.name, AudioNdac); dip->un.v.num_channels = 2; strcpy(dip->un.v.units.name, AudioNvolume); break; case SB_CD_PORT: dip->type = AUDIO_MIXER_VALUE; dip->mixer_class = SB_INPUT_CLASS; dip->prev = AUDIO_MIXER_LAST; dip->next = AUDIO_MIXER_LAST; strcpy(dip->label.name, AudioNcd); dip->un.v.num_channels = 2; strcpy(dip->un.v.units.name, AudioNvolume); break; case SB_FM_PORT: dip->type = AUDIO_MIXER_VALUE; dip->mixer_class = SB_INPUT_CLASS; dip->prev = AUDIO_MIXER_LAST; dip->next = AUDIO_MIXER_LAST; strcpy(dip->label.name, "fmsynth"); /* XXX move to audioio.h */ dip->un.v.num_channels = 2; strcpy(dip->un.v.units.name, AudioNvolume); break; case SB_MASTER_VOL: dip->type = AUDIO_MIXER_VALUE; dip->mixer_class = SB_OUTPUT_CLASS; dip->prev = AUDIO_MIXER_LAST; dip->next = SB_OUTPUT_MODE; strcpy(dip->label.name, AudioNvolume); dip->un.v.num_channels = 2; strcpy(dip->un.v.units.name, AudioNvolume); break; #if 0 case SB_OUTPUT_MODE: dip->mixer_class = SB_OUTPUT_CLASS; dip->type = AUDIO_MIXER_ENUM; dip->prev = SB_MASTER_VOL; dip->next = AUDIO_MIXER_LAST; strcpy(dip->label.name, AudioNmode); dip->un.e.num_mem = 2; strcpy(dip->un.e.member[0].label.name, AudioNmono); dip->un.e.member[0].ord = 1; /* nchans */ strcpy(dip->un.e.member[1].label.name, AudioNstereo); dip->un.e.member[1].ord = 2; /* nchans */ break; #endif case SB_RECORD_SOURCE: dip->mixer_class = SB_RECORD_CLASS; dip->type = AUDIO_MIXER_ENUM; dip->prev = AUDIO_MIXER_LAST; dip->next = AUDIO_MIXER_LAST; strcpy(dip->label.name, AudioNsource); dip->un.e.num_mem = 3; strcpy(dip->un.e.member[0].label.name, AudioNmicrophone); dip->un.e.member[0].ord = SB_MIC_PORT; strcpy(dip->un.e.member[1].label.name, AudioNcd); dip->un.e.member[1].ord = SB_CD_PORT; strcpy(dip->un.e.member[2].label.name, AudioNline); dip->un.e.member[2].ord = SB_LINE_IN_PORT; break; case SB_BASS: case SB_TREBLE: default: return ENXIO; /*NOTREACHED*/ } else return ENXIO; } DPRINTF(("AUDIO_MIXER_DEVINFO: name=%s\n", dip->label.name)); return 0; }