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sgio2audio.c

/*
 *   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;

      err = snd_card_create(index, id, THIS_MODULE, 0, &card);
      if (err < 0)
            return err;

      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)

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