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DEFINITIONS
This source file includes following definitions.
- read_ad1843_reg
- write_ad1843_reg
- sgio2audio_gain_info
- sgio2audio_gain_get
- sgio2audio_gain_put
- sgio2audio_source_info
- sgio2audio_source_get
- sgio2audio_source_put
- snd_sgio2audio_new_mixer
- snd_sgio2audio_dma_pull_frag
- snd_sgio2audio_dma_push_frag
- snd_sgio2audio_dma_start
- snd_sgio2audio_dma_stop
- snd_sgio2audio_dma_in_isr
- snd_sgio2audio_dma_out_isr
- snd_sgio2audio_error_isr
- snd_sgio2audio_playback1_open
- snd_sgio2audio_playback2_open
- snd_sgio2audio_capture_open
- snd_sgio2audio_pcm_close
- snd_sgio2audio_pcm_hw_params
- snd_sgio2audio_pcm_hw_free
- snd_sgio2audio_pcm_prepare
- snd_sgio2audio_pcm_trigger
- snd_sgio2audio_pcm_pointer
- snd_sgio2audio_page
- snd_sgio2audio_new_pcm
- snd_sgio2audio_free
- snd_sgio2audio_dev_free
- snd_sgio2audio_create
- snd_sgio2audio_probe
- snd_sgio2audio_remove
- alsa_card_sgio2audio_init
- alsa_card_sgio2audio_exit
/*
* Sound driver for Silicon Graphics O2 Workstations A/V board audio.
*
* Copyright 2003 Vivien Chappelier <vivien.chappelier@linux-mips.org>
* Copyright 2008 Thomas Bogendoerfer <tsbogend@alpha.franken.de>
* Mxier part taken from mace_audio.c:
* Copyright 2007 Thorben Jändling <tj.trevelyan@gmail.com>
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; either version 2 of the License, or
* (at your option) any later version.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with this program; if not, write to the Free Software
* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
*
*/
#include <linux/init.h>
#include <linux/delay.h>
#include <linux/spinlock.h>
#include <linux/gfp.h>
#include <linux/vmalloc.h>
#include <linux/interrupt.h>
#include <linux/dma-mapping.h>
#include <linux/platform_device.h>
#include <linux/io.h>
#include <asm/ip32/ip32_ints.h>
#include <asm/ip32/mace.h>
#include <sound/core.h>
#include <sound/control.h>
#include <sound/pcm.h>
#define SNDRV_GET_ID
#include <sound/initval.h>
#include <sound/ad1843.h>
MODULE_AUTHOR("Vivien Chappelier <vivien.chappelier@linux-mips.org>");
MODULE_DESCRIPTION("SGI O2 Audio");
MODULE_LICENSE("GPL");
MODULE_SUPPORTED_DEVICE("{{Silicon Graphics, O2 Audio}}");
static int index = SNDRV_DEFAULT_IDX1; /* Index 0-MAX */
static char *id = SNDRV_DEFAULT_STR1; /* ID for this card */
module_param(index, int, 0444);
MODULE_PARM_DESC(index, "Index value for SGI O2 soundcard.");
module_param(id, charp, 0444);
MODULE_PARM_DESC(id, "ID string for SGI O2 soundcard.");
#define AUDIO_CONTROL_RESET BIT(0) /* 1: reset audio interface */
#define AUDIO_CONTROL_CODEC_PRESENT BIT(1) /* 1: codec detected */
#define CODEC_CONTROL_WORD_SHIFT 0
#define CODEC_CONTROL_READ BIT(16)
#define CODEC_CONTROL_ADDRESS_SHIFT 17
#define CHANNEL_CONTROL_RESET BIT(10) /* 1: reset channel */
#define CHANNEL_DMA_ENABLE BIT(9) /* 1: enable DMA transfer */
#define CHANNEL_INT_THRESHOLD_DISABLED (0 << 5) /* interrupt disabled */
#define CHANNEL_INT_THRESHOLD_25 (1 << 5) /* int on buffer >25% full */
#define CHANNEL_INT_THRESHOLD_50 (2 << 5) /* int on buffer >50% full */
#define CHANNEL_INT_THRESHOLD_75 (3 << 5) /* int on buffer >75% full */
#define CHANNEL_INT_THRESHOLD_EMPTY (4 << 5) /* int on buffer empty */
#define CHANNEL_INT_THRESHOLD_NOT_EMPTY (5 << 5) /* int on buffer !empty */
#define CHANNEL_INT_THRESHOLD_FULL (6 << 5) /* int on buffer empty */
#define CHANNEL_INT_THRESHOLD_NOT_FULL (7 << 5) /* int on buffer !empty */
#define CHANNEL_RING_SHIFT 12
#define CHANNEL_RING_SIZE (1 << CHANNEL_RING_SHIFT)
#define CHANNEL_RING_MASK (CHANNEL_RING_SIZE - 1)
#define CHANNEL_LEFT_SHIFT 40
#define CHANNEL_RIGHT_SHIFT 8
struct snd_sgio2audio_chan {
int idx;
struct snd_pcm_substream *substream;
int pos;
snd_pcm_uframes_t size;
spinlock_t lock;
};
/* definition of the chip-specific record */
struct snd_sgio2audio {
struct snd_card *card;
/* codec */
struct snd_ad1843 ad1843;
spinlock_t ad1843_lock;
/* channels */
struct snd_sgio2audio_chan channel[3];
/* resources */
void *ring_base;
dma_addr_t ring_base_dma;
};
/* AD1843 access */
/*
* read_ad1843_reg returns the current contents of a 16 bit AD1843 register.
*
* Returns unsigned register value on success, -errno on failure.
*/
static int read_ad1843_reg(void *priv, int reg)
{
struct snd_sgio2audio *chip = priv;
int val;
unsigned long flags;
spin_lock_irqsave(&chip->ad1843_lock, flags);
writeq((reg << CODEC_CONTROL_ADDRESS_SHIFT) |
CODEC_CONTROL_READ, &mace->perif.audio.codec_control);
wmb();
val = readq(&mace->perif.audio.codec_control); /* flush bus */
udelay(200);
val = readq(&mace->perif.audio.codec_read);
spin_unlock_irqrestore(&chip->ad1843_lock, flags);
return val;
}
/*
* write_ad1843_reg writes the specified value to a 16 bit AD1843 register.
*/
static int write_ad1843_reg(void *priv, int reg, int word)
{
struct snd_sgio2audio *chip = priv;
int val;
unsigned long flags;
spin_lock_irqsave(&chip->ad1843_lock, flags);
writeq((reg << CODEC_CONTROL_ADDRESS_SHIFT) |
(word << CODEC_CONTROL_WORD_SHIFT),
&mace->perif.audio.codec_control);
wmb();
val = readq(&mace->perif.audio.codec_control); /* flush bus */
udelay(200);
spin_unlock_irqrestore(&chip->ad1843_lock, flags);
return 0;
}
static int sgio2audio_gain_info(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
struct snd_sgio2audio *chip = snd_kcontrol_chip(kcontrol);
uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
uinfo->count = 2;
uinfo->value.integer.min = 0;
uinfo->value.integer.max = ad1843_get_gain_max(&chip->ad1843,
(int)kcontrol->private_value);
return 0;
}
static int sgio2audio_gain_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_sgio2audio *chip = snd_kcontrol_chip(kcontrol);
int vol;
vol = ad1843_get_gain(&chip->ad1843, (int)kcontrol->private_value);
ucontrol->value.integer.value[0] = (vol >> 8) & 0xFF;
ucontrol->value.integer.value[1] = vol & 0xFF;
return 0;
}
static int sgio2audio_gain_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_sgio2audio *chip = snd_kcontrol_chip(kcontrol);
int newvol, oldvol;
oldvol = ad1843_get_gain(&chip->ad1843, kcontrol->private_value);
newvol = (ucontrol->value.integer.value[0] << 8) |
ucontrol->value.integer.value[1];
newvol = ad1843_set_gain(&chip->ad1843, kcontrol->private_value,
newvol);
return newvol != oldvol;
}
static int sgio2audio_source_info(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
static const char *texts[3] = {
"Cam Mic", "Mic", "Line"
};
uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
uinfo->count = 1;
uinfo->value.enumerated.items = 3;
if (uinfo->value.enumerated.item >= 3)
uinfo->value.enumerated.item = 1;
strcpy(uinfo->value.enumerated.name,
texts[uinfo->value.enumerated.item]);
return 0;
}
static int sgio2audio_source_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_sgio2audio *chip = snd_kcontrol_chip(kcontrol);
ucontrol->value.enumerated.item[0] = ad1843_get_recsrc(&chip->ad1843);
return 0;
}
static int sgio2audio_source_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_sgio2audio *chip = snd_kcontrol_chip(kcontrol);
int newsrc, oldsrc;
oldsrc = ad1843_get_recsrc(&chip->ad1843);
newsrc = ad1843_set_recsrc(&chip->ad1843,
ucontrol->value.enumerated.item[0]);
return newsrc != oldsrc;
}
/* dac1/pcm0 mixer control */
static struct snd_kcontrol_new sgio2audio_ctrl_pcm0 __devinitdata = {
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = "PCM Playback Volume",
.index = 0,
.access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
.private_value = AD1843_GAIN_PCM_0,
.info = sgio2audio_gain_info,
.get = sgio2audio_gain_get,
.put = sgio2audio_gain_put,
};
/* dac2/pcm1 mixer control */
static struct snd_kcontrol_new sgio2audio_ctrl_pcm1 __devinitdata = {
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = "PCM Playback Volume",
.index = 1,
.access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
.private_value = AD1843_GAIN_PCM_1,
.info = sgio2audio_gain_info,
.get = sgio2audio_gain_get,
.put = sgio2audio_gain_put,
};
/* record level mixer control */
static struct snd_kcontrol_new sgio2audio_ctrl_reclevel __devinitdata = {
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = "Capture Volume",
.access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
.private_value = AD1843_GAIN_RECLEV,
.info = sgio2audio_gain_info,
.get = sgio2audio_gain_get,
.put = sgio2audio_gain_put,
};
/* record level source control */
static struct snd_kcontrol_new sgio2audio_ctrl_recsource __devinitdata = {
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = "Capture Source",
.access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
.info = sgio2audio_source_info,
.get = sgio2audio_source_get,
.put = sgio2audio_source_put,
};
/* line mixer control */
static struct snd_kcontrol_new sgio2audio_ctrl_line __devinitdata = {
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = "Line Playback Volume",
.index = 0,
.access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
.private_value = AD1843_GAIN_LINE,
.info = sgio2audio_gain_info,
.get = sgio2audio_gain_get,
.put = sgio2audio_gain_put,
};
/* cd mixer control */
static struct snd_kcontrol_new sgio2audio_ctrl_cd __devinitdata = {
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = "Line Playback Volume",
.index = 1,
.access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
.private_value = AD1843_GAIN_LINE_2,
.info = sgio2audio_gain_info,
.get = sgio2audio_gain_get,
.put = sgio2audio_gain_put,
};
/* mic mixer control */
static struct snd_kcontrol_new sgio2audio_ctrl_mic __devinitdata = {
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = "Mic Playback Volume",
.access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
.private_value = AD1843_GAIN_MIC,
.info = sgio2audio_gain_info,
.get = sgio2audio_gain_get,
.put = sgio2audio_gain_put,
};
static int __devinit snd_sgio2audio_new_mixer(struct snd_sgio2audio *chip)
{
int err;
err = snd_ctl_add(chip->card,
snd_ctl_new1(&sgio2audio_ctrl_pcm0, chip));
if (err < 0)
return err;
err = snd_ctl_add(chip->card,
snd_ctl_new1(&sgio2audio_ctrl_pcm1, chip));
if (err < 0)
return err;
err = snd_ctl_add(chip->card,
snd_ctl_new1(&sgio2audio_ctrl_reclevel, chip));
if (err < 0)
return err;
err = snd_ctl_add(chip->card,
snd_ctl_new1(&sgio2audio_ctrl_recsource, chip));
if (err < 0)
return err;
err = snd_ctl_add(chip->card,
snd_ctl_new1(&sgio2audio_ctrl_line, chip));
if (err < 0)
return err;
err = snd_ctl_add(chip->card,
snd_ctl_new1(&sgio2audio_ctrl_cd, chip));
if (err < 0)
return err;
err = snd_ctl_add(chip->card,
snd_ctl_new1(&sgio2audio_ctrl_mic, chip));
if (err < 0)
return err;
return 0;
}
/* low-level audio interface DMA */
/* get data out of bounce buffer, count must be a multiple of 32 */
/* returns 1 if a period has elapsed */
static int snd_sgio2audio_dma_pull_frag(struct snd_sgio2audio *chip,
unsigned int ch, unsigned int count)
{
int ret;
unsigned long src_base, src_pos, dst_mask;
unsigned char *dst_base;
int dst_pos;
u64 *src;
s16 *dst;
u64 x;
unsigned long flags;
struct snd_pcm_runtime *runtime = chip->channel[ch].substream->runtime;
spin_lock_irqsave(&chip->channel[ch].lock, flags);
src_base = (unsigned long) chip->ring_base | (ch << CHANNEL_RING_SHIFT);
src_pos = readq(&mace->perif.audio.chan[ch].read_ptr);
dst_base = runtime->dma_area;
dst_pos = chip->channel[ch].pos;
dst_mask = frames_to_bytes(runtime, runtime->buffer_size) - 1;
/* check if a period has elapsed */
chip->channel[ch].size += (count >> 3); /* in frames */
ret = chip->channel[ch].size >= runtime->period_size;
chip->channel[ch].size %= runtime->period_size;
while (count) {
src = (u64 *)(src_base + src_pos);
dst = (s16 *)(dst_base + dst_pos);
x = *src;
dst[0] = (x >> CHANNEL_LEFT_SHIFT) & 0xffff;
dst[1] = (x >> CHANNEL_RIGHT_SHIFT) & 0xffff;
src_pos = (src_pos + sizeof(u64)) & CHANNEL_RING_MASK;
dst_pos = (dst_pos + 2 * sizeof(s16)) & dst_mask;
count -= sizeof(u64);
}
writeq(src_pos, &mace->perif.audio.chan[ch].read_ptr); /* in bytes */
chip->channel[ch].pos = dst_pos;
spin_unlock_irqrestore(&chip->channel[ch].lock, flags);
return ret;
}
/* put some DMA data in bounce buffer, count must be a multiple of 32 */
/* returns 1 if a period has elapsed */
static int snd_sgio2audio_dma_push_frag(struct snd_sgio2audio *chip,
unsigned int ch, unsigned int count)
{
int ret;
s64 l, r;
unsigned long dst_base, dst_pos, src_mask;
unsigned char *src_base;
int src_pos;
u64 *dst;
s16 *src;
unsigned long flags;
struct snd_pcm_runtime *runtime = chip->channel[ch].substream->runtime;
spin_lock_irqsave(&chip->channel[ch].lock, flags);
dst_base = (unsigned long)chip->ring_base | (ch << CHANNEL_RING_SHIFT);
dst_pos = readq(&mace->perif.audio.chan[ch].write_ptr);
src_base = runtime->dma_area;
src_pos = chip->channel[ch].pos;
src_mask = frames_to_bytes(runtime, runtime->buffer_size) - 1;
/* check if a period has elapsed */
chip->channel[ch].size += (count >> 3); /* in frames */
ret = chip->channel[ch].size >= runtime->period_size;
chip->channel[ch].size %= runtime->period_size;
while (count) {
src = (s16 *)(src_base + src_pos);
dst = (u64 *)(dst_base + dst_pos);
l = src[0]; /* sign extend */
r = src[1]; /* sign extend */
*dst = ((l & 0x00ffffff) << CHANNEL_LEFT_SHIFT) |
((r & 0x00ffffff) << CHANNEL_RIGHT_SHIFT);
dst_pos = (dst_pos + sizeof(u64)) & CHANNEL_RING_MASK;
src_pos = (src_pos + 2 * sizeof(s16)) & src_mask;
count -= sizeof(u64);
}
writeq(dst_pos, &mace->perif.audio.chan[ch].write_ptr); /* in bytes */
chip->channel[ch].pos = src_pos;
spin_unlock_irqrestore(&chip->channel[ch].lock, flags);
return ret;
}
static int snd_sgio2audio_dma_start(struct snd_pcm_substream *substream)
{
struct snd_sgio2audio *chip = snd_pcm_substream_chip(substream);
struct snd_sgio2audio_chan *chan = substream->runtime->private_data;
int ch = chan->idx;
/* reset DMA channel */
writeq(CHANNEL_CONTROL_RESET, &mace->perif.audio.chan[ch].control);
udelay(10);
writeq(0, &mace->perif.audio.chan[ch].control);
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
/* push a full buffer */
snd_sgio2audio_dma_push_frag(chip, ch, CHANNEL_RING_SIZE - 32);
}
/* set DMA to wake on 50% empty and enable interrupt */
writeq(CHANNEL_DMA_ENABLE | CHANNEL_INT_THRESHOLD_50,
&mace->perif.audio.chan[ch].control);
return 0;
}
static int snd_sgio2audio_dma_stop(struct snd_pcm_substream *substream)
{
struct snd_sgio2audio_chan *chan = substream->runtime->private_data;
writeq(0, &mace->perif.audio.chan[chan->idx].control);
return 0;
}
static irqreturn_t snd_sgio2audio_dma_in_isr(int irq, void *dev_id)
{
struct snd_sgio2audio_chan *chan = dev_id;
struct snd_pcm_substream *substream;
struct snd_sgio2audio *chip;
int count, ch;
substream = chan->substream;
chip = snd_pcm_substream_chip(substream);
ch = chan->idx;
/* empty the ring */
count = CHANNEL_RING_SIZE -
readq(&mace->perif.audio.chan[ch].depth) - 32;
if (snd_sgio2audio_dma_pull_frag(chip, ch, count))
snd_pcm_period_elapsed(substream);
return IRQ_HANDLED;
}
static irqreturn_t snd_sgio2audio_dma_out_isr(int irq, void *dev_id)
{
struct snd_sgio2audio_chan *chan = dev_id;
struct snd_pcm_substream *substream;
struct snd_sgio2audio *chip;
int count, ch;
substream = chan->substream;
chip = snd_pcm_substream_chip(substream);
ch = chan->idx;
/* fill the ring */
count = CHANNEL_RING_SIZE -
readq(&mace->perif.audio.chan[ch].depth) - 32;
if (snd_sgio2audio_dma_push_frag(chip, ch, count))
snd_pcm_period_elapsed(substream);
return IRQ_HANDLED;
}
static irqreturn_t snd_sgio2audio_error_isr(int irq, void *dev_id)
{
struct snd_sgio2audio_chan *chan = dev_id;
struct snd_pcm_substream *substream;
substream = chan->substream;
snd_sgio2audio_dma_stop(substream);
snd_sgio2audio_dma_start(substream);
return IRQ_HANDLED;
}
/* PCM part */
/* PCM hardware definition */
static struct snd_pcm_hardware snd_sgio2audio_pcm_hw = {
.info = (SNDRV_PCM_INFO_MMAP |
SNDRV_PCM_INFO_MMAP_VALID |
SNDRV_PCM_INFO_INTERLEAVED |
SNDRV_PCM_INFO_BLOCK_TRANSFER),
.formats = SNDRV_PCM_FMTBIT_S16_BE,
.rates = SNDRV_PCM_RATE_8000_48000,
.rate_min = 8000,
.rate_max = 48000,
.channels_min = 2,
.channels_max = 2,
.buffer_bytes_max = 65536,
.period_bytes_min = 32768,
.period_bytes_max = 65536,
.periods_min = 1,
.periods_max = 1024,
};
/* PCM playback open callback */
static int snd_sgio2audio_playback1_open(struct snd_pcm_substream *substream)
{
struct snd_sgio2audio *chip = snd_pcm_substream_chip(substream);
struct snd_pcm_runtime *runtime = substream->runtime;
runtime->hw = snd_sgio2audio_pcm_hw;
runtime->private_data = &chip->channel[1];
return 0;
}
static int snd_sgio2audio_playback2_open(struct snd_pcm_substream *substream)
{
struct snd_sgio2audio *chip = snd_pcm_substream_chip(substream);
struct snd_pcm_runtime *runtime = substream->runtime;
runtime->hw = snd_sgio2audio_pcm_hw;
runtime->private_data = &chip->channel[2];
return 0;
}
/* PCM capture open callback */
static int snd_sgio2audio_capture_open(struct snd_pcm_substream *substream)
{
struct snd_sgio2audio *chip = snd_pcm_substream_chip(substream);
struct snd_pcm_runtime *runtime = substream->runtime;
runtime->hw = snd_sgio2audio_pcm_hw;
runtime->private_data = &chip->channel[0];
return 0;
}
/* PCM close callback */
static int snd_sgio2audio_pcm_close(struct snd_pcm_substream *substream)
{
struct snd_pcm_runtime *runtime = substream->runtime;
runtime->private_data = NULL;
return 0;
}
/* hw_params callback */
static int snd_sgio2audio_pcm_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *hw_params)
{
struct snd_pcm_runtime *runtime = substream->runtime;
int size = params_buffer_bytes(hw_params);
/* alloc virtual 'dma' area */
if (runtime->dma_area)
vfree(runtime->dma_area);
runtime->dma_area = vmalloc(size);
if (runtime->dma_area == NULL)
return -ENOMEM;
runtime->dma_bytes = size;
return 0;
}
/* hw_free callback */
static int snd_sgio2audio_pcm_hw_free(struct snd_pcm_substream *substream)
{
if (substream->runtime->dma_area)
vfree(substream->runtime->dma_area);
substream->runtime->dma_area = NULL;
return 0;
}
/* prepare callback */
static int snd_sgio2audio_pcm_prepare(struct snd_pcm_substream *substream)
{
struct snd_sgio2audio *chip = snd_pcm_substream_chip(substream);
struct snd_pcm_runtime *runtime = substream->runtime;
struct snd_sgio2audio_chan *chan = substream->runtime->private_data;
int ch = chan->idx;
unsigned long flags;
spin_lock_irqsave(&chip->channel[ch].lock, flags);
/* Setup the pseudo-dma transfer pointers. */
chip->channel[ch].pos = 0;
chip->channel[ch].size = 0;
chip->channel[ch].substream = substream;
/* set AD1843 format */
/* hardware format is always S16_LE */
switch (substream->stream) {
case SNDRV_PCM_STREAM_PLAYBACK:
ad1843_setup_dac(&chip->ad1843,
ch - 1,
runtime->rate,
SNDRV_PCM_FORMAT_S16_LE,
runtime->channels);
break;
case SNDRV_PCM_STREAM_CAPTURE:
ad1843_setup_adc(&chip->ad1843,
runtime->rate,
SNDRV_PCM_FORMAT_S16_LE,
runtime->channels);
break;
}
spin_unlock_irqrestore(&chip->channel[ch].lock, flags);
return 0;
}
/* trigger callback */
static int snd_sgio2audio_pcm_trigger(struct snd_pcm_substream *substream,
int cmd)
{
switch (cmd) {
case SNDRV_PCM_TRIGGER_START:
/* start the PCM engine */
snd_sgio2audio_dma_start(substream);
break;
case SNDRV_PCM_TRIGGER_STOP:
/* stop the PCM engine */
snd_sgio2audio_dma_stop(substream);
break;
default:
return -EINVAL;
}
return 0;
}
/* pointer callback */
static snd_pcm_uframes_t
snd_sgio2audio_pcm_pointer(struct snd_pcm_substream *substream)
{
struct snd_sgio2audio *chip = snd_pcm_substream_chip(substream);
struct snd_sgio2audio_chan *chan = substream->runtime->private_data;
/* get the current hardware pointer */
return bytes_to_frames(substream->runtime,
chip->channel[chan->idx].pos);
}
/* get the physical page pointer on the given offset */
static struct page *snd_sgio2audio_page(struct snd_pcm_substream *substream,
unsigned long offset)
{
return vmalloc_to_page(substream->runtime->dma_area + offset);
}
/* operators */
static struct snd_pcm_ops snd_sgio2audio_playback1_ops = {
.open = snd_sgio2audio_playback1_open,
.close = snd_sgio2audio_pcm_close,
.ioctl = snd_pcm_lib_ioctl,
.hw_params = snd_sgio2audio_pcm_hw_params,
.hw_free = snd_sgio2audio_pcm_hw_free,
.prepare = snd_sgio2audio_pcm_prepare,
.trigger = snd_sgio2audio_pcm_trigger,
.pointer = snd_sgio2audio_pcm_pointer,
.page = snd_sgio2audio_page,
};
static struct snd_pcm_ops snd_sgio2audio_playback2_ops = {
.open = snd_sgio2audio_playback2_open,
.close = snd_sgio2audio_pcm_close,
.ioctl = snd_pcm_lib_ioctl,
.hw_params = snd_sgio2audio_pcm_hw_params,
.hw_free = snd_sgio2audio_pcm_hw_free,
.prepare = snd_sgio2audio_pcm_prepare,
.trigger = snd_sgio2audio_pcm_trigger,
.pointer = snd_sgio2audio_pcm_pointer,
.page = snd_sgio2audio_page,
};
static struct snd_pcm_ops snd_sgio2audio_capture_ops = {
.open = snd_sgio2audio_capture_open,
.close = snd_sgio2audio_pcm_close,
.ioctl = snd_pcm_lib_ioctl,
.hw_params = snd_sgio2audio_pcm_hw_params,
.hw_free = snd_sgio2audio_pcm_hw_free,
.prepare = snd_sgio2audio_pcm_prepare,
.trigger = snd_sgio2audio_pcm_trigger,
.pointer = snd_sgio2audio_pcm_pointer,
.page = snd_sgio2audio_page,
};
/*
* definitions of capture are omitted here...
*/
/* create a pcm device */
static int __devinit snd_sgio2audio_new_pcm(struct snd_sgio2audio *chip)
{
struct snd_pcm *pcm;
int err;
/* create first pcm device with one outputs and one input */
err = snd_pcm_new(chip->card, "SGI O2 Audio", 0, 1, 1, &pcm);
if (err < 0)
return err;
pcm->private_data = chip;
strcpy(pcm->name, "SGI O2 DAC1");
/* set operators */
snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK,
&snd_sgio2audio_playback1_ops);
snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE,
&snd_sgio2audio_capture_ops);
/* create second pcm device with one outputs and no input */
err = snd_pcm_new(chip->card, "SGI O2 Audio", 1, 1, 0, &pcm);
if (err < 0)
return err;
pcm->private_data = chip;
strcpy(pcm->name, "SGI O2 DAC2");
/* set operators */
snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK,
&snd_sgio2audio_playback2_ops);
return 0;
}
static struct {
int idx;
int irq;
irqreturn_t (*isr)(int, void *);
const char *desc;
} snd_sgio2_isr_table[] = {
{
.idx = 0,
.irq = MACEISA_AUDIO1_DMAT_IRQ,
.isr = snd_sgio2audio_dma_in_isr,
.desc = "Capture DMA Channel 0"
}, {
.idx = 0,
.irq = MACEISA_AUDIO1_OF_IRQ,
.isr = snd_sgio2audio_error_isr,
.desc = "Capture Overflow"
}, {
.idx = 1,
.irq = MACEISA_AUDIO2_DMAT_IRQ,
.isr = snd_sgio2audio_dma_out_isr,
.desc = "Playback DMA Channel 1"
}, {
.idx = 1,
.irq = MACEISA_AUDIO2_MERR_IRQ,
.isr = snd_sgio2audio_error_isr,
.desc = "Memory Error Channel 1"
}, {
.idx = 2,
.irq = MACEISA_AUDIO3_DMAT_IRQ,
.isr = snd_sgio2audio_dma_out_isr,
.desc = "Playback DMA Channel 2"
}, {
.idx = 2,
.irq = MACEISA_AUDIO3_MERR_IRQ,
.isr = snd_sgio2audio_error_isr,
.desc = "Memory Error Channel 2"
}
};
/* ALSA driver */
static int snd_sgio2audio_free(struct snd_sgio2audio *chip)
{
int i;
/* reset interface */
writeq(AUDIO_CONTROL_RESET, &mace->perif.audio.control);
udelay(1);
writeq(0, &mace->perif.audio.control);
/* release IRQ's */
for (i = 0; i < ARRAY_SIZE(snd_sgio2_isr_table); i++)
free_irq(snd_sgio2_isr_table[i].irq,
&chip->channel[snd_sgio2_isr_table[i].idx]);
dma_free_coherent(NULL, MACEISA_RINGBUFFERS_SIZE,
chip->ring_base, chip->ring_base_dma);
/* release card data */
kfree(chip);
return 0;
}
static int snd_sgio2audio_dev_free(struct snd_device *device)
{
struct snd_sgio2audio *chip = device->device_data;
return snd_sgio2audio_free(chip);
}
static struct snd_device_ops ops = {
.dev_free = snd_sgio2audio_dev_free,
};
static int __devinit snd_sgio2audio_create(struct snd_card *card,
struct snd_sgio2audio **rchip)
{
struct snd_sgio2audio *chip;
int i, err;
*rchip = NULL;
/* check if a codec is attached to the interface */
/* (Audio or Audio/Video board present) */
if (!(readq(&mace->perif.audio.control) & AUDIO_CONTROL_CODEC_PRESENT))
return -ENOENT;
chip = kzalloc(sizeof(struct snd_sgio2audio), GFP_KERNEL);
if (chip == NULL)
return -ENOMEM;
chip->card = card;
chip->ring_base = dma_alloc_coherent(NULL, MACEISA_RINGBUFFERS_SIZE,
&chip->ring_base_dma, GFP_USER);
if (chip->ring_base == NULL) {
printk(KERN_ERR
"sgio2audio: could not allocate ring buffers\n");
kfree(chip);
return -ENOMEM;
}
spin_lock_init(&chip->ad1843_lock);
/* initialize channels */
for (i = 0; i < 3; i++) {
spin_lock_init(&chip->channel[i].lock);
chip->channel[i].idx = i;
}
/* allocate IRQs */
for (i = 0; i < ARRAY_SIZE(snd_sgio2_isr_table); i++) {
if (request_irq(snd_sgio2_isr_table[i].irq,
snd_sgio2_isr_table[i].isr,
0,
snd_sgio2_isr_table[i].desc,
&chip->channel[snd_sgio2_isr_table[i].idx])) {
snd_sgio2audio_free(chip);
printk(KERN_ERR "sgio2audio: cannot allocate irq %d\n",
snd_sgio2_isr_table[i].irq);
return -EBUSY;
}
}
/* reset the interface */
writeq(AUDIO_CONTROL_RESET, &mace->perif.audio.control);
udelay(1);
writeq(0, &mace->perif.audio.control);
msleep_interruptible(1); /* give time to recover */
/* set ring base */
writeq(chip->ring_base_dma, &mace->perif.ctrl.ringbase);
/* attach the AD1843 codec */
chip->ad1843.read = read_ad1843_reg;
chip->ad1843.write = write_ad1843_reg;
chip->ad1843.chip = chip;
/* initialize the AD1843 codec */
err = ad1843_init(&chip->ad1843);
if (err < 0) {
snd_sgio2audio_free(chip);
return err;
}
err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops);
if (err < 0) {
snd_sgio2audio_free(chip);
return err;
}
*rchip = chip;
return 0;
}
static int __devinit snd_sgio2audio_probe(struct platform_device *pdev)
{
struct snd_card *card;
struct snd_sgio2audio *chip;
int err;
card = snd_card_new(index, id, THIS_MODULE, 0);
if (card == NULL)
return -ENOMEM;
err = snd_sgio2audio_create(card, &chip);
if (err < 0) {
snd_card_free(card);
return err;
}
snd_card_set_dev(card, &pdev->dev);
err = snd_sgio2audio_new_pcm(chip);
if (err < 0) {
snd_card_free(card);
return err;
}
err = snd_sgio2audio_new_mixer(chip);
if (err < 0) {
snd_card_free(card);
return err;
}
strcpy(card->driver, "SGI O2 Audio");
strcpy(card->shortname, "SGI O2 Audio");
sprintf(card->longname, "%s irq %i-%i",
card->shortname,
MACEISA_AUDIO1_DMAT_IRQ,
MACEISA_AUDIO3_MERR_IRQ);
err = snd_card_register(card);
if (err < 0) {
snd_card_free(card);
return err;
}
platform_set_drvdata(pdev, card);
return 0;
}
static int __exit snd_sgio2audio_remove(struct platform_device *pdev)
{
struct snd_card *card = platform_get_drvdata(pdev);
snd_card_free(card);
platform_set_drvdata(pdev, NULL);
return 0;
}
static struct platform_driver sgio2audio_driver = {
.probe = snd_sgio2audio_probe,
.remove = __devexit_p(snd_sgio2audio_remove),
.driver = {
.name = "sgio2audio",
.owner = THIS_MODULE,
}
};
static int __init alsa_card_sgio2audio_init(void)
{
return platform_driver_register(&sgio2audio_driver);
}
static void __exit alsa_card_sgio2audio_exit(void)
{
platform_driver_unregister(&sgio2audio_driver);
}
module_init(alsa_card_sgio2audio_init)
module_exit(alsa_card_sgio2audio_exit)