/* * Copyright (c) 1991, 1992, 1993 * The Regents of the University of California. All rights reserved. * * This software was developed by the Computer Systems Engineering group * at Lawrence Berkeley Laboratory under DARPA contract BG 91-66 and * contributed to Berkeley. * * All advertising materials mentioning features or use of this software * must display the following acknowledgement: * This product includes software developed by the University of * California, Lawrence Berkeley Laboratory. * * %sccs.include.redist.c% * * @(#)bsd_audio.c 8.1 (Berkeley) 06/11/93 * * from: $Header: bsd_audio.c,v 1.18 93/04/24 16:20:35 leres Exp $ (LBL) */ #include "bsdaudio.h" #if NBSDAUDIO > 0 #include #include #if BSD < 199103 #ifndef SUNOS #define SUNOS #endif #endif #include #include #include #include #include #include #include #ifndef SUNOS #include #endif #include #ifdef SUNOS #include #include #else #include #include #endif #include /* * Avoid name clashes with SunOS so we can config either the bsd or sun * streams driver in a SunOS kernel. */ #ifdef SUNOS #include #include #include struct selinfo { struct proc *si_proc; int si_coll; }; #else #include #include #include #endif #ifdef SUNOS #include "bsd_audiocompat.h" #endif /* * Initial/default block size is patchable. */ int audio_blocksize = DEFBLKSIZE; int audio_backlog = 400; /* 50ms in samples */ /* * Software state, per AMD79C30 audio chip. */ struct audio_softc { #ifndef SUNOS struct device sc_dev; /* base device */ struct intrhand sc_hwih; /* hardware interrupt vector */ struct intrhand sc_swih; /* software interrupt vector */ #endif int sc_interrupts; /* number of interrupts taken */ int sc_open; /* single use device */ u_long sc_wseek; /* timestamp of last frame written */ u_long sc_rseek; /* timestamp of last frame read */ struct mapreg sc_map; /* current contents of map registers */ struct selinfo sc_wsel; /* write selector */ struct selinfo sc_rsel; /* read selector */ /* * keep track of levels so we don't have to convert back from * MAP gain constants */ int sc_rlevel; /* record level */ int sc_plevel; /* play level */ int sc_mlevel; /* monitor level */ /* sc_au is special in that the hardware interrupt handler uses it */ struct auio sc_au; /* recv and xmit buffers, etc */ }; /* interrupt interfaces */ #ifndef AUDIO_C_HANDLER int audiohwintr __P((void *)); #endif int audioswintr __P((void *)); /* forward declarations */ int audio_sleep __P((struct aucb *, int)); void audio_setmap __P((volatile struct amd7930 *, struct mapreg *)); static void init_amd(); #if !defined(AUDIO_C_HANDLER) || defined(SUNOS) struct auio *audio_au; extern void audio_trap(); #endif #ifdef SUNOS struct audio_softc audio_softc; #define SOFTC(dev) &audio_softc #define UIOMOVE(cp, len, code, uio) uiomove(cp, len, code, uio) #define AUDIOOPEN(d, f, i, p)\ audioopen(d, f, i)\ dev_t d; int f, i; #define AUDIOCLOSE(d, f, i, p)\ audioclose(d, f, i)\ dev_t d; int f, i; #define AUDIOREAD(d, u, f) \ audioread(d, u) dev_t d; struct uio *u; #define AUDIOWRITE(d, u, f) \ audiowrite(d, u) dev_t d; struct uio *u; #define AUDIOIOCTL(d, c, a, f, o)\ audioioctl(d, c, a, f)\ dev_t d; int c; caddr_t a; int f; #define AUDIOSELECT(d, r, p)\ audio_select(d, r, p)\ dev_t d; int r; struct proc *p; #define AUDIO_SET_SWINTR set_intreg(IR_SOFT_INT4, 1) int audioselect(dev, rw) register dev_t dev; int rw; { return (audio_select(dev, rw, u.u_procp)); } static void selrecord(p, si) struct proc *p; struct selinfo *si; { if (si->si_proc != 0) si->si_coll = 1; else si->si_proc = p; } #define SELWAKEUP(si) \ {\ if ((si)->si_proc != 0) {\ selwakeup((si)->si_proc, (si)->si_coll); \ (si)->si_proc = 0;\ (si)->si_coll = 0;\ }\ } static int audioattach(); static int audioidentify(); struct dev_ops bsdaudio_ops = { 0, audioidentify, audioattach, }; static int audioidentify(cp) char *cp; { return (strcmp(cp, "audio") == 0); } static int audioattach(dev) struct dev_info *dev; { register struct audio_softc *sc; register volatile struct amd7930 *amd; struct dev_reg *reg; sc = &audio_softc; if (dev->devi_nreg != 1 || dev->devi_nintr != 1) { printf("audio: bad config\n"); return (-1); } reg = dev->devi_reg; amd = (struct amd7930 *)map_regs(reg->reg_addr, reg->reg_size, reg->reg_bustype); sc->sc_au.au_amd = amd; init_amd(amd); audio_au = &sc->sc_au; #ifndef AUDIO_C_HANDLER settrap(dev->devi_intr->int_pri, audio_trap); #else /* XXX */ addintr(dev->devi_intr->int_pri, audiohwintr, dev->devi_name, dev->devi_unit); #endif addintr(4, audioswintr, dev->devi_name, dev->devi_unit); report_dev(dev); return (0); } #else #define AUDIOOPEN(d, f, i, p) audioopen(dev_t d, int f, int i, struct proc *p) #define AUDIOCLOSE(d, f, i, p) audioclose(dev_t d, int f, int i, \ struct proc *p) #define AUDIOREAD(d, u, f) audioread(dev_t d, struct uio *u, int f) #define AUDIOWRITE(d, u, f) audiowrite(dev_t d, struct uio *u, int f) #define AUDIOIOCTL(d, c, a, f, o)\ audioioctl(dev_t dev, int c, caddr_t a, int f, struct proc *p) #define AUDIOSELECT(d, r, p) audioselect(dev_t dev, int rw, struct proc *p) #define SELWAKEUP selwakeup #define AUDIO_SET_SWINTR ienab_bis(IE_L6) /* autoconfiguration driver */ void audioattach(struct device *, struct device *, void *); struct cfdriver audiocd = { NULL, "audio", matchbyname, audioattach, DV_DULL, sizeof(struct audio_softc) }; #define SOFTC(dev) audiocd.cd_devs[minor(dev)] #define UIOMOVE(cp, len, code, uio) uiomove(cp, len, uio) /* * Audio chip found. */ void audioattach(parent, self, args) struct device *parent, *self; void *args; { register struct audio_softc *sc = (struct audio_softc *)self; register struct romaux *ra = args; register volatile struct amd7930 *amd; register int pri; if (ra->ra_nintr != 1) { printf(": expected 1 interrupt, got %d\n", ra->ra_nintr); return; } pri = ra->ra_intr[0].int_pri; printf(" pri %d, softpri %d\n", pri, PIL_AUSOFT); amd = (volatile struct amd7930 *)(ra->ra_vaddr ? ra->ra_vaddr : mapiodev(ra->ra_paddr, sizeof *amd)); sc->sc_au.au_amd = amd; init_amd(amd); #ifndef AUDIO_C_HANDLER audio_au = &sc->sc_au; intr_fasttrap(pri, audio_trap); #else sc->sc_hwih.ih_fun = audiohwintr; sc->sc_hwih.ih_arg = &sc->sc_au; intr_establish(pri, &sc->sc_hwih); #endif sc->sc_swih.ih_fun = audioswintr; sc->sc_swih.ih_arg = sc; intr_establish(PIL_AUSOFT, &sc->sc_swih); } #endif static void init_amd(amd) register volatile struct amd7930 *amd; { /* disable interrupts */ amd->cr = AMDR_INIT; amd->dr = AMD_INIT_PMS_ACTIVE | AMD_INIT_INT_DISABLE; /* * Initialize the mux unit. We use MCR3 to route audio (MAP) * through channel Bb. MCR1 and MCR2 are unused. * Setting the INT enable bit in MCR4 will generate an interrupt * on each converted audio sample. */ amd->cr = AMDR_MUX_1_4; amd->dr = 0; amd->dr = 0; amd->dr = (AMD_MCRCHAN_BB << 4) | AMD_MCRCHAN_BA; amd->dr = AMD_MCR4_INT_ENABLE; } static int audio_default_level = 150; static void ausetrgain __P((struct audio_softc *, int)); static void ausetpgain __P((struct audio_softc *, int)); static void ausetmgain __P((struct audio_softc *, int)); static int audiosetinfo __P((struct audio_softc *, struct audio_info *)); static int audiogetinfo __P((struct audio_softc *, struct audio_info *)); struct sun_audio_info; static int sunaudiosetinfo __P((struct audio_softc *, struct sun_audio_info *)); static int sunaudiogetinfo __P((struct audio_softc *, struct sun_audio_info *)); static void audio_setmmr2 __P((volatile struct amd7930 *, int)); /* ARGSUSED */ int AUDIOOPEN(dev, flags, ifmt, p) { register struct audio_softc *sc; register volatile struct amd7930 *amd; int unit = minor(dev); #ifdef SUNOS if (unit > 0) return (ENXIO); sc = &audio_softc; #else if (unit >= audiocd.cd_ndevs || (sc = audiocd.cd_devs[unit]) == NULL) return (ENXIO); #endif if (sc->sc_open) return (EBUSY); sc->sc_open = 1; sc->sc_au.au_lowat = audio_blocksize; sc->sc_au.au_hiwat = AUCB_SIZE - sc->sc_au.au_lowat; sc->sc_au.au_blksize = audio_blocksize; sc->sc_au.au_backlog = audio_backlog; /* set up read and write blocks and `dead sound' zero value. */ AUCB_INIT(&sc->sc_au.au_rb); sc->sc_au.au_rb.cb_thresh = AUCB_SIZE; AUCB_INIT(&sc->sc_au.au_wb); sc->sc_au.au_wb.cb_thresh = -1; /* nothing read or written yet */ sc->sc_rseek = 0; sc->sc_wseek = 0; bzero((char *)&sc->sc_map, sizeof sc->sc_map); /* default to speaker */ sc->sc_map.mr_mmr2 = AMD_MMR2_AINB | AMD_MMR2_LS; /* enable interrupts and set parameters established above */ amd = sc->sc_au.au_amd; audio_setmmr2(amd, sc->sc_map.mr_mmr2); ausetrgain(sc, audio_default_level); ausetpgain(sc, audio_default_level); ausetmgain(sc, 0); amd->cr = AMDR_INIT; amd->dr = AMD_INIT_PMS_ACTIVE; return (0); } static int audio_drain(sc) register struct audio_softc *sc; { register int error; while (!AUCB_EMPTY(&sc->sc_au.au_wb)) if ((error = audio_sleep(&sc->sc_au.au_wb, 0)) != 0) return (error); return (0); } /* * Close an audio chip. */ /* ARGSUSED */ int AUDIOCLOSE(dev, flags, ifmt, p) { register struct audio_softc *sc = SOFTC(dev); register volatile struct amd7930 *amd; register struct aucb *cb; register int s; /* * Block until output drains, but allow ^C interrupt. */ sc->sc_au.au_lowat = 0; /* avoid excessive wakeups */ s = splaudio(); /* * If there is pending output, let it drain (unless * the output is paused). */ cb = &sc->sc_au.au_wb; if (!AUCB_EMPTY(cb) && !cb->cb_pause) (void)audio_drain(sc); /* * Disable interrupts, clear open flag, and done. */ amd = sc->sc_au.au_amd; amd->cr = AMDR_INIT; amd->dr = AMD_INIT_PMS_ACTIVE | AMD_INIT_INT_DISABLE; splx(s); sc->sc_open = 0; return (0); } int audio_sleep(cb, thresh) register struct aucb *cb; register int thresh; { register int error; register int s = splaudio(); cb->cb_thresh = thresh; error = tsleep((caddr_t)cb, (PZERO + 1) | PCATCH, "audio", 0); splx(s); return (error); } /* ARGSUSED */ int AUDIOREAD(dev, uio, ioflag) { register struct audio_softc *sc = SOFTC(dev); register struct aucb *cb; register int n, head, taildata, error; register int blocksize = sc->sc_au.au_blksize; if (uio->uio_resid == 0) return (0); cb = &sc->sc_au.au_rb; error = 0; cb->cb_drops = 0; sc->sc_rseek = sc->sc_au.au_stamp - AUCB_LEN(cb); do { while (AUCB_LEN(cb) < blocksize) { #ifndef SUNOS if (ioflag & IO_NDELAY) { error = EWOULDBLOCK; return (error); } #endif if ((error = audio_sleep(cb, blocksize)) != 0) return (error); } /* * The space calculation can only err on the short * side if an interrupt occurs during processing: * only cb_tail is altered in the interrupt code. */ head = cb->cb_head; if ((n = AUCB_LEN(cb)) > uio->uio_resid) n = uio->uio_resid; taildata = AUCB_SIZE - head; if (n > taildata) { error = UIOMOVE((caddr_t)cb->cb_data + head, taildata, UIO_READ, uio); if (error == 0) error = UIOMOVE((caddr_t)cb->cb_data, n - taildata, UIO_READ, uio); } else error = UIOMOVE((caddr_t)cb->cb_data + head, n, UIO_READ, uio); if (error) break; head = AUCB_MOD(head + n); cb->cb_head = head; } while (uio->uio_resid >= blocksize); return (error); } /* ARGSUSED */ int AUDIOWRITE(dev, uio, ioflag) { register struct audio_softc *sc = SOFTC(dev); register struct aucb *cb = &sc->sc_au.au_wb; register int n, tail, tailspace, error, first, watermark; error = 0; first = 1; while (uio->uio_resid > 0) { watermark = sc->sc_au.au_hiwat; while (AUCB_LEN(cb) > watermark) { #ifndef SUNOS if (ioflag & IO_NDELAY) { error = EWOULDBLOCK; return (error); } #endif if ((error = audio_sleep(cb, watermark)) != 0) return (error); watermark = sc->sc_au.au_lowat; } /* * The only value that can change on an interrupt is * cb->cb_head. We only pull that out once to decide * how much to write into cb_data; if we lose a race * and cb_head changes, we will merely be overly * conservative. For a legitimate time stamp, * however, we need to synchronize the accesses to * au_stamp and cb_head at a high ipl below. */ tail = cb->cb_tail; if ((n = (AUCB_SIZE - 1) - AUCB_LEN(cb)) > uio->uio_resid) { n = uio->uio_resid; if (cb->cb_head == tail && n <= sc->sc_au.au_blksize && sc->sc_au.au_stamp - sc->sc_wseek > 400) { /* * the write is 'small', the buffer is empty * and we have been silent for at least 50ms * so we might be dealing with an application * that writes frames synchronously with * reading them. If so, we need an output * backlog to cover scheduling delays or * there will be gaps in the sound output. * Also take this opportunity to reset the * buffer pointers in case we ended up on * a bad boundary (odd byte, blksize bytes * from end, etc.). */ register u_int* ip; register int muzero = 0x7f7f7f7f; register int i = splaudio(); cb->cb_head = cb->cb_tail = 0; splx(i); tail = sc->sc_au.au_backlog; ip = (u_int*)cb->cb_data; for (i = tail >> 2; --i >= 0; ) *ip++ = muzero; } } tailspace = AUCB_SIZE - tail; if (n > tailspace) { /* write first part at tail and rest at head */ error = UIOMOVE((caddr_t)cb->cb_data + tail, tailspace, UIO_WRITE, uio); if (error == 0) error = UIOMOVE((caddr_t)cb->cb_data, n - tailspace, UIO_WRITE, uio); } else error = UIOMOVE((caddr_t)cb->cb_data + tail, n, UIO_WRITE, uio); if (error) break; tail = AUCB_MOD(tail + n); if (first) { register int s = splaudio(); sc->sc_wseek = AUCB_LEN(cb) + sc->sc_au.au_stamp + 1; /* * To guarantee that a write is contiguous in the * sample space, we clear the drop count the first * time through. If we later get drops, we will * break out of the loop below, before writing * a new frame. */ cb->cb_drops = 0; cb->cb_tail = tail; splx(s); first = 0; } else { if (cb->cb_drops != 0) break; cb->cb_tail = tail; } } return (error); } /* Sun audio compatibility */ struct sun_audio_prinfo { u_int sample_rate; u_int channels; u_int precision; u_int encoding; u_int gain; u_int port; u_int reserved0[4]; u_int samples; u_int eof; u_char pause; u_char error; u_char waiting; u_char reserved1[3]; u_char open; u_char active; }; struct sun_audio_info { struct sun_audio_prinfo play; struct sun_audio_prinfo record; u_int monitor_gain; u_int reserved[4]; }; #ifndef SUNOS #define SUNAUDIO_GETINFO _IOR('A', 1, struct sun_audio_info) #define SUNAUDIO_SETINFO _IOWR('A', 2, struct sun_audio_info) #else #define SUNAUDIO_GETINFO _IOR(A, 1, struct sun_audio_info) #define SUNAUDIO_SETINFO _IOWR(A, 2, struct sun_audio_info) #endif /* ARGSUSED */ int AUDIOIOCTL(dev, cmd, addr, flag, p) { register struct audio_softc *sc = SOFTC(dev); int error = 0, s; switch (cmd) { case AUDIO_GETMAP: bcopy((caddr_t)&sc->sc_map, addr, sizeof(sc->sc_map)); break; case AUDIO_SETMAP: bcopy(addr, (caddr_t)&sc->sc_map, sizeof(sc->sc_map)); sc->sc_map.mr_mmr2 &= 0x7f; audio_setmap(sc->sc_au.au_amd, &sc->sc_map); break; case AUDIO_FLUSH: s = splaudio(); AUCB_INIT(&sc->sc_au.au_rb); AUCB_INIT(&sc->sc_au.au_wb); sc->sc_au.au_stamp = 0; splx(s); sc->sc_wseek = 0; sc->sc_rseek = 0; break; /* * Number of read samples dropped. We don't know where or * when they were dropped. */ case AUDIO_RERROR: *(int *)addr = sc->sc_au.au_rb.cb_drops != 0; break; /* * How many samples will elapse until mike hears the first * sample of what we last wrote? */ case AUDIO_WSEEK: s = splaudio(); *(u_long *)addr = sc->sc_wseek - sc->sc_au.au_stamp + AUCB_LEN(&sc->sc_au.au_rb); splx(s); break; case AUDIO_SETINFO: error = audiosetinfo(sc, (struct audio_info *)addr); break; case AUDIO_GETINFO: error = audiogetinfo(sc, (struct audio_info *)addr); break; case SUNAUDIO_GETINFO: error = sunaudiogetinfo(sc, (struct sun_audio_info *)addr); break; case SUNAUDIO_SETINFO: error = sunaudiosetinfo(sc, (struct sun_audio_info *)addr); break; case AUDIO_DRAIN: error = audio_drain(sc); break; default: error = EINVAL; break; } return (error); } /* ARGSUSED */ int AUDIOSELECT(dev, rw, p) { register struct audio_softc *sc = SOFTC(dev); register struct aucb *cb; register int s = splaudio(); switch (rw) { case FREAD: cb = &sc->sc_au.au_rb; if (AUCB_LEN(cb) >= sc->sc_au.au_blksize) { splx(s); return (1); } selrecord(p, &sc->sc_rsel); cb->cb_thresh = sc->sc_au.au_blksize; break; case FWRITE: cb = &sc->sc_au.au_wb; if (AUCB_LEN(cb) <= sc->sc_au.au_lowat) { splx(s); return (1); } selrecord(p, &sc->sc_wsel); cb->cb_thresh = sc->sc_au.au_lowat; break; } splx(s); return (0); } #ifdef AUDIO_C_HANDLER int audiohwintr(au0) void *au0; { #ifdef SUNOS register struct auio *au = audio_au; #else register struct auio *au = au0; #endif register volatile struct amd7930 *amd = au->au_amd; register struct aucb *cb; register int h, t, k; k = amd->ir; /* clear interrupt */ ++au->au_stamp; /* receive incoming data */ cb = &au->au_rb; h = cb->cb_head; t = cb->cb_tail; k = AUCB_MOD(t + 1); if (h == k) cb->cb_drops++; else if (cb->cb_pause != 0) cb->cb_pdrops++; else { cb->cb_data[t] = amd->bbrb; cb->cb_tail = t = k; } if (AUCB_MOD(t - h) >= cb->cb_thresh) { cb->cb_thresh = AUCB_SIZE; cb->cb_waking = 1; AUDIO_SET_SWINTR; } /* send outgoing data */ cb = &au->au_wb; h = cb->cb_head; t = cb->cb_tail; if (h == t) cb->cb_drops++; else if (cb->cb_pause != 0) cb->cb_pdrops++; else { cb->cb_head = h = AUCB_MOD(h + 1); amd->bbtb = cb->cb_data[h]; } if (AUCB_MOD(t - h) <= cb->cb_thresh) { cb->cb_thresh = -1; cb->cb_waking = 1; AUDIO_SET_SWINTR; } return (1); } #endif /* ARGSUSED */ int audioswintr(sc0) void *sc0; { register struct audio_softc *sc; register int s, ret = 0; #ifdef SUNOS sc = &audio_softc; #else sc = sc0; #endif s = splaudio(); if (sc->sc_au.au_rb.cb_waking != 0) { sc->sc_au.au_rb.cb_waking = 0; splx(s); ret = 1; wakeup((caddr_t)&sc->sc_au.au_rb); SELWAKEUP(&sc->sc_rsel); } if (sc->sc_au.au_wb.cb_waking != 0) { sc->sc_au.au_wb.cb_waking = 0; splx(s); ret = 1; wakeup((caddr_t)&sc->sc_au.au_wb); SELWAKEUP(&sc->sc_wsel); } else splx(s); return (ret); } /* Write 16 bits of data from variable v to the data port of the audio chip */ #define WAMD16(amd, v) ((amd)->dr = (v), (amd)->dr = (v) >> 8) void audio_setmap(amd, map) register volatile struct amd7930 *amd; register struct mapreg *map; { register int i, s, v; s = splaudio(); amd->cr = AMDR_MAP_1_10; for (i = 0; i < 8; i++) { v = map->mr_x[i]; WAMD16(amd, v); } for (i = 0; i < 8; ++i) { v = map->mr_r[i]; WAMD16(amd, v); } v = map->mr_gx; WAMD16(amd, v); v = map->mr_gr; WAMD16(amd, v); v = map->mr_ger; WAMD16(amd, v); v = map->mr_stgr; WAMD16(amd, v); v = map->mr_ftgr; WAMD16(amd, v); v = map->mr_atgr; WAMD16(amd, v); amd->dr = map->mr_mmr1; amd->dr = map->mr_mmr2; splx(s); } /* * Set the mmr1 register and one other 16 bit register in the audio chip. * The other register is indicated by op and val. */ void audio_setmmr1(amd, mmr1, op, val) register volatile struct amd7930 *amd; register int mmr1; register int op; register int val; { register int s = splaudio(); amd->cr = AMDR_MAP_MMR1; amd->dr = mmr1; amd->cr = op; WAMD16(amd, val); splx(s); } /* * Set the mmr2 register. */ static void audio_setmmr2(amd, mmr2) register volatile struct amd7930 *amd; register int mmr2; { register int s = splaudio(); amd->cr = AMDR_MAP_MMR2; amd->dr = mmr2; splx(s); } /* * gx, gr & stg gains. this table must contain 256 elements with * the 0th being "infinity" (the magic value 9008). The remaining * elements match sun's gain curve (but with higher resolution): * -18 to 0dB in .16dB steps then 0 to 12dB in .08dB steps. */ static const u_short gx_coeff[256] = { 0x9008, 0x8b7c, 0x8b51, 0x8b45, 0x8b42, 0x8b3b, 0x8b36, 0x8b33, 0x8b32, 0x8b2a, 0x8b2b, 0x8b2c, 0x8b25, 0x8b23, 0x8b22, 0x8b22, 0x9122, 0x8b1a, 0x8aa3, 0x8aa3, 0x8b1c, 0x8aa6, 0x912d, 0x912b, 0x8aab, 0x8b12, 0x8aaa, 0x8ab2, 0x9132, 0x8ab4, 0x913c, 0x8abb, 0x9142, 0x9144, 0x9151, 0x8ad5, 0x8aeb, 0x8a79, 0x8a5a, 0x8a4a, 0x8b03, 0x91c2, 0x91bb, 0x8a3f, 0x8a33, 0x91b2, 0x9212, 0x9213, 0x8a2c, 0x921d, 0x8a23, 0x921a, 0x9222, 0x9223, 0x922d, 0x9231, 0x9234, 0x9242, 0x925b, 0x92dd, 0x92c1, 0x92b3, 0x92ab, 0x92a4, 0x92a2, 0x932b, 0x9341, 0x93d3, 0x93b2, 0x93a2, 0x943c, 0x94b2, 0x953a, 0x9653, 0x9782, 0x9e21, 0x9d23, 0x9cd2, 0x9c23, 0x9baa, 0x9bde, 0x9b33, 0x9b22, 0x9b1d, 0x9ab2, 0xa142, 0xa1e5, 0x9a3b, 0xa213, 0xa1a2, 0xa231, 0xa2eb, 0xa313, 0xa334, 0xa421, 0xa54b, 0xada4, 0xac23, 0xab3b, 0xaaab, 0xaa5c, 0xb1a3, 0xb2ca, 0xb3bd, 0xbe24, 0xbb2b, 0xba33, 0xc32b, 0xcb5a, 0xd2a2, 0xe31d, 0x0808, 0x72ba, 0x62c2, 0x5c32, 0x52db, 0x513e, 0x4cce, 0x43b2, 0x4243, 0x41b4, 0x3b12, 0x3bc3, 0x3df2, 0x34bd, 0x3334, 0x32c2, 0x3224, 0x31aa, 0x2a7b, 0x2aaa, 0x2b23, 0x2bba, 0x2c42, 0x2e23, 0x25bb, 0x242b, 0x240f, 0x231a, 0x22bb, 0x2241, 0x2223, 0x221f, 0x1a33, 0x1a4a, 0x1acd, 0x2132, 0x1b1b, 0x1b2c, 0x1b62, 0x1c12, 0x1c32, 0x1d1b, 0x1e71, 0x16b1, 0x1522, 0x1434, 0x1412, 0x1352, 0x1323, 0x1315, 0x12bc, 0x127a, 0x1235, 0x1226, 0x11a2, 0x1216, 0x0a2a, 0x11bc, 0x11d1, 0x1163, 0x0ac2, 0x0ab2, 0x0aab, 0x0b1b, 0x0b23, 0x0b33, 0x0c0f, 0x0bb3, 0x0c1b, 0x0c3e, 0x0cb1, 0x0d4c, 0x0ec1, 0x079a, 0x0614, 0x0521, 0x047c, 0x0422, 0x03b1, 0x03e3, 0x0333, 0x0322, 0x031c, 0x02aa, 0x02ba, 0x02f2, 0x0242, 0x0232, 0x0227, 0x0222, 0x021b, 0x01ad, 0x0212, 0x01b2, 0x01bb, 0x01cb, 0x01f6, 0x0152, 0x013a, 0x0133, 0x0131, 0x012c, 0x0123, 0x0122, 0x00a2, 0x011b, 0x011e, 0x0114, 0x00b1, 0x00aa, 0x00b3, 0x00bd, 0x00ba, 0x00c5, 0x00d3, 0x00f3, 0x0062, 0x0051, 0x0042, 0x003b, 0x0033, 0x0032, 0x002a, 0x002c, 0x0025, 0x0023, 0x0022, 0x001a, 0x0021, 0x001b, 0x001b, 0x001d, 0x0015, 0x0013, 0x0013, 0x0012, 0x0012, 0x000a, 0x000a, 0x0011, 0x0011, 0x000b, 0x000b, 0x000c, 0x000e, }; /* * second stage play gain. */ static const u_short ger_coeff[] = { 0x431f, /* 5. dB */ 0x331f, /* 5.5 dB */ 0x40dd, /* 6. dB */ 0x11dd, /* 6.5 dB */ 0x440f, /* 7. dB */ 0x411f, /* 7.5 dB */ 0x311f, /* 8. dB */ 0x5520, /* 8.5 dB */ 0x10dd, /* 9. dB */ 0x4211, /* 9.5 dB */ 0x410f, /* 10. dB */ 0x111f, /* 10.5 dB */ 0x600b, /* 11. dB */ 0x00dd, /* 11.5 dB */ 0x4210, /* 12. dB */ 0x110f, /* 13. dB */ 0x7200, /* 14. dB */ 0x2110, /* 15. dB */ 0x2200, /* 15.9 dB */ 0x000b, /* 16.9 dB */ 0x000f /* 18. dB */ #define NGER (sizeof(ger_coeff) / sizeof(ger_coeff[0])) }; static void ausetrgain(sc, level) register struct audio_softc *sc; register int level; { level &= 0xff; sc->sc_rlevel = level; sc->sc_map.mr_mmr1 |= AMD_MMR1_GX; sc->sc_map.mr_gx = gx_coeff[level]; audio_setmmr1(sc->sc_au.au_amd, sc->sc_map.mr_mmr1, AMDR_MAP_GX, sc->sc_map.mr_gx); } static void ausetpgain(sc, level) register struct audio_softc *sc; register int level; { register int gi, s; register volatile struct amd7930 *amd; level &= 0xff; sc->sc_plevel = level; sc->sc_map.mr_mmr1 |= AMD_MMR1_GER|AMD_MMR1_GR; level *= 256 + NGER; level >>= 8; if (level >= 256) { gi = level - 256; level = 255; } else gi = 0; sc->sc_map.mr_ger = ger_coeff[gi]; sc->sc_map.mr_gr = gx_coeff[level]; amd = sc->sc_au.au_amd; s = splaudio(); amd->cr = AMDR_MAP_MMR1; amd->dr = sc->sc_map.mr_mmr1; amd->cr = AMDR_MAP_GR; gi = sc->sc_map.mr_gr; WAMD16(amd, gi); amd->cr = AMDR_MAP_GER; gi = sc->sc_map.mr_ger; WAMD16(amd, gi); splx(s); } static void ausetmgain(sc, level) register struct audio_softc *sc; register int level; { level &= 0xff; sc->sc_mlevel = level; sc->sc_map.mr_mmr1 |= AMD_MMR1_STG; sc->sc_map.mr_stgr = gx_coeff[level]; audio_setmmr1(sc->sc_au.au_amd, sc->sc_map.mr_mmr1, AMDR_MAP_STG, sc->sc_map.mr_stgr); } static int audiosetinfo(sc, ai) struct audio_softc *sc; struct audio_info *ai; { struct audio_prinfo *r = &ai->record, *p = &ai->play; register int s, bsize; if (p->gain != ~0) ausetpgain(sc, p->gain); if (r->gain != ~0) ausetrgain(sc, r->gain); if (ai->monitor_gain != ~0) ausetmgain(sc, ai->monitor_gain); if (p->port == AUDIO_SPEAKER) { sc->sc_map.mr_mmr2 |= AMD_MMR2_LS; audio_setmmr2(sc->sc_au.au_amd, sc->sc_map.mr_mmr2); } else if (p->port == AUDIO_HEADPHONE) { sc->sc_map.mr_mmr2 &=~ AMD_MMR2_LS; audio_setmmr2(sc->sc_au.au_amd, sc->sc_map.mr_mmr2); } if (p->pause != (u_char)~0) sc->sc_au.au_wb.cb_pause = p->pause; if (r->pause != (u_char)~0) sc->sc_au.au_rb.cb_pause = r->pause; if (ai->blocksize != ~0) { if (ai->blocksize == 0) bsize = ai->blocksize = DEFBLKSIZE; else if (ai->blocksize > MAXBLKSIZE) bsize = ai->blocksize = MAXBLKSIZE; else bsize = ai->blocksize; s = splaudio(); sc->sc_au.au_blksize = bsize; /* AUDIO_FLUSH */ AUCB_INIT(&sc->sc_au.au_rb); AUCB_INIT(&sc->sc_au.au_wb); splx(s); } if (ai->hiwat != ~0 && (unsigned)ai->hiwat < AUCB_SIZE) sc->sc_au.au_hiwat = ai->hiwat; if (ai->lowat != ~0 && ai->lowat < AUCB_SIZE) sc->sc_au.au_lowat = ai->lowat; if (ai->backlog != ~0 && ai->backlog < (AUCB_SIZE/2)) sc->sc_au.au_backlog = ai->backlog; return (0); } static int sunaudiosetinfo(sc, ai) struct audio_softc *sc; struct sun_audio_info *ai; { struct sun_audio_prinfo *r = &ai->record, *p = &ai->play; if (p->gain != ~0) ausetpgain(sc, p->gain); if (r->gain != ~0) ausetrgain(sc, r->gain); if (ai->monitor_gain != ~0) ausetmgain(sc, ai->monitor_gain); if (p->port == AUDIO_SPEAKER) { sc->sc_map.mr_mmr2 |= AMD_MMR2_LS; audio_setmmr2(sc->sc_au.au_amd, sc->sc_map.mr_mmr2); } else if (p->port == AUDIO_HEADPHONE) { sc->sc_map.mr_mmr2 &=~ AMD_MMR2_LS; audio_setmmr2(sc->sc_au.au_amd, sc->sc_map.mr_mmr2); } /* * The bsd driver does not distinguish between paused and active. * (In the sun driver, not active means samples are not ouput * at all, but paused means the last streams buffer is drained * and then output stops.) If either are 0, then when stop output. * Otherwise, if either are non-zero, we resume. */ if (p->pause == 0 || p->active == 0) sc->sc_au.au_wb.cb_pause = 0; else if (p->pause != (u_char)~0 || p->active != (u_char)~0) sc->sc_au.au_wb.cb_pause = 1; if (r->pause == 0 || r->active == 0) sc->sc_au.au_rb.cb_pause = 0; else if (r->pause != (u_char)~0 || r->active != (u_char)~0) sc->sc_au.au_rb.cb_pause = 1; return (0); } static int audiogetinfo(sc, ai) struct audio_softc *sc; struct audio_info *ai; { struct audio_prinfo *r = &ai->record, *p = &ai->play; p->sample_rate = r->sample_rate = 8000; p->channels = r->channels = 1; p->precision = r->precision = 8; p->encoding = r->encoding = AUDIO_ENCODING_ULAW; ai->monitor_gain = sc->sc_mlevel; r->gain = sc->sc_rlevel; p->gain = sc->sc_plevel; r->port = 1; p->port = (sc->sc_map.mr_mmr2 & AMD_MMR2_LS) ? AUDIO_SPEAKER : AUDIO_HEADPHONE; p->pause = sc->sc_au.au_wb.cb_pause; r->pause = sc->sc_au.au_rb.cb_pause; p->error = sc->sc_au.au_wb.cb_drops != 0; r->error = sc->sc_au.au_rb.cb_drops != 0; p->open = sc->sc_open; r->open = sc->sc_open; p->samples = sc->sc_au.au_stamp - sc->sc_au.au_wb.cb_pdrops; r->samples = sc->sc_au.au_stamp - sc->sc_au.au_rb.cb_pdrops; p->seek = sc->sc_wseek; r->seek = sc->sc_rseek; ai->blocksize = sc->sc_au.au_blksize; ai->hiwat = sc->sc_au.au_hiwat; ai->lowat = sc->sc_au.au_lowat; ai->backlog = sc->sc_au.au_backlog; return (0); } static int sunaudiogetinfo(sc, ai) struct audio_softc *sc; struct sun_audio_info *ai; { struct sun_audio_prinfo *r = &ai->record, *p = &ai->play; p->sample_rate = r->sample_rate = 8000; p->channels = r->channels = 1; p->precision = r->precision = 8; p->encoding = r->encoding = AUDIO_ENCODING_ULAW; ai->monitor_gain = sc->sc_mlevel; r->gain = sc->sc_rlevel; p->gain = sc->sc_plevel; r->port = 1; p->port = (sc->sc_map.mr_mmr2 & AMD_MMR2_LS) ? AUDIO_SPEAKER : AUDIO_HEADPHONE; p->active = p->pause = sc->sc_au.au_wb.cb_pause; r->active = r->pause = sc->sc_au.au_rb.cb_pause; p->error = sc->sc_au.au_wb.cb_drops != 0; r->error = sc->sc_au.au_rb.cb_drops != 0; p->waiting = 0; r->waiting = 0; p->eof = 0; r->eof = 0; p->open = sc->sc_open; r->open = sc->sc_open; p->samples = sc->sc_au.au_stamp - sc->sc_au.au_wb.cb_pdrops; r->samples = sc->sc_au.au_stamp - sc->sc_au.au_rb.cb_pdrops; return (0); } #endif