freebsd-dev/sys/dev/sound/pci/als4000.c
Ariff Abdullah 90da2b2859 Sound Mega-commit. Expect further cleanup until code freeze.
For a slightly thorough explaination, please refer to
	[1] http://people.freebsd.org/~ariff/SOUND_4.TXT.html .

Summary of changes includes:

1 Volume Per-Channel (vpc).  Provides private / standalone volume control
  unique per-stream pcm channel without touching master volume / pcm.
  Applications can directly use SNDCTL_DSP_[GET|SET][PLAY|REC]VOL, or for
  backwards compatibility, SOUND_MIXER_PCM through the opened dsp device
  instead of /dev/mixer.  Special "bypass" mode is enabled through
  /dev/mixer which will automatically detect if the adjustment is made
  through /dev/mixer and forward its request to this private volume
  controller.  Changes to this volume object will not interfere with
  other channels.

  Requirements:
    - SNDCTL_DSP_[GET|SET][PLAY|REC]_VOL are newer ioctls (OSSv4) which
      require specific application modifications (preferred).
    - No modifications required for using bypass mode, so applications
      like mplayer or xmms should work out of the box.

  Kernel hints:
    - hint.pcm.%d.vpc (0 = disable vpc).

  Kernel sysctls:
    - hw.snd.vpc_mixer_bypass (default: 1).  Enable or disable /dev/mixer
      bypass mode.
    - hw.snd.vpc_autoreset (default: 1).  By default, closing/opening
      /dev/dsp will reset the volume back to 0 db gain/attenuation.
      Setting this to 0 will preserve its settings across device
      closing/opening.
    - hw.snd.vpc_reset (default: 0).  Panic/reset button to reset all
      volume settings back to 0 db.
    - hw.snd.vpc_0db (default: 45).  0 db relative to linear mixer value.

2 High quality fixed-point Bandlimited SINC sampling rate converter,
  based on Julius O'Smith's Digital Audio Resampling -
  http://ccrma.stanford.edu/~jos/resample/.  It includes a filter design
  script written in awk (the clumsiest joke I've ever written)
    - 100% 32bit fixed-point, 64bit accumulator.
    - Possibly among the fastest (if not fastest) of its kind.
    - Resampling quality is tunable, either runtime or during kernel
      compilation (FEEDER_RATE_PRESETS).
    - Quality can be further customized during kernel compilation by
      defining FEEDER_RATE_PRESETS in /etc/make.conf.

  Kernel sysctls:
    - hw.snd.feeder_rate_quality.
      0 - Zero-order Hold (ZOH).  Fastest, bad quality.
      1 - Linear Interpolation (LINEAR).  Slightly slower than ZOH,
          better quality but still does not eliminate aliasing.
      2 - (and above) - Sinc Interpolation(SINC).  Best quality.  SINC
          quality always start from 2 and above.

  Rough quality comparisons:
    - http://people.freebsd.org/~ariff/z_comparison/

3 Bit-perfect mode.  Bypasses all feeder/dsp effects.  Pure sound will be
  directly fed into the hardware.

4 Parametric (compile time) Software Equalizer (Bass/Treble mixer). Can
  be customized by defining FEEDER_EQ_PRESETS in /etc/make.conf.

5 Transparent/Adaptive Virtual Channel. Now you don't have to disable
  vchans in order to make digital format pass through.  It also makes
  vchans more dynamic by choosing a better format/rate among all the
  concurrent streams, which means that dev.pcm.X.play.vchanformat/rate
  becomes sort of optional.

6 Exclusive Stream, with special open() mode O_EXCL.  This will "mute"
  other concurrent vchan streams and only allow a single channel with
  O_EXCL set to keep producing sound.

Other Changes:
    * most feeder_* stuffs are compilable in userland. Let's not
      speculate whether we should go all out for it (save that for
      FreeBSD 16.0-RELEASE).
    * kobj signature fixups, thanks to Andriy Gapon <avg@freebsd.org>
    * pull out channel mixing logic out of vchan.c and create its own
      feeder_mixer for world justice.
    * various refactoring here and there, for good or bad.
    * activation of few more OSSv4 ioctls() (see [1] above).
    * opt_snd.h for possible compile time configuration:
      (mostly for debugging purposes, don't try these at home)
        SND_DEBUG
        SND_DIAGNOSTIC
        SND_FEEDER_MULTIFORMAT
        SND_FEEDER_FULL_MULTIFORMAT
        SND_FEEDER_RATE_HP
        SND_PCM_64
        SND_OLDSTEREO

Manual page updates are on the way.

Tested by:	joel, Olivier SMEDTS <olivier at gid0 d org>, too many
          	unsung / unnamed heroes.
2009-06-07 19:12:08 +00:00

955 lines
23 KiB
C

/*-
* Copyright (c) 2001 Orion Hodson <oho@acm.org>
* 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.
*
* THIS SOFTWARE IS PROVIDED BY THE AUTHOR 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 AUTHOR 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, WHETHERIN CONTRACT, STRICT
* LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY
* OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THEPOSSIBILITY OF
* SUCH DAMAGE.
*/
/*
* als4000.c - driver for the Avance Logic ALS 4000 chipset.
*
* The ALS4000 is effectively an SB16 with a PCI interface.
*
* This driver derives from ALS4000a.PDF, Bart Hartgers alsa driver, and
* SB16 register descriptions.
*/
#ifdef HAVE_KERNEL_OPTION_HEADERS
#include "opt_snd.h"
#endif
#include <dev/sound/pcm/sound.h>
#include <dev/sound/isa/sb.h>
#include <dev/sound/pci/als4000.h>
#include <dev/pci/pcireg.h>
#include <dev/pci/pcivar.h>
#include "mixer_if.h"
SND_DECLARE_FILE("$FreeBSD$");
/* Debugging macro's */
#undef DEB
#ifndef DEB
#define DEB(x) /* x */
#endif /* DEB */
#define ALS_DEFAULT_BUFSZ 16384
/* ------------------------------------------------------------------------- */
/* Structures */
struct sc_info;
struct sc_chinfo {
struct sc_info *parent;
struct pcm_channel *channel;
struct snd_dbuf *buffer;
u_int32_t format, speed, phys_buf, bps;
u_int32_t dma_active:1, dma_was_active:1;
u_int8_t gcr_fifo_status;
int dir;
};
struct sc_info {
device_t dev;
bus_space_tag_t st;
bus_space_handle_t sh;
bus_dma_tag_t parent_dmat;
struct resource *reg, *irq;
int regid, irqid;
void *ih;
struct mtx *lock;
unsigned int bufsz;
struct sc_chinfo pch, rch;
};
/* Channel caps */
static u_int32_t als_format[] = {
SND_FORMAT(AFMT_U8, 1, 0),
SND_FORMAT(AFMT_U8, 2, 0),
SND_FORMAT(AFMT_S16_LE, 1, 0),
SND_FORMAT(AFMT_S16_LE, 2, 0),
0
};
/*
* I don't believe this rotten soundcard can do 48k, really,
* trust me.
*/
static struct pcmchan_caps als_caps = { 4000, 44100, als_format, 0 };
/* ------------------------------------------------------------------------- */
/* Register Utilities */
static u_int32_t
als_gcr_rd(struct sc_info *sc, int index)
{
bus_space_write_1(sc->st, sc->sh, ALS_GCR_INDEX, index);
return bus_space_read_4(sc->st, sc->sh, ALS_GCR_DATA);
}
static void
als_gcr_wr(struct sc_info *sc, int index, int data)
{
bus_space_write_1(sc->st, sc->sh, ALS_GCR_INDEX, index);
bus_space_write_4(sc->st, sc->sh, ALS_GCR_DATA, data);
}
static u_int8_t
als_intr_rd(struct sc_info *sc)
{
return bus_space_read_1(sc->st, sc->sh, ALS_SB_MPU_IRQ);
}
static void
als_intr_wr(struct sc_info *sc, u_int8_t data)
{
bus_space_write_1(sc->st, sc->sh, ALS_SB_MPU_IRQ, data);
}
static u_int8_t
als_mix_rd(struct sc_info *sc, u_int8_t index)
{
bus_space_write_1(sc->st, sc->sh, ALS_MIXER_INDEX, index);
return bus_space_read_1(sc->st, sc->sh, ALS_MIXER_DATA);
}
static void
als_mix_wr(struct sc_info *sc, u_int8_t index, u_int8_t data)
{
bus_space_write_1(sc->st, sc->sh, ALS_MIXER_INDEX, index);
bus_space_write_1(sc->st, sc->sh, ALS_MIXER_DATA, data);
}
static void
als_esp_wr(struct sc_info *sc, u_int8_t data)
{
u_int32_t tries, v;
tries = 1000;
do {
v = bus_space_read_1(sc->st, sc->sh, ALS_ESP_WR_STATUS);
if (~v & 0x80)
break;
DELAY(20);
} while (--tries != 0);
if (tries == 0)
device_printf(sc->dev, "als_esp_wr timeout");
bus_space_write_1(sc->st, sc->sh, ALS_ESP_WR_DATA, data);
}
static int
als_esp_reset(struct sc_info *sc)
{
u_int32_t tries, u, v;
bus_space_write_1(sc->st, sc->sh, ALS_ESP_RST, 1);
DELAY(10);
bus_space_write_1(sc->st, sc->sh, ALS_ESP_RST, 0);
DELAY(30);
tries = 1000;
do {
u = bus_space_read_1(sc->st, sc->sh, ALS_ESP_RD_STATUS8);
if (u & 0x80) {
v = bus_space_read_1(sc->st, sc->sh, ALS_ESP_RD_DATA);
if (v == 0xaa)
return 0;
else
break;
}
DELAY(20);
} while (--tries != 0);
if (tries == 0)
device_printf(sc->dev, "als_esp_reset timeout");
return 1;
}
static u_int8_t
als_ack_read(struct sc_info *sc, u_int8_t addr)
{
u_int8_t r = bus_space_read_1(sc->st, sc->sh, addr);
return r;
}
/* ------------------------------------------------------------------------- */
/* Common pcm channel implementation */
static void *
alschan_init(kobj_t obj, void *devinfo,
struct snd_dbuf *b, struct pcm_channel *c, int dir)
{
struct sc_info *sc = devinfo;
struct sc_chinfo *ch;
snd_mtxlock(sc->lock);
if (dir == PCMDIR_PLAY) {
ch = &sc->pch;
ch->gcr_fifo_status = ALS_GCR_FIFO0_STATUS;
} else {
ch = &sc->rch;
ch->gcr_fifo_status = ALS_GCR_FIFO1_STATUS;
}
ch->dir = dir;
ch->parent = sc;
ch->channel = c;
ch->bps = 1;
ch->format = SND_FORMAT(AFMT_U8, 1, 0);
ch->speed = DSP_DEFAULT_SPEED;
ch->buffer = b;
snd_mtxunlock(sc->lock);
if (sndbuf_alloc(ch->buffer, sc->parent_dmat, 0, sc->bufsz) != 0)
return NULL;
return ch;
}
static int
alschan_setformat(kobj_t obj, void *data, u_int32_t format)
{
struct sc_chinfo *ch = data;
ch->format = format;
return 0;
}
static u_int32_t
alschan_setspeed(kobj_t obj, void *data, u_int32_t speed)
{
struct sc_chinfo *ch = data, *other;
struct sc_info *sc = ch->parent;
other = (ch->dir == PCMDIR_PLAY) ? &sc->rch : &sc->pch;
/* Deny request if other dma channel is active */
if (other->dma_active) {
ch->speed = other->speed;
return other->speed;
}
ch->speed = speed;
return speed;
}
static u_int32_t
alschan_setblocksize(kobj_t obj, void *data, u_int32_t blocksize)
{
struct sc_chinfo *ch = data;
struct sc_info *sc = ch->parent;
if (blocksize > sc->bufsz / 2) {
blocksize = sc->bufsz / 2;
}
sndbuf_resize(ch->buffer, 2, blocksize);
return blocksize;
}
static u_int32_t
alschan_getptr(kobj_t obj, void *data)
{
struct sc_chinfo *ch = data;
struct sc_info *sc = ch->parent;
int32_t pos, sz;
snd_mtxlock(sc->lock);
pos = als_gcr_rd(ch->parent, ch->gcr_fifo_status) & 0xffff;
snd_mtxunlock(sc->lock);
sz = sndbuf_getsize(ch->buffer);
return (2 * sz - pos - 1) % sz;
}
static struct pcmchan_caps*
alschan_getcaps(kobj_t obj, void *data)
{
return &als_caps;
}
static void
als_set_speed(struct sc_chinfo *ch)
{
struct sc_info *sc = ch->parent;
struct sc_chinfo *other;
other = (ch->dir == PCMDIR_PLAY) ? &sc->rch : &sc->pch;
if (other->dma_active == 0) {
als_esp_wr(sc, ALS_ESP_SAMPLE_RATE);
als_esp_wr(sc, ch->speed >> 8);
als_esp_wr(sc, ch->speed & 0xff);
} else {
DEB(printf("speed locked at %d (tried %d)\n",
other->speed, ch->speed));
}
}
/* ------------------------------------------------------------------------- */
/* Playback channel implementation */
#define ALS_8BIT_CMD(x, y) { (x), (y), DSP_DMA8, DSP_CMD_DMAPAUSE_8 }
#define ALS_16BIT_CMD(x, y) { (x), (y), DSP_DMA16, DSP_CMD_DMAPAUSE_16 }
struct playback_command {
u_int32_t pcm_format; /* newpcm format */
u_int8_t format_val; /* sb16 format value */
u_int8_t dma_prog; /* sb16 dma program */
u_int8_t dma_stop; /* sb16 stop register */
} static const playback_cmds[] = {
ALS_8BIT_CMD(SND_FORMAT(AFMT_U8, 1, 0), DSP_MODE_U8MONO),
ALS_8BIT_CMD(SND_FORMAT(AFMT_U8, 2, 0), DSP_MODE_U8STEREO),
ALS_16BIT_CMD(SND_FORMAT(AFMT_S16_LE, 1, 0), DSP_MODE_S16MONO),
ALS_16BIT_CMD(SND_FORMAT(AFMT_S16_LE, 2, 0), DSP_MODE_S16STEREO),
};
static const struct playback_command*
als_get_playback_command(u_int32_t format)
{
u_int32_t i, n;
n = sizeof(playback_cmds) / sizeof(playback_cmds[0]);
for (i = 0; i < n; i++) {
if (playback_cmds[i].pcm_format == format) {
return &playback_cmds[i];
}
}
DEB(printf("als_get_playback_command: invalid format 0x%08x\n",
format));
return &playback_cmds[0];
}
static void
als_playback_start(struct sc_chinfo *ch)
{
const struct playback_command *p;
struct sc_info *sc = ch->parent;
u_int32_t buf, bufsz, count, dma_prog;
buf = sndbuf_getbufaddr(ch->buffer);
bufsz = sndbuf_getsize(ch->buffer);
count = bufsz / 2;
if (ch->format & AFMT_16BIT)
count /= 2;
count--;
als_esp_wr(sc, DSP_CMD_SPKON);
als_set_speed(ch);
als_gcr_wr(sc, ALS_GCR_DMA0_START, buf);
als_gcr_wr(sc, ALS_GCR_DMA0_MODE, (bufsz - 1) | 0x180000);
p = als_get_playback_command(ch->format);
dma_prog = p->dma_prog | DSP_F16_DAC | DSP_F16_AUTO | DSP_F16_FIFO_ON;
als_esp_wr(sc, dma_prog);
als_esp_wr(sc, p->format_val);
als_esp_wr(sc, count & 0xff);
als_esp_wr(sc, count >> 8);
ch->dma_active = 1;
}
static int
als_playback_stop(struct sc_chinfo *ch)
{
const struct playback_command *p;
struct sc_info *sc = ch->parent;
u_int32_t active;
active = ch->dma_active;
if (active) {
p = als_get_playback_command(ch->format);
als_esp_wr(sc, p->dma_stop);
}
ch->dma_active = 0;
return active;
}
static int
alspchan_trigger(kobj_t obj, void *data, int go)
{
struct sc_chinfo *ch = data;
struct sc_info *sc = ch->parent;
if (!PCMTRIG_COMMON(go))
return 0;
snd_mtxlock(sc->lock);
switch(go) {
case PCMTRIG_START:
als_playback_start(ch);
break;
case PCMTRIG_STOP:
case PCMTRIG_ABORT:
als_playback_stop(ch);
break;
default:
break;
}
snd_mtxunlock(sc->lock);
return 0;
}
static kobj_method_t alspchan_methods[] = {
KOBJMETHOD(channel_init, alschan_init),
KOBJMETHOD(channel_setformat, alschan_setformat),
KOBJMETHOD(channel_setspeed, alschan_setspeed),
KOBJMETHOD(channel_setblocksize, alschan_setblocksize),
KOBJMETHOD(channel_trigger, alspchan_trigger),
KOBJMETHOD(channel_getptr, alschan_getptr),
KOBJMETHOD(channel_getcaps, alschan_getcaps),
KOBJMETHOD_END
};
CHANNEL_DECLARE(alspchan);
/* ------------------------------------------------------------------------- */
/* Capture channel implementation */
static u_int8_t
als_get_fifo_format(struct sc_info *sc, u_int32_t format)
{
switch (format) {
case SND_FORMAT(AFMT_U8, 1, 0):
return ALS_FIFO1_8BIT;
case SND_FORMAT(AFMT_U8, 2, 0):
return ALS_FIFO1_8BIT | ALS_FIFO1_STEREO;
case SND_FORMAT(AFMT_S16_LE, 1, 0):
return ALS_FIFO1_SIGNED;
case SND_FORMAT(AFMT_S16_LE, 2, 0):
return ALS_FIFO1_SIGNED | ALS_FIFO1_STEREO;
}
device_printf(sc->dev, "format not found: 0x%08x\n", format);
return ALS_FIFO1_8BIT;
}
static void
als_capture_start(struct sc_chinfo *ch)
{
struct sc_info *sc = ch->parent;
u_int32_t buf, bufsz, count, dma_prog;
buf = sndbuf_getbufaddr(ch->buffer);
bufsz = sndbuf_getsize(ch->buffer);
count = bufsz / 2;
if (ch->format & AFMT_16BIT)
count /= 2;
count--;
als_esp_wr(sc, DSP_CMD_SPKON);
als_set_speed(ch);
als_gcr_wr(sc, ALS_GCR_FIFO1_START, buf);
als_gcr_wr(sc, ALS_GCR_FIFO1_COUNT, (bufsz - 1));
als_mix_wr(sc, ALS_FIFO1_LENGTH_LO, count & 0xff);
als_mix_wr(sc, ALS_FIFO1_LENGTH_HI, count >> 8);
dma_prog = ALS_FIFO1_RUN | als_get_fifo_format(sc, ch->format);
als_mix_wr(sc, ALS_FIFO1_CONTROL, dma_prog);
ch->dma_active = 1;
}
static int
als_capture_stop(struct sc_chinfo *ch)
{
struct sc_info *sc = ch->parent;
u_int32_t active;
active = ch->dma_active;
if (active) {
als_mix_wr(sc, ALS_FIFO1_CONTROL, ALS_FIFO1_STOP);
}
ch->dma_active = 0;
return active;
}
static int
alsrchan_trigger(kobj_t obj, void *data, int go)
{
struct sc_chinfo *ch = data;
struct sc_info *sc = ch->parent;
snd_mtxlock(sc->lock);
switch(go) {
case PCMTRIG_START:
als_capture_start(ch);
break;
case PCMTRIG_STOP:
case PCMTRIG_ABORT:
als_capture_stop(ch);
break;
}
snd_mtxunlock(sc->lock);
return 0;
}
static kobj_method_t alsrchan_methods[] = {
KOBJMETHOD(channel_init, alschan_init),
KOBJMETHOD(channel_setformat, alschan_setformat),
KOBJMETHOD(channel_setspeed, alschan_setspeed),
KOBJMETHOD(channel_setblocksize, alschan_setblocksize),
KOBJMETHOD(channel_trigger, alsrchan_trigger),
KOBJMETHOD(channel_getptr, alschan_getptr),
KOBJMETHOD(channel_getcaps, alschan_getcaps),
KOBJMETHOD_END
};
CHANNEL_DECLARE(alsrchan);
/* ------------------------------------------------------------------------- */
/* Mixer related */
/*
* ALS4000 has an sb16 mixer, with some additional controls that we do
* not yet a means to support.
*/
struct sb16props {
u_int8_t lreg;
u_int8_t rreg;
u_int8_t bits;
u_int8_t oselect;
u_int8_t iselect; /* left input mask */
} static const amt[SOUND_MIXER_NRDEVICES] = {
[SOUND_MIXER_VOLUME] = { 0x30, 0x31, 5, 0x00, 0x00 },
[SOUND_MIXER_PCM] = { 0x32, 0x33, 5, 0x00, 0x00 },
[SOUND_MIXER_SYNTH] = { 0x34, 0x35, 5, 0x60, 0x40 },
[SOUND_MIXER_CD] = { 0x36, 0x37, 5, 0x06, 0x04 },
[SOUND_MIXER_LINE] = { 0x38, 0x39, 5, 0x18, 0x10 },
[SOUND_MIXER_MIC] = { 0x3a, 0x00, 5, 0x01, 0x01 },
[SOUND_MIXER_SPEAKER] = { 0x3b, 0x00, 2, 0x00, 0x00 },
[SOUND_MIXER_IGAIN] = { 0x3f, 0x40, 2, 0x00, 0x00 },
[SOUND_MIXER_OGAIN] = { 0x41, 0x42, 2, 0x00, 0x00 },
/* The following have register values but no h/w implementation */
[SOUND_MIXER_TREBLE] = { 0x44, 0x45, 4, 0x00, 0x00 },
[SOUND_MIXER_BASS] = { 0x46, 0x47, 4, 0x00, 0x00 }
};
static int
alsmix_init(struct snd_mixer *m)
{
u_int32_t i, v;
for (i = v = 0; i < SOUND_MIXER_NRDEVICES; i++) {
if (amt[i].bits) v |= 1 << i;
}
mix_setdevs(m, v);
for (i = v = 0; i < SOUND_MIXER_NRDEVICES; i++) {
if (amt[i].iselect) v |= 1 << i;
}
mix_setrecdevs(m, v);
return 0;
}
static int
alsmix_set(struct snd_mixer *m, unsigned dev, unsigned left, unsigned right)
{
struct sc_info *sc = mix_getdevinfo(m);
u_int32_t r, l, v, mask;
/* Fill upper n bits in mask with 1's */
mask = ((1 << amt[dev].bits) - 1) << (8 - amt[dev].bits);
l = (left * mask / 100) & mask;
v = als_mix_rd(sc, amt[dev].lreg) & ~mask;
als_mix_wr(sc, amt[dev].lreg, l | v);
if (amt[dev].rreg) {
r = (right * mask / 100) & mask;
v = als_mix_rd(sc, amt[dev].rreg) & ~mask;
als_mix_wr(sc, amt[dev].rreg, r | v);
} else {
r = 0;
}
/* Zero gain does not mute channel from output, but this does. */
v = als_mix_rd(sc, SB16_OMASK);
if (l == 0 && r == 0) {
v &= ~amt[dev].oselect;
} else {
v |= amt[dev].oselect;
}
als_mix_wr(sc, SB16_OMASK, v);
return 0;
}
static u_int32_t
alsmix_setrecsrc(struct snd_mixer *m, u_int32_t src)
{
struct sc_info *sc = mix_getdevinfo(m);
u_int32_t i, l, r;
for (i = l = r = 0; i < SOUND_MIXER_NRDEVICES; i++) {
if (src & (1 << i)) {
if (amt[i].iselect == 1) { /* microphone */
l |= amt[i].iselect;
r |= amt[i].iselect;
} else {
l |= amt[i].iselect;
r |= amt[i].iselect >> 1;
}
}
}
als_mix_wr(sc, SB16_IMASK_L, l);
als_mix_wr(sc, SB16_IMASK_R, r);
return src;
}
static kobj_method_t als_mixer_methods[] = {
KOBJMETHOD(mixer_init, alsmix_init),
KOBJMETHOD(mixer_set, alsmix_set),
KOBJMETHOD(mixer_setrecsrc, alsmix_setrecsrc),
KOBJMETHOD_END
};
MIXER_DECLARE(als_mixer);
/* ------------------------------------------------------------------------- */
/* Interrupt Handler */
static void
als_intr(void *p)
{
struct sc_info *sc = (struct sc_info *)p;
u_int8_t intr, sb_status;
snd_mtxlock(sc->lock);
intr = als_intr_rd(sc);
if (intr & 0x80) {
snd_mtxunlock(sc->lock);
chn_intr(sc->pch.channel);
snd_mtxlock(sc->lock);
}
if (intr & 0x40) {
snd_mtxunlock(sc->lock);
chn_intr(sc->rch.channel);
snd_mtxlock(sc->lock);
}
/* ACK interrupt in PCI core */
als_intr_wr(sc, intr);
/* ACK interrupt in SB core */
sb_status = als_mix_rd(sc, IRQ_STAT);
if (sb_status & ALS_IRQ_STATUS8)
als_ack_read(sc, ALS_ESP_RD_STATUS8);
if (sb_status & ALS_IRQ_STATUS16)
als_ack_read(sc, ALS_ESP_RD_STATUS16);
if (sb_status & ALS_IRQ_MPUIN)
als_ack_read(sc, ALS_MIDI_DATA);
if (sb_status & ALS_IRQ_CR1E)
als_ack_read(sc, ALS_CR1E_ACK_PORT);
snd_mtxunlock(sc->lock);
return;
}
/* ------------------------------------------------------------------------- */
/* H/W initialization */
static int
als_init(struct sc_info *sc)
{
u_int32_t i, v;
/* Reset Chip */
if (als_esp_reset(sc)) {
return 1;
}
/* Enable write on DMA_SETUP register */
v = als_mix_rd(sc, ALS_SB16_CONFIG);
als_mix_wr(sc, ALS_SB16_CONFIG, v | 0x80);
/* Select DMA0 */
als_mix_wr(sc, ALS_SB16_DMA_SETUP, 0x01);
/* Disable write on DMA_SETUP register */
als_mix_wr(sc, ALS_SB16_CONFIG, v & 0x7f);
/* Enable interrupts */
v = als_gcr_rd(sc, ALS_GCR_MISC);
als_gcr_wr(sc, ALS_GCR_MISC, v | 0x28000);
/* Black out GCR DMA registers */
for (i = 0x91; i <= 0x96; i++) {
als_gcr_wr(sc, i, 0);
}
/* Emulation mode */
v = als_gcr_rd(sc, ALS_GCR_DMA_EMULATION);
als_gcr_wr(sc, ALS_GCR_DMA_EMULATION, v);
DEB(printf("GCR_DMA_EMULATION 0x%08x\n", v));
return 0;
}
static void
als_uninit(struct sc_info *sc)
{
/* Disable interrupts */
als_gcr_wr(sc, ALS_GCR_MISC, 0);
}
/* ------------------------------------------------------------------------- */
/* Probe and attach card */
static int
als_pci_probe(device_t dev)
{
if (pci_get_devid(dev) == ALS_PCI_ID0) {
device_set_desc(dev, "Avance Logic ALS4000");
return BUS_PROBE_DEFAULT;
}
return ENXIO;
}
static void
als_resource_free(device_t dev, struct sc_info *sc)
{
if (sc->reg) {
bus_release_resource(dev, SYS_RES_IOPORT, sc->regid, sc->reg);
sc->reg = 0;
}
if (sc->ih) {
bus_teardown_intr(dev, sc->irq, sc->ih);
sc->ih = 0;
}
if (sc->irq) {
bus_release_resource(dev, SYS_RES_IRQ, sc->irqid, sc->irq);
sc->irq = 0;
}
if (sc->parent_dmat) {
bus_dma_tag_destroy(sc->parent_dmat);
sc->parent_dmat = 0;
}
if (sc->lock) {
snd_mtxfree(sc->lock);
sc->lock = NULL;
}
}
static int
als_resource_grab(device_t dev, struct sc_info *sc)
{
sc->regid = PCIR_BAR(0);
sc->reg = bus_alloc_resource(dev, SYS_RES_IOPORT, &sc->regid, 0, ~0,
ALS_CONFIG_SPACE_BYTES, RF_ACTIVE);
if (sc->reg == 0) {
device_printf(dev, "unable to allocate register space\n");
goto bad;
}
sc->st = rman_get_bustag(sc->reg);
sc->sh = rman_get_bushandle(sc->reg);
sc->irq = bus_alloc_resource_any(dev, SYS_RES_IRQ, &sc->irqid,
RF_ACTIVE | RF_SHAREABLE);
if (sc->irq == 0) {
device_printf(dev, "unable to allocate interrupt\n");
goto bad;
}
if (snd_setup_intr(dev, sc->irq, INTR_MPSAFE, als_intr,
sc, &sc->ih)) {
device_printf(dev, "unable to setup interrupt\n");
goto bad;
}
sc->bufsz = pcm_getbuffersize(dev, 4096, ALS_DEFAULT_BUFSZ, 65536);
if (bus_dma_tag_create(/*parent*/bus_get_dma_tag(dev),
/*alignment*/2, /*boundary*/0,
/*lowaddr*/BUS_SPACE_MAXADDR_24BIT,
/*highaddr*/BUS_SPACE_MAXADDR,
/*filter*/NULL, /*filterarg*/NULL,
/*maxsize*/sc->bufsz,
/*nsegments*/1, /*maxsegz*/0x3ffff,
/*flags*/0, /*lockfunc*/NULL,
/*lockarg*/NULL, &sc->parent_dmat) != 0) {
device_printf(dev, "unable to create dma tag\n");
goto bad;
}
return 0;
bad:
als_resource_free(dev, sc);
return ENXIO;
}
static int
als_pci_attach(device_t dev)
{
struct sc_info *sc;
u_int32_t data;
char status[SND_STATUSLEN];
sc = malloc(sizeof(*sc), M_DEVBUF, M_WAITOK | M_ZERO);
sc->lock = snd_mtxcreate(device_get_nameunit(dev), "snd_als4000 softc");
sc->dev = dev;
data = pci_read_config(dev, PCIR_COMMAND, 2);
data |= (PCIM_CMD_PORTEN | PCIM_CMD_MEMEN | PCIM_CMD_BUSMASTEREN);
pci_write_config(dev, PCIR_COMMAND, data, 2);
/*
* By default the power to the various components on the
* ALS4000 is entirely controlled by the pci powerstate. We
* could attempt finer grained control by setting GCR6.31.
*/
#if __FreeBSD_version > 500000
if (pci_get_powerstate(dev) != PCI_POWERSTATE_D0) {
/* Reset the power state. */
device_printf(dev, "chip is in D%d power mode "
"-- setting to D0\n", pci_get_powerstate(dev));
pci_set_powerstate(dev, PCI_POWERSTATE_D0);
}
#else
data = pci_read_config(dev, ALS_PCI_POWERREG, 2);
if ((data & 0x03) != 0) {
device_printf(dev, "chip is in D%d power mode "
"-- setting to D0\n", data & 0x03);
data &= ~0x03;
pci_write_config(dev, ALS_PCI_POWERREG, data, 2);
}
#endif
if (als_resource_grab(dev, sc)) {
device_printf(dev, "failed to allocate resources\n");
goto bad_attach;
}
if (als_init(sc)) {
device_printf(dev, "failed to initialize hardware\n");
goto bad_attach;
}
if (mixer_init(dev, &als_mixer_class, sc)) {
device_printf(dev, "failed to initialize mixer\n");
goto bad_attach;
}
if (pcm_register(dev, sc, 1, 1)) {
device_printf(dev, "failed to register pcm entries\n");
goto bad_attach;
}
pcm_addchan(dev, PCMDIR_PLAY, &alspchan_class, sc);
pcm_addchan(dev, PCMDIR_REC, &alsrchan_class, sc);
snprintf(status, SND_STATUSLEN, "at io 0x%lx irq %ld %s",
rman_get_start(sc->reg), rman_get_start(sc->irq),PCM_KLDSTRING(snd_als4000));
pcm_setstatus(dev, status);
return 0;
bad_attach:
als_resource_free(dev, sc);
free(sc, M_DEVBUF);
return ENXIO;
}
static int
als_pci_detach(device_t dev)
{
struct sc_info *sc;
int r;
r = pcm_unregister(dev);
if (r)
return r;
sc = pcm_getdevinfo(dev);
als_uninit(sc);
als_resource_free(dev, sc);
free(sc, M_DEVBUF);
return 0;
}
static int
als_pci_suspend(device_t dev)
{
struct sc_info *sc = pcm_getdevinfo(dev);
snd_mtxlock(sc->lock);
sc->pch.dma_was_active = als_playback_stop(&sc->pch);
sc->rch.dma_was_active = als_capture_stop(&sc->rch);
als_uninit(sc);
snd_mtxunlock(sc->lock);
return 0;
}
static int
als_pci_resume(device_t dev)
{
struct sc_info *sc = pcm_getdevinfo(dev);
snd_mtxlock(sc->lock);
if (als_init(sc) != 0) {
device_printf(dev, "unable to reinitialize the card\n");
snd_mtxunlock(sc->lock);
return ENXIO;
}
if (mixer_reinit(dev) != 0) {
device_printf(dev, "unable to reinitialize the mixer\n");
snd_mtxunlock(sc->lock);
return ENXIO;
}
if (sc->pch.dma_was_active) {
als_playback_start(&sc->pch);
}
if (sc->rch.dma_was_active) {
als_capture_start(&sc->rch);
}
snd_mtxunlock(sc->lock);
return 0;
}
static device_method_t als_methods[] = {
/* Device interface */
DEVMETHOD(device_probe, als_pci_probe),
DEVMETHOD(device_attach, als_pci_attach),
DEVMETHOD(device_detach, als_pci_detach),
DEVMETHOD(device_suspend, als_pci_suspend),
DEVMETHOD(device_resume, als_pci_resume),
{ 0, 0 }
};
static driver_t als_driver = {
"pcm",
als_methods,
PCM_SOFTC_SIZE,
};
DRIVER_MODULE(snd_als4000, pci, als_driver, pcm_devclass, 0, 0);
MODULE_DEPEND(snd_als4000, sound, SOUND_MINVER, SOUND_PREFVER, SOUND_MAXVER);
MODULE_VERSION(snd_als4000, 1);