bsd_audio.c

早期freebsd实现

	}
	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