/*
 *   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/interrupt.h>
#include <linux/dma-mapping.h>
#include <linux/platform_device.h>
#include <linux/io.h>
#include <linux/slab.h>
#include <linux/module.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 * const texts[3] = {
                "Cam Mic", "Mic", "Line"
        };
        return snd_ctl_enum_info(uinfo, 1, 3, texts);
}

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 const struct snd_kcontrol_new sgio2audio_ctrl_pcm0 = {
        .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 const struct snd_kcontrol_new sgio2audio_ctrl_pcm1 = {
        .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 const struct snd_kcontrol_new sgio2audio_ctrl_reclevel = {
        .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 const struct snd_kcontrol_new sgio2audio_ctrl_recsource = {
        .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 const struct snd_kcontrol_new sgio2audio_ctrl_line = {
        .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 const struct snd_kcontrol_new sgio2audio_ctrl_cd = {
        .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 const struct snd_kcontrol_new sgio2audio_ctrl_mic = {
        .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 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 const 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)
{
        return snd_pcm_lib_alloc_vmalloc_buffer(substream,
                                                params_buffer_bytes(hw_params));
}

/* hw_free callback */
static int snd_sgio2audio_pcm_hw_free(struct snd_pcm_substream *substream)
{
        return snd_pcm_lib_free_vmalloc_buffer(substream);
}

/* 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);
}

/* operators */
static const 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_pcm_lib_get_vmalloc_page,
        .mmap =        snd_pcm_lib_mmap_vmalloc,
};

static const 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_pcm_lib_get_vmalloc_page,
        .mmap =        snd_pcm_lib_mmap_vmalloc,
};

static const 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_pcm_lib_get_vmalloc_page,
        .mmap =        snd_pcm_lib_mmap_vmalloc,
};

/*
 *  definitions of capture are omitted here...
 */

/* create a pcm device */
static int 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 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(*chip), 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 snd_sgio2audio_probe(struct platform_device *pdev)
{
        struct snd_card *card;
        struct snd_sgio2audio *chip;
        int err;

        err = snd_card_new(&pdev->dev, 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;
        }

        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 snd_sgio2audio_remove(struct platform_device *pdev)
{
        struct snd_card *card = platform_get_drvdata(pdev);

        snd_card_free(card);
        return 0;
}

static struct platform_driver sgio2audio_driver = {
        .probe  = snd_sgio2audio_probe,
        .remove = snd_sgio2audio_remove,
        .driver = {
                .name   = "sgio2audio",
        }
};

module_platform_driver(sgio2audio_driver);