git.oblomov.eu Git - linux-2.6/blob - sound/mips/sgio2audio.c

   1 /*
   2  *   Sound driver for Silicon Graphics O2 Workstations A/V board audio.
   3  *
   4  *   Copyright 2003 Vivien Chappelier <vivien.chappelier@linux-mips.org>
   5  *   Copyright 2008 Thomas Bogendoerfer <tsbogend@alpha.franken.de>
   6  *   Mxier part taken from mace_audio.c:
   7  *   Copyright 2007 Thorben Jändling <tj.trevelyan@gmail.com>
   8  *
   9  *   This program is free software; you can redistribute it and/or modify
  10  *   it under the terms of the GNU General Public License as published by
  11  *   the Free Software Foundation; either version 2 of the License, or
  12  *   (at your option) any later version.
  13  *
  14  *   This program is distributed in the hope that it will be useful,
  15  *   but WITHOUT ANY WARRANTY; without even the implied warranty of
  16  *   MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
  17  *   GNU General Public License for more details.
  18  *
  19  *   You should have received a copy of the GNU General Public License
  20  *   along with this program; if not, write to the Free Software
  21  *   Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307 USA
  22  *
  23  */
  24
  25 #include <linux/init.h>
  26 #include <linux/delay.h>
  27 #include <linux/spinlock.h>
  28 #include <linux/gfp.h>
  29 #include <linux/vmalloc.h>
  30 #include <linux/interrupt.h>
  31 #include <linux/dma-mapping.h>
  32 #include <linux/platform_device.h>
  33 #include <linux/io.h>
  34
  35 #include <asm/ip32/ip32_ints.h>
  36 #include <asm/ip32/mace.h>
  37
  38 #include <sound/core.h>
  39 #include <sound/control.h>
  40 #include <sound/pcm.h>
  41 #define SNDRV_GET_ID
  42 #include <sound/initval.h>
  43 #include <sound/ad1843.h>
  44
  45
  46 MODULE_AUTHOR("Vivien Chappelier <vivien.chappelier@linux-mips.org>");
  47 MODULE_DESCRIPTION("SGI O2 Audio");
  48 MODULE_LICENSE("GPL");
  49 MODULE_SUPPORTED_DEVICE("{{Silicon Graphics, O2 Audio}}");
  50
  51 static int index = SNDRV_DEFAULT_IDX1;  /* Index 0-MAX */
  52 static char *id = SNDRV_DEFAULT_STR1;   /* ID for this card */
  53
  54 module_param(index, int, 0444);
  55 MODULE_PARM_DESC(index, "Index value for SGI O2 soundcard.");
  56 module_param(id, charp, 0444);
  57 MODULE_PARM_DESC(id, "ID string for SGI O2 soundcard.");
  58
  59
  60 #define AUDIO_CONTROL_RESET              BIT(0) /* 1: reset audio interface */
  61 #define AUDIO_CONTROL_CODEC_PRESENT      BIT(1) /* 1: codec detected */
  62
  63 #define CODEC_CONTROL_WORD_SHIFT        0
  64 #define CODEC_CONTROL_READ              BIT(16)
  65 #define CODEC_CONTROL_ADDRESS_SHIFT     17
  66
  67 #define CHANNEL_CONTROL_RESET           BIT(10) /* 1: reset channel */
  68 #define CHANNEL_DMA_ENABLE              BIT(9)  /* 1: enable DMA transfer */
  69 #define CHANNEL_INT_THRESHOLD_DISABLED  (0 << 5) /* interrupt disabled */
  70 #define CHANNEL_INT_THRESHOLD_25        (1 << 5) /* int on buffer >25% full */
  71 #define CHANNEL_INT_THRESHOLD_50        (2 << 5) /* int on buffer >50% full */
  72 #define CHANNEL_INT_THRESHOLD_75        (3 << 5) /* int on buffer >75% full */
  73 #define CHANNEL_INT_THRESHOLD_EMPTY     (4 << 5) /* int on buffer empty */
  74 #define CHANNEL_INT_THRESHOLD_NOT_EMPTY (5 << 5) /* int on buffer !empty */
  75 #define CHANNEL_INT_THRESHOLD_FULL      (6 << 5) /* int on buffer empty */
  76 #define CHANNEL_INT_THRESHOLD_NOT_FULL  (7 << 5) /* int on buffer !empty */
  77
  78 #define CHANNEL_RING_SHIFT              12
  79 #define CHANNEL_RING_SIZE               (1 << CHANNEL_RING_SHIFT)
  80 #define CHANNEL_RING_MASK               (CHANNEL_RING_SIZE - 1)
  81
  82 #define CHANNEL_LEFT_SHIFT 40
  83 #define CHANNEL_RIGHT_SHIFT 8
  84
  85 struct snd_sgio2audio_chan {
  86         int idx;
  87         struct snd_pcm_substream *substream;
  88         int pos;
  89         snd_pcm_uframes_t size;
  90         spinlock_t lock;
  91 };
  92
  93 /* definition of the chip-specific record */
  94 struct snd_sgio2audio {
  95         struct snd_card *card;
  96
  97         /* codec */
  98         struct snd_ad1843 ad1843;
  99         spinlock_t ad1843_lock;
 100
 101         /* channels */
 102         struct snd_sgio2audio_chan channel[3];
 103
 104         /* resources */
 105         void *ring_base;
 106         dma_addr_t ring_base_dma;
 107 };
 108
 109 /* AD1843 access */
 110
 111 /*
 112  * read_ad1843_reg returns the current contents of a 16 bit AD1843 register.
 113  *
 114  * Returns unsigned register value on success, -errno on failure.
 115  */
 116 static int read_ad1843_reg(void *priv, int reg)
 117 {
 118         struct snd_sgio2audio *chip = priv;
 119         int val;
 120         unsigned long flags;
 121
 122         spin_lock_irqsave(&chip->ad1843_lock, flags);
 123
 124         writeq((reg << CODEC_CONTROL_ADDRESS_SHIFT) |
 125                CODEC_CONTROL_READ, &mace->perif.audio.codec_control);
 126         wmb();
 127         val = readq(&mace->perif.audio.codec_control); /* flush bus */
 128         udelay(200);
 129
 130         val = readq(&mace->perif.audio.codec_read);
 131
 132         spin_unlock_irqrestore(&chip->ad1843_lock, flags);
 133         return val;
 134 }
 135
 136 /*
 137  * write_ad1843_reg writes the specified value to a 16 bit AD1843 register.
 138  */
 139 static int write_ad1843_reg(void *priv, int reg, int word)
 140 {
 141         struct snd_sgio2audio *chip = priv;
 142         int val;
 143         unsigned long flags;
 144
 145         spin_lock_irqsave(&chip->ad1843_lock, flags);
 146
 147         writeq((reg << CODEC_CONTROL_ADDRESS_SHIFT) |
 148                (word << CODEC_CONTROL_WORD_SHIFT),
 149                &mace->perif.audio.codec_control);
 150         wmb();
 151         val = readq(&mace->perif.audio.codec_control); /* flush bus */
 152         udelay(200);
 153
 154         spin_unlock_irqrestore(&chip->ad1843_lock, flags);
 155         return 0;
 156 }
 157
 158 static int sgio2audio_gain_info(struct snd_kcontrol *kcontrol,
 159                                struct snd_ctl_elem_info *uinfo)
 160 {
 161         struct snd_sgio2audio *chip = snd_kcontrol_chip(kcontrol);
 162
 163         uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
 164         uinfo->count = 2;
 165         uinfo->value.integer.min = 0;
 166         uinfo->value.integer.max = ad1843_get_gain_max(&chip->ad1843,
 167                                              (int)kcontrol->private_value);
 168         return 0;
 169 }
 170
 171 static int sgio2audio_gain_get(struct snd_kcontrol *kcontrol,
 172                                struct snd_ctl_elem_value *ucontrol)
 173 {
 174         struct snd_sgio2audio *chip = snd_kcontrol_chip(kcontrol);
 175         int vol;
 176
 177         vol = ad1843_get_gain(&chip->ad1843, (int)kcontrol->private_value);
 178
 179         ucontrol->value.integer.value[0] = (vol >> 8) & 0xFF;
 180         ucontrol->value.integer.value[1] = vol & 0xFF;
 181
 182         return 0;
 183 }
 184
 185 static int sgio2audio_gain_put(struct snd_kcontrol *kcontrol,
 186                         struct snd_ctl_elem_value *ucontrol)
 187 {
 188         struct snd_sgio2audio *chip = snd_kcontrol_chip(kcontrol);
 189         int newvol, oldvol;
 190
 191         oldvol = ad1843_get_gain(&chip->ad1843, kcontrol->private_value);
 192         newvol = (ucontrol->value.integer.value[0] << 8) |
 193                 ucontrol->value.integer.value[1];
 194
 195         newvol = ad1843_set_gain(&chip->ad1843, kcontrol->private_value,
 196                 newvol);
 197
 198         return newvol != oldvol;
 199 }
 200
 201 static int sgio2audio_source_info(struct snd_kcontrol *kcontrol,
 202                                struct snd_ctl_elem_info *uinfo)
 203 {
 204         static const char *texts[3] = {
 205                 "Cam Mic", "Mic", "Line"
 206         };
 207         uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
 208         uinfo->count = 1;
 209         uinfo->value.enumerated.items = 3;
 210         if (uinfo->value.enumerated.item >= 3)
 211                 uinfo->value.enumerated.item = 1;
 212         strcpy(uinfo->value.enumerated.name,
 213                texts[uinfo->value.enumerated.item]);
 214         return 0;
 215 }
 216
 217 static int sgio2audio_source_get(struct snd_kcontrol *kcontrol,
 218                                struct snd_ctl_elem_value *ucontrol)
 219 {
 220         struct snd_sgio2audio *chip = snd_kcontrol_chip(kcontrol);
 221
 222         ucontrol->value.enumerated.item[0] = ad1843_get_recsrc(&chip->ad1843);
 223         return 0;
 224 }
 225
 226 static int sgio2audio_source_put(struct snd_kcontrol *kcontrol,
 227                         struct snd_ctl_elem_value *ucontrol)
 228 {
 229         struct snd_sgio2audio *chip = snd_kcontrol_chip(kcontrol);
 230         int newsrc, oldsrc;
 231
 232         oldsrc = ad1843_get_recsrc(&chip->ad1843);
 233         newsrc = ad1843_set_recsrc(&chip->ad1843,
 234                                    ucontrol->value.enumerated.item[0]);
 235
 236         return newsrc != oldsrc;
 237 }
 238
 239 /* dac1/pcm0 mixer control */
 240 static struct snd_kcontrol_new sgio2audio_ctrl_pcm0 __devinitdata = {
 241         .iface          = SNDRV_CTL_ELEM_IFACE_MIXER,
 242         .name           = "PCM Playback Volume",
 243         .index          = 0,
 244         .access         = SNDRV_CTL_ELEM_ACCESS_READWRITE,
 245         .private_value  = AD1843_GAIN_PCM_0,
 246         .info           = sgio2audio_gain_info,
 247         .get            = sgio2audio_gain_get,
 248         .put            = sgio2audio_gain_put,
 249 };
 250
 251 /* dac2/pcm1 mixer control */
 252 static struct snd_kcontrol_new sgio2audio_ctrl_pcm1 __devinitdata = {
 253         .iface          = SNDRV_CTL_ELEM_IFACE_MIXER,
 254         .name           = "PCM Playback Volume",
 255         .index          = 1,
 256         .access         = SNDRV_CTL_ELEM_ACCESS_READWRITE,
 257         .private_value  = AD1843_GAIN_PCM_1,
 258         .info           = sgio2audio_gain_info,
 259         .get            = sgio2audio_gain_get,
 260         .put            = sgio2audio_gain_put,
 261 };
 262
 263 /* record level mixer control */
 264 static struct snd_kcontrol_new sgio2audio_ctrl_reclevel __devinitdata = {
 265         .iface          = SNDRV_CTL_ELEM_IFACE_MIXER,
 266         .name           = "Capture Volume",
 267         .access         = SNDRV_CTL_ELEM_ACCESS_READWRITE,
 268         .private_value  = AD1843_GAIN_RECLEV,
 269         .info           = sgio2audio_gain_info,
 270         .get            = sgio2audio_gain_get,
 271         .put            = sgio2audio_gain_put,
 272 };
 273
 274 /* record level source control */
 275 static struct snd_kcontrol_new sgio2audio_ctrl_recsource __devinitdata = {
 276         .iface          = SNDRV_CTL_ELEM_IFACE_MIXER,
 277         .name           = "Capture Source",
 278         .access         = SNDRV_CTL_ELEM_ACCESS_READWRITE,
 279         .info           = sgio2audio_source_info,
 280         .get            = sgio2audio_source_get,
 281         .put            = sgio2audio_source_put,
 282 };
 283
 284 /* line mixer control */
 285 static struct snd_kcontrol_new sgio2audio_ctrl_line __devinitdata = {
 286         .iface          = SNDRV_CTL_ELEM_IFACE_MIXER,
 287         .name           = "Line Playback Volume",
 288         .index          = 0,
 289         .access         = SNDRV_CTL_ELEM_ACCESS_READWRITE,
 290         .private_value  = AD1843_GAIN_LINE,
 291         .info           = sgio2audio_gain_info,
 292         .get            = sgio2audio_gain_get,
 293         .put            = sgio2audio_gain_put,
 294 };
 295
 296 /* cd mixer control */
 297 static struct snd_kcontrol_new sgio2audio_ctrl_cd __devinitdata = {
 298         .iface          = SNDRV_CTL_ELEM_IFACE_MIXER,
 299         .name           = "Line Playback Volume",
 300         .index          = 1,
 301         .access         = SNDRV_CTL_ELEM_ACCESS_READWRITE,
 302         .private_value  = AD1843_GAIN_LINE_2,
 303         .info           = sgio2audio_gain_info,
 304         .get            = sgio2audio_gain_get,
 305         .put            = sgio2audio_gain_put,
 306 };
 307
 308 /* mic mixer control */
 309 static struct snd_kcontrol_new sgio2audio_ctrl_mic __devinitdata = {
 310         .iface          = SNDRV_CTL_ELEM_IFACE_MIXER,
 311         .name           = "Mic Playback Volume",
 312         .access         = SNDRV_CTL_ELEM_ACCESS_READWRITE,
 313         .private_value  = AD1843_GAIN_MIC,
 314         .info           = sgio2audio_gain_info,
 315         .get            = sgio2audio_gain_get,
 316         .put            = sgio2audio_gain_put,
 317 };
 318
 319
 320 static int __devinit snd_sgio2audio_new_mixer(struct snd_sgio2audio *chip)
 321 {
 322         int err;
 323
 324         err = snd_ctl_add(chip->card,
 325                           snd_ctl_new1(&sgio2audio_ctrl_pcm0, chip));
 326         if (err < 0)
 327                 return err;
 328
 329         err = snd_ctl_add(chip->card,
 330                           snd_ctl_new1(&sgio2audio_ctrl_pcm1, chip));
 331         if (err < 0)
 332                 return err;
 333
 334         err = snd_ctl_add(chip->card,
 335                           snd_ctl_new1(&sgio2audio_ctrl_reclevel, chip));
 336         if (err < 0)
 337                 return err;
 338
 339         err = snd_ctl_add(chip->card,
 340                           snd_ctl_new1(&sgio2audio_ctrl_recsource, chip));
 341         if (err < 0)
 342                 return err;
 343         err = snd_ctl_add(chip->card,
 344                           snd_ctl_new1(&sgio2audio_ctrl_line, chip));
 345         if (err < 0)
 346                 return err;
 347
 348         err = snd_ctl_add(chip->card,
 349                           snd_ctl_new1(&sgio2audio_ctrl_cd, chip));
 350         if (err < 0)
 351                 return err;
 352
 353         err = snd_ctl_add(chip->card,
 354                           snd_ctl_new1(&sgio2audio_ctrl_mic, chip));
 355         if (err < 0)
 356                 return err;
 357
 358         return 0;
 359 }
 360
 361 /* low-level audio interface DMA */
 362
 363 /* get data out of bounce buffer, count must be a multiple of 32 */
 364 /* returns 1 if a period has elapsed */
 365 static int snd_sgio2audio_dma_pull_frag(struct snd_sgio2audio *chip,
 366                                         unsigned int ch, unsigned int count)
 367 {
 368         int ret;
 369         unsigned long src_base, src_pos, dst_mask;
 370         unsigned char *dst_base;
 371         int dst_pos;
 372         u64 *src;
 373         s16 *dst;
 374         u64 x;
 375         unsigned long flags;
 376         struct snd_pcm_runtime *runtime = chip->channel[ch].substream->runtime;
 377
 378         spin_lock_irqsave(&chip->channel[ch].lock, flags);
 379
 380         src_base = (unsigned long) chip->ring_base | (ch << CHANNEL_RING_SHIFT);
 381         src_pos = readq(&mace->perif.audio.chan[ch].read_ptr);
 382         dst_base = runtime->dma_area;
 383         dst_pos = chip->channel[ch].pos;
 384         dst_mask = frames_to_bytes(runtime, runtime->buffer_size) - 1;
 385
 386         /* check if a period has elapsed */
 387         chip->channel[ch].size += (count >> 3); /* in frames */
 388         ret = chip->channel[ch].size >= runtime->period_size;
 389         chip->channel[ch].size %= runtime->period_size;
 390
 391         while (count) {
 392                 src = (u64 *)(src_base + src_pos);
 393                 dst = (s16 *)(dst_base + dst_pos);
 394
 395                 x = *src;
 396                 dst[0] = (x >> CHANNEL_LEFT_SHIFT) & 0xffff;
 397                 dst[1] = (x >> CHANNEL_RIGHT_SHIFT) & 0xffff;
 398
 399                 src_pos = (src_pos + sizeof(u64)) & CHANNEL_RING_MASK;
 400                 dst_pos = (dst_pos + 2 * sizeof(s16)) & dst_mask;
 401                 count -= sizeof(u64);
 402         }
 403
 404         writeq(src_pos, &mace->perif.audio.chan[ch].read_ptr); /* in bytes */
 405         chip->channel[ch].pos = dst_pos;
 406
 407         spin_unlock_irqrestore(&chip->channel[ch].lock, flags);
 408         return ret;
 409 }
 410
 411 /* put some DMA data in bounce buffer, count must be a multiple of 32 */
 412 /* returns 1 if a period has elapsed */
 413 static int snd_sgio2audio_dma_push_frag(struct snd_sgio2audio *chip,
 414                                         unsigned int ch, unsigned int count)
 415 {
 416         int ret;
 417         s64 l, r;
 418         unsigned long dst_base, dst_pos, src_mask;
 419         unsigned char *src_base;
 420         int src_pos;
 421         u64 *dst;
 422         s16 *src;
 423         unsigned long flags;
 424         struct snd_pcm_runtime *runtime = chip->channel[ch].substream->runtime;
 425
 426         spin_lock_irqsave(&chip->channel[ch].lock, flags);
 427
 428         dst_base = (unsigned long)chip->ring_base | (ch << CHANNEL_RING_SHIFT);
 429         dst_pos = readq(&mace->perif.audio.chan[ch].write_ptr);
 430         src_base = runtime->dma_area;
 431         src_pos = chip->channel[ch].pos;
 432         src_mask = frames_to_bytes(runtime, runtime->buffer_size) - 1;
 433
 434         /* check if a period has elapsed */
 435         chip->channel[ch].size += (count >> 3); /* in frames */
 436         ret = chip->channel[ch].size >= runtime->period_size;
 437         chip->channel[ch].size %= runtime->period_size;
 438
 439         while (count) {
 440                 src = (s16 *)(src_base + src_pos);
 441                 dst = (u64 *)(dst_base + dst_pos);
 442
 443                 l = src[0]; /* sign extend */
 444                 r = src[1]; /* sign extend */
 445
 446                 *dst = ((l & 0x00ffffff) << CHANNEL_LEFT_SHIFT) |
 447                         ((r & 0x00ffffff) << CHANNEL_RIGHT_SHIFT);
 448
 449                 dst_pos = (dst_pos + sizeof(u64)) & CHANNEL_RING_MASK;
 450                 src_pos = (src_pos + 2 * sizeof(s16)) & src_mask;
 451                 count -= sizeof(u64);
 452         }
 453
 454         writeq(dst_pos, &mace->perif.audio.chan[ch].write_ptr); /* in bytes */
 455         chip->channel[ch].pos = src_pos;
 456
 457         spin_unlock_irqrestore(&chip->channel[ch].lock, flags);
 458         return ret;
 459 }
 460
 461 static int snd_sgio2audio_dma_start(struct snd_pcm_substream *substream)
 462 {
 463         struct snd_sgio2audio *chip = snd_pcm_substream_chip(substream);
 464         struct snd_sgio2audio_chan *chan = substream->runtime->private_data;
 465         int ch = chan->idx;
 466
 467         /* reset DMA channel */
 468         writeq(CHANNEL_CONTROL_RESET, &mace->perif.audio.chan[ch].control);
 469         udelay(10);
 470         writeq(0, &mace->perif.audio.chan[ch].control);
 471
 472         if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
 473                 /* push a full buffer */
 474                 snd_sgio2audio_dma_push_frag(chip, ch, CHANNEL_RING_SIZE - 32);
 475         }
 476         /* set DMA to wake on 50% empty and enable interrupt */
 477         writeq(CHANNEL_DMA_ENABLE | CHANNEL_INT_THRESHOLD_50,
 478                &mace->perif.audio.chan[ch].control);
 479         return 0;
 480 }
 481
 482 static int snd_sgio2audio_dma_stop(struct snd_pcm_substream *substream)
 483 {
 484         struct snd_sgio2audio_chan *chan = substream->runtime->private_data;
 485
 486         writeq(0, &mace->perif.audio.chan[chan->idx].control);
 487         return 0;
 488 }
 489
 490 static irqreturn_t snd_sgio2audio_dma_in_isr(int irq, void *dev_id)
 491 {
 492         struct snd_sgio2audio_chan *chan = dev_id;
 493         struct snd_pcm_substream *substream;
 494         struct snd_sgio2audio *chip;
 495         int count, ch;
 496
 497         substream = chan->substream;
 498         chip = snd_pcm_substream_chip(substream);
 499         ch = chan->idx;
 500
 501         /* empty the ring */
 502         count = CHANNEL_RING_SIZE -
 503                 readq(&mace->perif.audio.chan[ch].depth) - 32;
 504         if (snd_sgio2audio_dma_pull_frag(chip, ch, count))
 505                 snd_pcm_period_elapsed(substream);
 506
 507         return IRQ_HANDLED;
 508 }
 509
 510 static irqreturn_t snd_sgio2audio_dma_out_isr(int irq, void *dev_id)
 511 {
 512         struct snd_sgio2audio_chan *chan = dev_id;
 513         struct snd_pcm_substream *substream;
 514         struct snd_sgio2audio *chip;
 515         int count, ch;
 516
 517         substream = chan->substream;
 518         chip = snd_pcm_substream_chip(substream);
 519         ch = chan->idx;
 520         /* fill the ring */
 521         count = CHANNEL_RING_SIZE -
 522                 readq(&mace->perif.audio.chan[ch].depth) - 32;
 523         if (snd_sgio2audio_dma_push_frag(chip, ch, count))
 524                 snd_pcm_period_elapsed(substream);
 525
 526         return IRQ_HANDLED;
 527 }
 528
 529 static irqreturn_t snd_sgio2audio_error_isr(int irq, void *dev_id)
 530 {
 531         struct snd_sgio2audio_chan *chan = dev_id;
 532         struct snd_pcm_substream *substream;
 533
 534         substream = chan->substream;
 535         snd_sgio2audio_dma_stop(substream);
 536         snd_sgio2audio_dma_start(substream);
 537         return IRQ_HANDLED;
 538 }
 539
 540 /* PCM part */
 541 /* PCM hardware definition */
 542 static struct snd_pcm_hardware snd_sgio2audio_pcm_hw = {
 543         .info = (SNDRV_PCM_INFO_MMAP |
 544                  SNDRV_PCM_INFO_MMAP_VALID |
 545                  SNDRV_PCM_INFO_INTERLEAVED |
 546                  SNDRV_PCM_INFO_BLOCK_TRANSFER),
 547         .formats =          SNDRV_PCM_FMTBIT_S16_BE,
 548         .rates =            SNDRV_PCM_RATE_8000_48000,
 549         .rate_min =         8000,
 550         .rate_max =         48000,
 551         .channels_min =     2,
 552         .channels_max =     2,
 553         .buffer_bytes_max = 65536,
 554         .period_bytes_min = 32768,
 555         .period_bytes_max = 65536,
 556         .periods_min =      1,
 557         .periods_max =      1024,
 558 };
 559
 560 /* PCM playback open callback */
 561 static int snd_sgio2audio_playback1_open(struct snd_pcm_substream *substream)
 562 {
 563         struct snd_sgio2audio *chip = snd_pcm_substream_chip(substream);
 564         struct snd_pcm_runtime *runtime = substream->runtime;
 565
 566         runtime->hw = snd_sgio2audio_pcm_hw;
 567         runtime->private_data = &chip->channel[1];
 568         return 0;
 569 }
 570
 571 static int snd_sgio2audio_playback2_open(struct snd_pcm_substream *substream)
 572 {
 573         struct snd_sgio2audio *chip = snd_pcm_substream_chip(substream);
 574         struct snd_pcm_runtime *runtime = substream->runtime;
 575
 576         runtime->hw = snd_sgio2audio_pcm_hw;
 577         runtime->private_data = &chip->channel[2];
 578         return 0;
 579 }
 580
 581 /* PCM capture open callback */
 582 static int snd_sgio2audio_capture_open(struct snd_pcm_substream *substream)
 583 {
 584         struct snd_sgio2audio *chip = snd_pcm_substream_chip(substream);
 585         struct snd_pcm_runtime *runtime = substream->runtime;
 586
 587         runtime->hw = snd_sgio2audio_pcm_hw;
 588         runtime->private_data = &chip->channel[0];
 589         return 0;
 590 }
 591
 592 /* PCM close callback */
 593 static int snd_sgio2audio_pcm_close(struct snd_pcm_substream *substream)
 594 {
 595         struct snd_pcm_runtime *runtime = substream->runtime;
 596
 597         runtime->private_data = NULL;
 598         return 0;
 599 }
 600
 601
 602 /* hw_params callback */
 603 static int snd_sgio2audio_pcm_hw_params(struct snd_pcm_substream *substream,
 604                                         struct snd_pcm_hw_params *hw_params)
 605 {
 606         struct snd_pcm_runtime *runtime = substream->runtime;
 607         int size = params_buffer_bytes(hw_params);
 608
 609         /* alloc virtual 'dma' area */
 610         if (runtime->dma_area)
 611                 vfree(runtime->dma_area);
 612         runtime->dma_area = vmalloc(size);
 613         if (runtime->dma_area == NULL)
 614                 return -ENOMEM;
 615         runtime->dma_bytes = size;
 616         return 0;
 617 }
 618
 619 /* hw_free callback */
 620 static int snd_sgio2audio_pcm_hw_free(struct snd_pcm_substream *substream)
 621 {
 622         vfree(substream->runtime->dma_area);
 623         substream->runtime->dma_area = NULL;
 624         return 0;
 625 }
 626
 627 /* prepare callback */
 628 static int snd_sgio2audio_pcm_prepare(struct snd_pcm_substream *substream)
 629 {
 630         struct snd_sgio2audio *chip = snd_pcm_substream_chip(substream);
 631         struct snd_pcm_runtime *runtime = substream->runtime;
 632         struct snd_sgio2audio_chan *chan = substream->runtime->private_data;
 633         int ch = chan->idx;
 634         unsigned long flags;
 635
 636         spin_lock_irqsave(&chip->channel[ch].lock, flags);
 637
 638         /* Setup the pseudo-dma transfer pointers.  */
 639         chip->channel[ch].pos = 0;
 640         chip->channel[ch].size = 0;
 641         chip->channel[ch].substream = substream;
 642
 643         /* set AD1843 format */
 644         /* hardware format is always S16_LE */
 645         switch (substream->stream) {
 646         case SNDRV_PCM_STREAM_PLAYBACK:
 647                 ad1843_setup_dac(&chip->ad1843,
 648                                  ch - 1,
 649                                  runtime->rate,
 650                                  SNDRV_PCM_FORMAT_S16_LE,
 651                                  runtime->channels);
 652                 break;
 653         case SNDRV_PCM_STREAM_CAPTURE:
 654                 ad1843_setup_adc(&chip->ad1843,
 655                                  runtime->rate,
 656                                  SNDRV_PCM_FORMAT_S16_LE,
 657                                  runtime->channels);
 658                 break;
 659         }
 660         spin_unlock_irqrestore(&chip->channel[ch].lock, flags);
 661         return 0;
 662 }
 663
 664 /* trigger callback */
 665 static int snd_sgio2audio_pcm_trigger(struct snd_pcm_substream *substream,
 666                                       int cmd)
 667 {
 668         switch (cmd) {
 669         case SNDRV_PCM_TRIGGER_START:
 670                 /* start the PCM engine */
 671                 snd_sgio2audio_dma_start(substream);
 672                 break;
 673         case SNDRV_PCM_TRIGGER_STOP:
 674                 /* stop the PCM engine */
 675                 snd_sgio2audio_dma_stop(substream);
 676                 break;
 677         default:
 678                 return -EINVAL;
 679         }
 680         return 0;
 681 }
 682
 683 /* pointer callback */
 684 static snd_pcm_uframes_t
 685 snd_sgio2audio_pcm_pointer(struct snd_pcm_substream *substream)
 686 {
 687         struct snd_sgio2audio *chip = snd_pcm_substream_chip(substream);
 688         struct snd_sgio2audio_chan *chan = substream->runtime->private_data;
 689
 690         /* get the current hardware pointer */
 691         return bytes_to_frames(substream->runtime,
 692                                chip->channel[chan->idx].pos);
 693 }
 694
 695 /* get the physical page pointer on the given offset */
 696 static struct page *snd_sgio2audio_page(struct snd_pcm_substream *substream,
 697                                         unsigned long offset)
 698 {
 699         return vmalloc_to_page(substream->runtime->dma_area + offset);
 700 }
 701
 702 /* operators */
 703 static struct snd_pcm_ops snd_sgio2audio_playback1_ops = {
 704         .open =        snd_sgio2audio_playback1_open,
 705         .close =       snd_sgio2audio_pcm_close,
 706         .ioctl =       snd_pcm_lib_ioctl,
 707         .hw_params =   snd_sgio2audio_pcm_hw_params,
 708         .hw_free =     snd_sgio2audio_pcm_hw_free,
 709         .prepare =     snd_sgio2audio_pcm_prepare,
 710         .trigger =     snd_sgio2audio_pcm_trigger,
 711         .pointer =     snd_sgio2audio_pcm_pointer,
 712         .page =        snd_sgio2audio_page,
 713 };
 714
 715 static struct snd_pcm_ops snd_sgio2audio_playback2_ops = {
 716         .open =        snd_sgio2audio_playback2_open,
 717         .close =       snd_sgio2audio_pcm_close,
 718         .ioctl =       snd_pcm_lib_ioctl,
 719         .hw_params =   snd_sgio2audio_pcm_hw_params,
 720         .hw_free =     snd_sgio2audio_pcm_hw_free,
 721         .prepare =     snd_sgio2audio_pcm_prepare,
 722         .trigger =     snd_sgio2audio_pcm_trigger,
 723         .pointer =     snd_sgio2audio_pcm_pointer,
 724         .page =        snd_sgio2audio_page,
 725 };
 726
 727 static struct snd_pcm_ops snd_sgio2audio_capture_ops = {
 728         .open =        snd_sgio2audio_capture_open,
 729         .close =       snd_sgio2audio_pcm_close,
 730         .ioctl =       snd_pcm_lib_ioctl,
 731         .hw_params =   snd_sgio2audio_pcm_hw_params,
 732         .hw_free =     snd_sgio2audio_pcm_hw_free,
 733         .prepare =     snd_sgio2audio_pcm_prepare,
 734         .trigger =     snd_sgio2audio_pcm_trigger,
 735         .pointer =     snd_sgio2audio_pcm_pointer,
 736         .page =        snd_sgio2audio_page,
 737 };
 738
 739 /*
 740  *  definitions of capture are omitted here...
 741  */
 742
 743 /* create a pcm device */
 744 static int __devinit snd_sgio2audio_new_pcm(struct snd_sgio2audio *chip)
 745 {
 746         struct snd_pcm *pcm;
 747         int err;
 748
 749         /* create first pcm device with one outputs and one input */
 750         err = snd_pcm_new(chip->card, "SGI O2 Audio", 0, 1, 1, &pcm);
 751         if (err < 0)
 752                 return err;
 753
 754         pcm->private_data = chip;
 755         strcpy(pcm->name, "SGI O2 DAC1");
 756
 757         /* set operators */
 758         snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK,
 759                         &snd_sgio2audio_playback1_ops);
 760         snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE,
 761                         &snd_sgio2audio_capture_ops);
 762
 763         /* create second  pcm device with one outputs and no input */
 764         err = snd_pcm_new(chip->card, "SGI O2 Audio", 1, 1, 0, &pcm);
 765         if (err < 0)
 766                 return err;
 767
 768         pcm->private_data = chip;
 769         strcpy(pcm->name, "SGI O2 DAC2");
 770
 771         /* set operators */
 772         snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK,
 773                         &snd_sgio2audio_playback2_ops);
 774
 775         return 0;
 776 }
 777
 778 static struct {
 779         int idx;
 780         int irq;
 781         irqreturn_t (*isr)(int, void *);
 782         const char *desc;
 783 } snd_sgio2_isr_table[] = {
 784         {
 785                 .idx = 0,
 786                 .irq = MACEISA_AUDIO1_DMAT_IRQ,
 787                 .isr = snd_sgio2audio_dma_in_isr,
 788                 .desc = "Capture DMA Channel 0"
 789         }, {
 790                 .idx = 0,
 791                 .irq = MACEISA_AUDIO1_OF_IRQ,
 792                 .isr = snd_sgio2audio_error_isr,
 793                 .desc = "Capture Overflow"
 794         }, {
 795                 .idx = 1,
 796                 .irq = MACEISA_AUDIO2_DMAT_IRQ,
 797                 .isr = snd_sgio2audio_dma_out_isr,
 798                 .desc = "Playback DMA Channel 1"
 799         }, {
 800                 .idx = 1,
 801                 .irq = MACEISA_AUDIO2_MERR_IRQ,
 802                 .isr = snd_sgio2audio_error_isr,
 803                 .desc = "Memory Error Channel 1"
 804         }, {
 805                 .idx = 2,
 806                 .irq = MACEISA_AUDIO3_DMAT_IRQ,
 807                 .isr = snd_sgio2audio_dma_out_isr,
 808                 .desc = "Playback DMA Channel 2"
 809         }, {
 810                 .idx = 2,
 811                 .irq = MACEISA_AUDIO3_MERR_IRQ,
 812                 .isr = snd_sgio2audio_error_isr,
 813                 .desc = "Memory Error Channel 2"
 814         }
 815 };
 816
 817 /* ALSA driver */
 818
 819 static int snd_sgio2audio_free(struct snd_sgio2audio *chip)
 820 {
 821         int i;
 822
 823         /* reset interface */
 824         writeq(AUDIO_CONTROL_RESET, &mace->perif.audio.control);
 825         udelay(1);
 826         writeq(0, &mace->perif.audio.control);
 827
 828         /* release IRQ's */
 829         for (i = 0; i < ARRAY_SIZE(snd_sgio2_isr_table); i++)
 830                 free_irq(snd_sgio2_isr_table[i].irq,
 831                          &chip->channel[snd_sgio2_isr_table[i].idx]);
 832
 833         dma_free_coherent(NULL, MACEISA_RINGBUFFERS_SIZE,
 834                           chip->ring_base, chip->ring_base_dma);
 835
 836         /* release card data */
 837         kfree(chip);
 838         return 0;
 839 }
 840
 841 static int snd_sgio2audio_dev_free(struct snd_device *device)
 842 {
 843         struct snd_sgio2audio *chip = device->device_data;
 844
 845         return snd_sgio2audio_free(chip);
 846 }
 847
 848 static struct snd_device_ops ops = {
 849         .dev_free = snd_sgio2audio_dev_free,
 850 };
 851
 852 static int __devinit snd_sgio2audio_create(struct snd_card *card,
 853                                            struct snd_sgio2audio **rchip)
 854 {
 855         struct snd_sgio2audio *chip;
 856         int i, err;
 857
 858         *rchip = NULL;
 859
 860         /* check if a codec is attached to the interface */
 861         /* (Audio or Audio/Video board present) */
 862         if (!(readq(&mace->perif.audio.control) & AUDIO_CONTROL_CODEC_PRESENT))
 863                 return -ENOENT;
 864
 865         chip = kzalloc(sizeof(struct snd_sgio2audio), GFP_KERNEL);
 866         if (chip == NULL)
 867                 return -ENOMEM;
 868
 869         chip->card = card;
 870
 871         chip->ring_base = dma_alloc_coherent(NULL, MACEISA_RINGBUFFERS_SIZE,
 872                                              &chip->ring_base_dma, GFP_USER);
 873         if (chip->ring_base == NULL) {
 874                 printk(KERN_ERR
 875                        "sgio2audio: could not allocate ring buffers\n");
 876                 kfree(chip);
 877                 return -ENOMEM;
 878         }
 879
 880         spin_lock_init(&chip->ad1843_lock);
 881
 882         /* initialize channels */
 883         for (i = 0; i < 3; i++) {
 884                 spin_lock_init(&chip->channel[i].lock);
 885                 chip->channel[i].idx = i;
 886         }
 887
 888         /* allocate IRQs */
 889         for (i = 0; i < ARRAY_SIZE(snd_sgio2_isr_table); i++) {
 890                 if (request_irq(snd_sgio2_isr_table[i].irq,
 891                                 snd_sgio2_isr_table[i].isr,
 892                                 0,
 893                                 snd_sgio2_isr_table[i].desc,
 894                                 &chip->channel[snd_sgio2_isr_table[i].idx])) {
 895                         snd_sgio2audio_free(chip);
 896                         printk(KERN_ERR "sgio2audio: cannot allocate irq %d\n",
 897                                snd_sgio2_isr_table[i].irq);
 898                         return -EBUSY;
 899                 }
 900         }
 901
 902         /* reset the interface */
 903         writeq(AUDIO_CONTROL_RESET, &mace->perif.audio.control);
 904         udelay(1);
 905         writeq(0, &mace->perif.audio.control);
 906         msleep_interruptible(1); /* give time to recover */
 907
 908         /* set ring base */
 909         writeq(chip->ring_base_dma, &mace->perif.ctrl.ringbase);
 910
 911         /* attach the AD1843 codec */
 912         chip->ad1843.read = read_ad1843_reg;
 913         chip->ad1843.write = write_ad1843_reg;
 914         chip->ad1843.chip = chip;
 915
 916         /* initialize the AD1843 codec */
 917         err = ad1843_init(&chip->ad1843);
 918         if (err < 0) {
 919                 snd_sgio2audio_free(chip);
 920                 return err;
 921         }
 922
 923         err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops);
 924         if (err < 0) {
 925                 snd_sgio2audio_free(chip);
 926                 return err;
 927         }
 928         *rchip = chip;
 929         return 0;
 930 }
 931
 932 static int __devinit snd_sgio2audio_probe(struct platform_device *pdev)
 933 {
 934         struct snd_card *card;
 935         struct snd_sgio2audio *chip;
 936         int err;
 937
 938         err = snd_card_create(index, id, THIS_MODULE, 0, &card);
 939         if (err < 0)
 940                 return err;
 941
 942         err = snd_sgio2audio_create(card, &chip);
 943         if (err < 0) {
 944                 snd_card_free(card);
 945                 return err;
 946         }
 947         snd_card_set_dev(card, &pdev->dev);
 948
 949         err = snd_sgio2audio_new_pcm(chip);
 950         if (err < 0) {
 951                 snd_card_free(card);
 952                 return err;
 953         }
 954         err = snd_sgio2audio_new_mixer(chip);
 955         if (err < 0) {
 956                 snd_card_free(card);
 957                 return err;
 958         }
 959
 960         strcpy(card->driver, "SGI O2 Audio");
 961         strcpy(card->shortname, "SGI O2 Audio");
 962         sprintf(card->longname, "%s irq %i-%i",
 963                 card->shortname,
 964                 MACEISA_AUDIO1_DMAT_IRQ,
 965                 MACEISA_AUDIO3_MERR_IRQ);
 966
 967         err = snd_card_register(card);
 968         if (err < 0) {
 969                 snd_card_free(card);
 970                 return err;
 971         }
 972         platform_set_drvdata(pdev, card);
 973         return 0;
 974 }
 975
 976 static int __exit snd_sgio2audio_remove(struct platform_device *pdev)
 977 {
 978         struct snd_card *card = platform_get_drvdata(pdev);
 979
 980         snd_card_free(card);
 981         platform_set_drvdata(pdev, NULL);
 982         return 0;
 983 }
 984
 985 static struct platform_driver sgio2audio_driver = {
 986         .probe  = snd_sgio2audio_probe,
 987         .remove = __devexit_p(snd_sgio2audio_remove),
 988         .driver = {
 989                 .name   = "sgio2audio",
 990                 .owner  = THIS_MODULE,
 991         }
 992 };
 993
 994 static int __init alsa_card_sgio2audio_init(void)
 995 {
 996         return platform_driver_register(&sgio2audio_driver);
 997 }
 998
 999 static void __exit alsa_card_sgio2audio_exit(void)
1000 {
1001         platform_driver_unregister(&sgio2audio_driver);
1002 }
1003
1004 module_init(alsa_card_sgio2audio_init)
1005 module_exit(alsa_card_sgio2audio_exit)