ALSA: ALSA driver for SGI O2 audio board

This patch adds a new ALSA driver for the audio device found inside
most of the SGI O2 workstation. The hardware uses a SGI custom chip,
which feeds a AD codec chip.

Signed-off-by: Thomas Bogendoerfer <tsbogend@alpha.franken.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>

1 files changed, 1006 insertions, 0 deletions

diff --git a/sound/mips/sgio2audio.c b/sound/mips/sgio2audio.c
new file mode 100644
index 000000000000..4c63504348dc
--- /dev/null
+++ b/sound/mips/sgio2audio.c
@@ -0,0 +1,1006 @@
+/*
+ *   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)