sound/mips/sgio2audio.c

   1 // SPDX-License-Identifier: GPL-2.0-or-later
   2 /*
   3  *   Sound driver for Silicon Graphics O2 Workstations A/V board audio.
   4  *
   5  *   Copyright 2003 Vivien Chappelier <vivien.chappelier@linux-mips.org>
   6  *   Copyright 2008 Thomas Bogendoerfer <tsbogend@alpha.franken.de>
   7  *   Mxier part taken from mace_audio.c:
   8  *   Copyright 2007 Thorben Jändling <tj.trevelyan@gmail.com>
   9  */
  10
  11 #include <linux/init.h>
  12 #include <linux/delay.h>
  13 #include <linux/spinlock.h>
  14 #include <linux/interrupt.h>
  15 #include <linux/dma-mapping.h>
  16 #include <linux/platform_device.h>
  17 #include <linux/io.h>
  18 #include <linux/slab.h>
  19 #include <linux/module.h>
  20
  21 #include <asm/ip32/ip32_ints.h>
  22 #include <asm/ip32/mace.h>
  23
  24 #include <sound/core.h>
  25 #include <sound/control.h>
  26 #include <sound/pcm.h>
  27 #define SNDRV_GET_ID
  28 #include <sound/initval.h>
  29 #include <sound/ad1843.h>
  30
  31
  32 MODULE_AUTHOR("Vivien Chappelier <vivien.chappelier@linux-mips.org>");
  33 MODULE_DESCRIPTION("SGI O2 Audio");
  34 MODULE_LICENSE("GPL");
  35
  36 static int index = SNDRV_DEFAULT_IDX1;  /* Index 0-MAX */
  37 static char *id = SNDRV_DEFAULT_STR1;   /* ID for this card */
  38
  39 module_param(index, int, 0444);
  40 MODULE_PARM_DESC(index, "Index value for SGI O2 soundcard.");
  41 module_param(id, charp, 0444);
  42 MODULE_PARM_DESC(id, "ID string for SGI O2 soundcard.");
  43
  44
  45 #define AUDIO_CONTROL_RESET              BIT(0) /* 1: reset audio interface */
  46 #define AUDIO_CONTROL_CODEC_PRESENT      BIT(1) /* 1: codec detected */
  47
  48 #define CODEC_CONTROL_WORD_SHIFT        0
  49 #define CODEC_CONTROL_READ              BIT(16)
  50 #define CODEC_CONTROL_ADDRESS_SHIFT     17
  51
  52 #define CHANNEL_CONTROL_RESET           BIT(10) /* 1: reset channel */
  53 #define CHANNEL_DMA_ENABLE              BIT(9)  /* 1: enable DMA transfer */
  54 #define CHANNEL_INT_THRESHOLD_DISABLED  (0 << 5) /* interrupt disabled */
  55 #define CHANNEL_INT_THRESHOLD_25        (1 << 5) /* int on buffer >25% full */
  56 #define CHANNEL_INT_THRESHOLD_50        (2 << 5) /* int on buffer >50% full */
  57 #define CHANNEL_INT_THRESHOLD_75        (3 << 5) /* int on buffer >75% full */
  58 #define CHANNEL_INT_THRESHOLD_EMPTY     (4 << 5) /* int on buffer empty */
  59 #define CHANNEL_INT_THRESHOLD_NOT_EMPTY (5 << 5) /* int on buffer !empty */
  60 #define CHANNEL_INT_THRESHOLD_FULL      (6 << 5) /* int on buffer empty */
  61 #define CHANNEL_INT_THRESHOLD_NOT_FULL  (7 << 5) /* int on buffer !empty */
  62
  63 #define CHANNEL_RING_SHIFT              12
  64 #define CHANNEL_RING_SIZE               (1 << CHANNEL_RING_SHIFT)
  65 #define CHANNEL_RING_MASK               (CHANNEL_RING_SIZE - 1)
  66
  67 #define CHANNEL_LEFT_SHIFT 40
  68 #define CHANNEL_RIGHT_SHIFT 8
  69
  70 struct snd_sgio2audio_chan {
  71         int idx;
  72         struct snd_pcm_substream *substream;
  73         int pos;
  74         snd_pcm_uframes_t size;
  75         spinlock_t lock;
  76 };
  77
  78 /* definition of the chip-specific record */
  79 struct snd_sgio2audio {
  80         struct snd_card *card;
  81
  82         /* codec */
  83         struct snd_ad1843 ad1843;
  84         spinlock_t ad1843_lock;
  85
  86         /* channels */
  87         struct snd_sgio2audio_chan channel[3];
  88
  89         /* resources */
  90         void *ring_base;
  91         dma_addr_t ring_base_dma;
  92 };
  93
  94 /* AD1843 access */
  95
  96 /*
  97  * read_ad1843_reg returns the current contents of a 16 bit AD1843 register.
  98  *
  99  * Returns unsigned register value on success, -errno on failure.
 100  */
 101 static int read_ad1843_reg(void *priv, int reg)
 102 {
 103         struct snd_sgio2audio *chip = priv;
 104         int val;
 105         unsigned long flags;
 106
 107         spin_lock_irqsave(&chip->ad1843_lock, flags);
 108
 109         writeq((reg << CODEC_CONTROL_ADDRESS_SHIFT) |
 110                CODEC_CONTROL_READ, &mace->perif.audio.codec_control);
 111         wmb();
 112         val = readq(&mace->perif.audio.codec_control); /* flush bus */
 113         udelay(200);
 114
 115         val = readq(&mace->perif.audio.codec_read);
 116
 117         spin_unlock_irqrestore(&chip->ad1843_lock, flags);
 118         return val;
 119 }
 120
 121 /*
 122  * write_ad1843_reg writes the specified value to a 16 bit AD1843 register.
 123  */
 124 static int write_ad1843_reg(void *priv, int reg, int word)
 125 {
 126         struct snd_sgio2audio *chip = priv;
 127         int val;
 128         unsigned long flags;
 129
 130         spin_lock_irqsave(&chip->ad1843_lock, flags);
 131
 132         writeq((reg << CODEC_CONTROL_ADDRESS_SHIFT) |
 133                (word << CODEC_CONTROL_WORD_SHIFT),
 134                &mace->perif.audio.codec_control);
 135         wmb();
 136         val = readq(&mace->perif.audio.codec_control); /* flush bus */
 137         udelay(200);
 138
 139         spin_unlock_irqrestore(&chip->ad1843_lock, flags);
 140         return 0;
 141 }
 142
 143 static int sgio2audio_gain_info(struct snd_kcontrol *kcontrol,
 144                                struct snd_ctl_elem_info *uinfo)
 145 {
 146         struct snd_sgio2audio *chip = snd_kcontrol_chip(kcontrol);
 147
 148         uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
 149         uinfo->count = 2;
 150         uinfo->value.integer.min = 0;
 151         uinfo->value.integer.max = ad1843_get_gain_max(&chip->ad1843,
 152                                              (int)kcontrol->private_value);
 153         return 0;
 154 }
 155
 156 static int sgio2audio_gain_get(struct snd_kcontrol *kcontrol,
 157                                struct snd_ctl_elem_value *ucontrol)
 158 {
 159         struct snd_sgio2audio *chip = snd_kcontrol_chip(kcontrol);
 160         int vol;
 161
 162         vol = ad1843_get_gain(&chip->ad1843, (int)kcontrol->private_value);
 163
 164         ucontrol->value.integer.value[0] = (vol >> 8) & 0xFF;
 165         ucontrol->value.integer.value[1] = vol & 0xFF;
 166
 167         return 0;
 168 }
 169
 170 static int sgio2audio_gain_put(struct snd_kcontrol *kcontrol,
 171                         struct snd_ctl_elem_value *ucontrol)
 172 {
 173         struct snd_sgio2audio *chip = snd_kcontrol_chip(kcontrol);
 174         int newvol, oldvol;
 175
 176         oldvol = ad1843_get_gain(&chip->ad1843, kcontrol->private_value);
 177         newvol = (ucontrol->value.integer.value[0] << 8) |
 178                 ucontrol->value.integer.value[1];
 179
 180         newvol = ad1843_set_gain(&chip->ad1843, kcontrol->private_value,
 181                 newvol);
 182
 183         return newvol != oldvol;
 184 }
 185
 186 static int sgio2audio_source_info(struct snd_kcontrol *kcontrol,
 187                                struct snd_ctl_elem_info *uinfo)
 188 {
 189         static const char * const texts[3] = {
 190                 "Cam Mic", "Mic", "Line"
 191         };
 192         return snd_ctl_enum_info(uinfo, 1, 3, texts);
 193 }
 194
 195 static int sgio2audio_source_get(struct snd_kcontrol *kcontrol,
 196                                struct snd_ctl_elem_value *ucontrol)
 197 {
 198         struct snd_sgio2audio *chip = snd_kcontrol_chip(kcontrol);
 199
 200         ucontrol->value.enumerated.item[0] = ad1843_get_recsrc(&chip->ad1843);
 201         return 0;
 202 }
 203
 204 static int sgio2audio_source_put(struct snd_kcontrol *kcontrol,
 205                         struct snd_ctl_elem_value *ucontrol)
 206 {
 207         struct snd_sgio2audio *chip = snd_kcontrol_chip(kcontrol);
 208         int newsrc, oldsrc;
 209
 210         oldsrc = ad1843_get_recsrc(&chip->ad1843);
 211         newsrc = ad1843_set_recsrc(&chip->ad1843,
 212                                    ucontrol->value.enumerated.item[0]);
 213
 214         return newsrc != oldsrc;
 215 }
 216
 217 /* dac1/pcm0 mixer control */
 218 static const struct snd_kcontrol_new sgio2audio_ctrl_pcm0 = {
 219         .iface          = SNDRV_CTL_ELEM_IFACE_MIXER,
 220         .name           = "PCM Playback Volume",
 221         .index          = 0,
 222         .access         = SNDRV_CTL_ELEM_ACCESS_READWRITE,
 223         .private_value  = AD1843_GAIN_PCM_0,
 224         .info           = sgio2audio_gain_info,
 225         .get            = sgio2audio_gain_get,
 226         .put            = sgio2audio_gain_put,
 227 };
 228
 229 /* dac2/pcm1 mixer control */
 230 static const struct snd_kcontrol_new sgio2audio_ctrl_pcm1 = {
 231         .iface          = SNDRV_CTL_ELEM_IFACE_MIXER,
 232         .name           = "PCM Playback Volume",
 233         .index          = 1,
 234         .access         = SNDRV_CTL_ELEM_ACCESS_READWRITE,
 235         .private_value  = AD1843_GAIN_PCM_1,
 236         .info           = sgio2audio_gain_info,
 237         .get            = sgio2audio_gain_get,
 238         .put            = sgio2audio_gain_put,
 239 };
 240
 241 /* record level mixer control */
 242 static const struct snd_kcontrol_new sgio2audio_ctrl_reclevel = {
 243         .iface          = SNDRV_CTL_ELEM_IFACE_MIXER,
 244         .name           = "Capture Volume",
 245         .access         = SNDRV_CTL_ELEM_ACCESS_READWRITE,
 246         .private_value  = AD1843_GAIN_RECLEV,
 247         .info           = sgio2audio_gain_info,
 248         .get            = sgio2audio_gain_get,
 249         .put            = sgio2audio_gain_put,
 250 };
 251
 252 /* record level source control */
 253 static const struct snd_kcontrol_new sgio2audio_ctrl_recsource = {
 254         .iface          = SNDRV_CTL_ELEM_IFACE_MIXER,
 255         .name           = "Capture Source",
 256         .access         = SNDRV_CTL_ELEM_ACCESS_READWRITE,
 257         .info           = sgio2audio_source_info,
 258         .get            = sgio2audio_source_get,
 259         .put            = sgio2audio_source_put,
 260 };
 261
 262 /* line mixer control */
 263 static const struct snd_kcontrol_new sgio2audio_ctrl_line = {
 264         .iface          = SNDRV_CTL_ELEM_IFACE_MIXER,
 265         .name           = "Line Playback Volume",
 266         .index          = 0,
 267         .access         = SNDRV_CTL_ELEM_ACCESS_READWRITE,
 268         .private_value  = AD1843_GAIN_LINE,
 269         .info           = sgio2audio_gain_info,
 270         .get            = sgio2audio_gain_get,
 271         .put            = sgio2audio_gain_put,
 272 };
 273
 274 /* cd mixer control */
 275 static const struct snd_kcontrol_new sgio2audio_ctrl_cd = {
 276         .iface          = SNDRV_CTL_ELEM_IFACE_MIXER,
 277         .name           = "Line Playback Volume",
 278         .index          = 1,
 279         .access         = SNDRV_CTL_ELEM_ACCESS_READWRITE,
 280         .private_value  = AD1843_GAIN_LINE_2,
 281         .info           = sgio2audio_gain_info,
 282         .get            = sgio2audio_gain_get,
 283         .put            = sgio2audio_gain_put,
 284 };
 285
 286 /* mic mixer control */
 287 static const struct snd_kcontrol_new sgio2audio_ctrl_mic = {
 288         .iface          = SNDRV_CTL_ELEM_IFACE_MIXER,
 289         .name           = "Mic Playback Volume",
 290         .access         = SNDRV_CTL_ELEM_ACCESS_READWRITE,
 291         .private_value  = AD1843_GAIN_MIC,
 292         .info           = sgio2audio_gain_info,
 293         .get            = sgio2audio_gain_get,
 294         .put            = sgio2audio_gain_put,
 295 };
 296
 297
 298 static int snd_sgio2audio_new_mixer(struct snd_sgio2audio *chip)
 299 {
 300         int err;
 301
 302         err = snd_ctl_add(chip->card,
 303                           snd_ctl_new1(&sgio2audio_ctrl_pcm0, chip));
 304         if (err < 0)
 305                 return err;
 306
 307         err = snd_ctl_add(chip->card,
 308                           snd_ctl_new1(&sgio2audio_ctrl_pcm1, chip));
 309         if (err < 0)
 310                 return err;
 311
 312         err = snd_ctl_add(chip->card,
 313                           snd_ctl_new1(&sgio2audio_ctrl_reclevel, chip));
 314         if (err < 0)
 315                 return err;
 316
 317         err = snd_ctl_add(chip->card,
 318                           snd_ctl_new1(&sgio2audio_ctrl_recsource, chip));
 319         if (err < 0)
 320                 return err;
 321         err = snd_ctl_add(chip->card,
 322                           snd_ctl_new1(&sgio2audio_ctrl_line, chip));
 323         if (err < 0)
 324                 return err;
 325
 326         err = snd_ctl_add(chip->card,
 327                           snd_ctl_new1(&sgio2audio_ctrl_cd, chip));
 328         if (err < 0)
 329                 return err;
 330
 331         err = snd_ctl_add(chip->card,
 332                           snd_ctl_new1(&sgio2audio_ctrl_mic, chip));
 333         if (err < 0)
 334                 return err;
 335
 336         return 0;
 337 }
 338
 339 /* low-level audio interface DMA */
 340
 341 /* get data out of bounce buffer, count must be a multiple of 32 */
 342 /* returns 1 if a period has elapsed */
 343 static int snd_sgio2audio_dma_pull_frag(struct snd_sgio2audio *chip,
 344                                         unsigned int ch, unsigned int count)
 345 {
 346         int ret;
 347         unsigned long src_base, src_pos, dst_mask;
 348         unsigned char *dst_base;
 349         int dst_pos;
 350         u64 *src;
 351         s16 *dst;
 352         u64 x;
 353         unsigned long flags;
 354         struct snd_pcm_runtime *runtime = chip->channel[ch].substream->runtime;
 355
 356         spin_lock_irqsave(&chip->channel[ch].lock, flags);
 357
 358         src_base = (unsigned long) chip->ring_base | (ch << CHANNEL_RING_SHIFT);
 359         src_pos = readq(&mace->perif.audio.chan[ch].read_ptr);
 360         dst_base = runtime->dma_area;
 361         dst_pos = chip->channel[ch].pos;
 362         dst_mask = frames_to_bytes(runtime, runtime->buffer_size) - 1;
 363
 364         /* check if a period has elapsed */
 365         chip->channel[ch].size += (count >> 3); /* in frames */
 366         ret = chip->channel[ch].size >= runtime->period_size;
 367         chip->channel[ch].size %= runtime->period_size;
 368
 369         while (count) {
 370                 src = (u64 *)(src_base + src_pos);
 371                 dst = (s16 *)(dst_base + dst_pos);
 372
 373                 x = *src;
 374                 dst[0] = (x >> CHANNEL_LEFT_SHIFT) & 0xffff;
 375                 dst[1] = (x >> CHANNEL_RIGHT_SHIFT) & 0xffff;
 376
 377                 src_pos = (src_pos + sizeof(u64)) & CHANNEL_RING_MASK;
 378                 dst_pos = (dst_pos + 2 * sizeof(s16)) & dst_mask;
 379                 count -= sizeof(u64);
 380         }
 381
 382         writeq(src_pos, &mace->perif.audio.chan[ch].read_ptr); /* in bytes */
 383         chip->channel[ch].pos = dst_pos;
 384
 385         spin_unlock_irqrestore(&chip->channel[ch].lock, flags);
 386         return ret;
 387 }
 388
 389 /* put some DMA data in bounce buffer, count must be a multiple of 32 */
 390 /* returns 1 if a period has elapsed */
 391 static int snd_sgio2audio_dma_push_frag(struct snd_sgio2audio *chip,
 392                                         unsigned int ch, unsigned int count)
 393 {
 394         int ret;
 395         s64 l, r;
 396         unsigned long dst_base, dst_pos, src_mask;
 397         unsigned char *src_base;
 398         int src_pos;
 399         u64 *dst;
 400         s16 *src;
 401         unsigned long flags;
 402         struct snd_pcm_runtime *runtime = chip->channel[ch].substream->runtime;
 403
 404         spin_lock_irqsave(&chip->channel[ch].lock, flags);
 405
 406         dst_base = (unsigned long)chip->ring_base | (ch << CHANNEL_RING_SHIFT);
 407         dst_pos = readq(&mace->perif.audio.chan[ch].write_ptr);
 408         src_base = runtime->dma_area;
 409         src_pos = chip->channel[ch].pos;
 410         src_mask = frames_to_bytes(runtime, runtime->buffer_size) - 1;
 411
 412         /* check if a period has elapsed */
 413         chip->channel[ch].size += (count >> 3); /* in frames */
 414         ret = chip->channel[ch].size >= runtime->period_size;
 415         chip->channel[ch].size %= runtime->period_size;
 416
 417         while (count) {
 418                 src = (s16 *)(src_base + src_pos);
 419                 dst = (u64 *)(dst_base + dst_pos);
 420
 421                 l = src[0]; /* sign extend */
 422                 r = src[1]; /* sign extend */
 423
 424                 *dst = ((l & 0x00ffffff) << CHANNEL_LEFT_SHIFT) |
 425                         ((r & 0x00ffffff) << CHANNEL_RIGHT_SHIFT);
 426
 427                 dst_pos = (dst_pos + sizeof(u64)) & CHANNEL_RING_MASK;
 428                 src_pos = (src_pos + 2 * sizeof(s16)) & src_mask;
 429                 count -= sizeof(u64);
 430         }
 431
 432         writeq(dst_pos, &mace->perif.audio.chan[ch].write_ptr); /* in bytes */
 433         chip->channel[ch].pos = src_pos;
 434
 435         spin_unlock_irqrestore(&chip->channel[ch].lock, flags);
 436         return ret;
 437 }
 438
 439 static int snd_sgio2audio_dma_start(struct snd_pcm_substream *substream)
 440 {
 441         struct snd_sgio2audio *chip = snd_pcm_substream_chip(substream);
 442         struct snd_sgio2audio_chan *chan = substream->runtime->private_data;
 443         int ch = chan->idx;
 444
 445         /* reset DMA channel */
 446         writeq(CHANNEL_CONTROL_RESET, &mace->perif.audio.chan[ch].control);
 447         udelay(10);
 448         writeq(0, &mace->perif.audio.chan[ch].control);
 449
 450         if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
 451                 /* push a full buffer */
 452                 snd_sgio2audio_dma_push_frag(chip, ch, CHANNEL_RING_SIZE - 32);
 453         }
 454         /* set DMA to wake on 50% empty and enable interrupt */
 455         writeq(CHANNEL_DMA_ENABLE | CHANNEL_INT_THRESHOLD_50,
 456                &mace->perif.audio.chan[ch].control);
 457         return 0;
 458 }
 459
 460 static int snd_sgio2audio_dma_stop(struct snd_pcm_substream *substream)
 461 {
 462         struct snd_sgio2audio_chan *chan = substream->runtime->private_data;
 463
 464         writeq(0, &mace->perif.audio.chan[chan->idx].control);
 465         return 0;
 466 }
 467
 468 static irqreturn_t snd_sgio2audio_dma_in_isr(int irq, void *dev_id)
 469 {
 470         struct snd_sgio2audio_chan *chan = dev_id;
 471         struct snd_pcm_substream *substream;
 472         struct snd_sgio2audio *chip;
 473         int count, ch;
 474
 475         substream = chan->substream;
 476         chip = snd_pcm_substream_chip(substream);
 477         ch = chan->idx;
 478
 479         /* empty the ring */
 480         count = CHANNEL_RING_SIZE -
 481                 readq(&mace->perif.audio.chan[ch].depth) - 32;
 482         if (snd_sgio2audio_dma_pull_frag(chip, ch, count))
 483                 snd_pcm_period_elapsed(substream);
 484
 485         return IRQ_HANDLED;
 486 }
 487
 488 static irqreturn_t snd_sgio2audio_dma_out_isr(int irq, void *dev_id)
 489 {
 490         struct snd_sgio2audio_chan *chan = dev_id;
 491         struct snd_pcm_substream *substream;
 492         struct snd_sgio2audio *chip;
 493         int count, ch;
 494
 495         substream = chan->substream;
 496         chip = snd_pcm_substream_chip(substream);
 497         ch = chan->idx;
 498         /* fill the ring */
 499         count = CHANNEL_RING_SIZE -
 500                 readq(&mace->perif.audio.chan[ch].depth) - 32;
 501         if (snd_sgio2audio_dma_push_frag(chip, ch, count))
 502                 snd_pcm_period_elapsed(substream);
 503
 504         return IRQ_HANDLED;
 505 }
 506
 507 static irqreturn_t snd_sgio2audio_error_isr(int irq, void *dev_id)
 508 {
 509         struct snd_sgio2audio_chan *chan = dev_id;
 510         struct snd_pcm_substream *substream;
 511
 512         substream = chan->substream;
 513         snd_sgio2audio_dma_stop(substream);
 514         snd_sgio2audio_dma_start(substream);
 515         return IRQ_HANDLED;
 516 }
 517
 518 /* PCM part */
 519 /* PCM hardware definition */
 520 static const struct snd_pcm_hardware snd_sgio2audio_pcm_hw = {
 521         .info = (SNDRV_PCM_INFO_MMAP |
 522                  SNDRV_PCM_INFO_MMAP_VALID |
 523                  SNDRV_PCM_INFO_INTERLEAVED |
 524                  SNDRV_PCM_INFO_BLOCK_TRANSFER),
 525         .formats =          SNDRV_PCM_FMTBIT_S16_BE,
 526         .rates =            SNDRV_PCM_RATE_8000_48000,
 527         .rate_min =         8000,
 528         .rate_max =         48000,
 529         .channels_min =     2,
 530         .channels_max =     2,
 531         .buffer_bytes_max = 65536,
 532         .period_bytes_min = 32768,
 533         .period_bytes_max = 65536,
 534         .periods_min =      1,
 535         .periods_max =      1024,
 536 };
 537
 538 /* PCM playback open callback */
 539 static int snd_sgio2audio_playback1_open(struct snd_pcm_substream *substream)
 540 {
 541         struct snd_sgio2audio *chip = snd_pcm_substream_chip(substream);
 542         struct snd_pcm_runtime *runtime = substream->runtime;
 543
 544         runtime->hw = snd_sgio2audio_pcm_hw;
 545         runtime->private_data = &chip->channel[1];
 546         return 0;
 547 }
 548
 549 static int snd_sgio2audio_playback2_open(struct snd_pcm_substream *substream)
 550 {
 551         struct snd_sgio2audio *chip = snd_pcm_substream_chip(substream);
 552         struct snd_pcm_runtime *runtime = substream->runtime;
 553
 554         runtime->hw = snd_sgio2audio_pcm_hw;
 555         runtime->private_data = &chip->channel[2];
 556         return 0;
 557 }
 558
 559 /* PCM capture open callback */
 560 static int snd_sgio2audio_capture_open(struct snd_pcm_substream *substream)
 561 {
 562         struct snd_sgio2audio *chip = snd_pcm_substream_chip(substream);
 563         struct snd_pcm_runtime *runtime = substream->runtime;
 564
 565         runtime->hw = snd_sgio2audio_pcm_hw;
 566         runtime->private_data = &chip->channel[0];
 567         return 0;
 568 }
 569
 570 /* PCM close callback */
 571 static int snd_sgio2audio_pcm_close(struct snd_pcm_substream *substream)
 572 {
 573         struct snd_pcm_runtime *runtime = substream->runtime;
 574
 575         runtime->private_data = NULL;
 576         return 0;
 577 }
 578
 579 /* prepare callback */
 580 static int snd_sgio2audio_pcm_prepare(struct snd_pcm_substream *substream)
 581 {
 582         struct snd_sgio2audio *chip = snd_pcm_substream_chip(substream);
 583         struct snd_pcm_runtime *runtime = substream->runtime;
 584         struct snd_sgio2audio_chan *chan = substream->runtime->private_data;
 585         int ch = chan->idx;
 586         unsigned long flags;
 587
 588         spin_lock_irqsave(&chip->channel[ch].lock, flags);
 589
 590         /* Setup the pseudo-dma transfer pointers.  */
 591         chip->channel[ch].pos = 0;
 592         chip->channel[ch].size = 0;
 593         chip->channel[ch].substream = substream;
 594
 595         /* set AD1843 format */
 596         /* hardware format is always S16_LE */
 597         switch (substream->stream) {
 598         case SNDRV_PCM_STREAM_PLAYBACK:
 599                 ad1843_setup_dac(&chip->ad1843,
 600                                  ch - 1,
 601                                  runtime->rate,
 602                                  SNDRV_PCM_FORMAT_S16_LE,
 603                                  runtime->channels);
 604                 break;
 605         case SNDRV_PCM_STREAM_CAPTURE:
 606                 ad1843_setup_adc(&chip->ad1843,
 607                                  runtime->rate,
 608                                  SNDRV_PCM_FORMAT_S16_LE,
 609                                  runtime->channels);
 610                 break;
 611         }
 612         spin_unlock_irqrestore(&chip->channel[ch].lock, flags);
 613         return 0;
 614 }
 615
 616 /* trigger callback */
 617 static int snd_sgio2audio_pcm_trigger(struct snd_pcm_substream *substream,
 618                                       int cmd)
 619 {
 620         switch (cmd) {
 621         case SNDRV_PCM_TRIGGER_START:
 622                 /* start the PCM engine */
 623                 snd_sgio2audio_dma_start(substream);
 624                 break;
 625         case SNDRV_PCM_TRIGGER_STOP:
 626                 /* stop the PCM engine */
 627                 snd_sgio2audio_dma_stop(substream);
 628                 break;
 629         default:
 630                 return -EINVAL;
 631         }
 632         return 0;
 633 }
 634
 635 /* pointer callback */
 636 static snd_pcm_uframes_t
 637 snd_sgio2audio_pcm_pointer(struct snd_pcm_substream *substream)
 638 {
 639         struct snd_sgio2audio *chip = snd_pcm_substream_chip(substream);
 640         struct snd_sgio2audio_chan *chan = substream->runtime->private_data;
 641
 642         /* get the current hardware pointer */
 643         return bytes_to_frames(substream->runtime,
 644                                chip->channel[chan->idx].pos);
 645 }
 646
 647 /* operators */
 648 static const struct snd_pcm_ops snd_sgio2audio_playback1_ops = {
 649         .open =        snd_sgio2audio_playback1_open,
 650         .close =       snd_sgio2audio_pcm_close,
 651         .prepare =     snd_sgio2audio_pcm_prepare,
 652         .trigger =     snd_sgio2audio_pcm_trigger,
 653         .pointer =     snd_sgio2audio_pcm_pointer,
 654 };
 655
 656 static const struct snd_pcm_ops snd_sgio2audio_playback2_ops = {
 657         .open =        snd_sgio2audio_playback2_open,
 658         .close =       snd_sgio2audio_pcm_close,
 659         .prepare =     snd_sgio2audio_pcm_prepare,
 660         .trigger =     snd_sgio2audio_pcm_trigger,
 661         .pointer =     snd_sgio2audio_pcm_pointer,
 662 };
 663
 664 static const struct snd_pcm_ops snd_sgio2audio_capture_ops = {
 665         .open =        snd_sgio2audio_capture_open,
 666         .close =       snd_sgio2audio_pcm_close,
 667         .prepare =     snd_sgio2audio_pcm_prepare,
 668         .trigger =     snd_sgio2audio_pcm_trigger,
 669         .pointer =     snd_sgio2audio_pcm_pointer,
 670 };
 671
 672 /*
 673  *  definitions of capture are omitted here...
 674  */
 675
 676 /* create a pcm device */
 677 static int snd_sgio2audio_new_pcm(struct snd_sgio2audio *chip)
 678 {
 679         struct snd_pcm *pcm;
 680         int err;
 681
 682         /* create first pcm device with one outputs and one input */
 683         err = snd_pcm_new(chip->card, "SGI O2 Audio", 0, 1, 1, &pcm);
 684         if (err < 0)
 685                 return err;
 686
 687         pcm->private_data = chip;
 688         strcpy(pcm->name, "SGI O2 DAC1");
 689
 690         /* set operators */
 691         snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK,
 692                         &snd_sgio2audio_playback1_ops);
 693         snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE,
 694                         &snd_sgio2audio_capture_ops);
 695         snd_pcm_set_managed_buffer_all(pcm, SNDRV_DMA_TYPE_VMALLOC, NULL, 0, 0);
 696
 697         /* create second  pcm device with one outputs and no input */
 698         err = snd_pcm_new(chip->card, "SGI O2 Audio", 1, 1, 0, &pcm);
 699         if (err < 0)
 700                 return err;
 701
 702         pcm->private_data = chip;
 703         strcpy(pcm->name, "SGI O2 DAC2");
 704
 705         /* set operators */
 706         snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK,
 707                         &snd_sgio2audio_playback2_ops);
 708         snd_pcm_set_managed_buffer_all(pcm, SNDRV_DMA_TYPE_VMALLOC, NULL, 0, 0);
 709
 710         return 0;
 711 }
 712
 713 static struct {
 714         int idx;
 715         int irq;
 716         irqreturn_t (*isr)(int, void *);
 717         const char *desc;
 718 } snd_sgio2_isr_table[] = {
 719         {
 720                 .idx = 0,
 721                 .irq = MACEISA_AUDIO1_DMAT_IRQ,
 722                 .isr = snd_sgio2audio_dma_in_isr,
 723                 .desc = "Capture DMA Channel 0"
 724         }, {
 725                 .idx = 0,
 726                 .irq = MACEISA_AUDIO1_OF_IRQ,
 727                 .isr = snd_sgio2audio_error_isr,
 728                 .desc = "Capture Overflow"
 729         }, {
 730                 .idx = 1,
 731                 .irq = MACEISA_AUDIO2_DMAT_IRQ,
 732                 .isr = snd_sgio2audio_dma_out_isr,
 733                 .desc = "Playback DMA Channel 1"
 734         }, {
 735                 .idx = 1,
 736                 .irq = MACEISA_AUDIO2_MERR_IRQ,
 737                 .isr = snd_sgio2audio_error_isr,
 738                 .desc = "Memory Error Channel 1"
 739         }, {
 740                 .idx = 2,
 741                 .irq = MACEISA_AUDIO3_DMAT_IRQ,
 742                 .isr = snd_sgio2audio_dma_out_isr,
 743                 .desc = "Playback DMA Channel 2"
 744         }, {
 745                 .idx = 2,
 746                 .irq = MACEISA_AUDIO3_MERR_IRQ,
 747                 .isr = snd_sgio2audio_error_isr,
 748                 .desc = "Memory Error Channel 2"
 749         }
 750 };
 751
 752 /* ALSA driver */
 753
 754 static int snd_sgio2audio_free(struct snd_sgio2audio *chip)
 755 {
 756         int i;
 757
 758         /* reset interface */
 759         writeq(AUDIO_CONTROL_RESET, &mace->perif.audio.control);
 760         udelay(1);
 761         writeq(0, &mace->perif.audio.control);
 762
 763         /* release IRQ's */
 764         for (i = 0; i < ARRAY_SIZE(snd_sgio2_isr_table); i++)
 765                 free_irq(snd_sgio2_isr_table[i].irq,
 766                          &chip->channel[snd_sgio2_isr_table[i].idx]);
 767
 768         dma_free_coherent(chip->card->dev, MACEISA_RINGBUFFERS_SIZE,
 769                           chip->ring_base, chip->ring_base_dma);
 770
 771         /* release card data */
 772         kfree(chip);
 773         return 0;
 774 }
 775
 776 static int snd_sgio2audio_dev_free(struct snd_device *device)
 777 {
 778         struct snd_sgio2audio *chip = device->device_data;
 779
 780         return snd_sgio2audio_free(chip);
 781 }
 782
 783 static const struct snd_device_ops ops = {
 784         .dev_free = snd_sgio2audio_dev_free,
 785 };
 786
 787 static int snd_sgio2audio_create(struct snd_card *card,
 788                                  struct snd_sgio2audio **rchip)
 789 {
 790         struct snd_sgio2audio *chip;
 791         int i, err;
 792
 793         *rchip = NULL;
 794
 795         /* check if a codec is attached to the interface */
 796         /* (Audio or Audio/Video board present) */
 797         if (!(readq(&mace->perif.audio.control) & AUDIO_CONTROL_CODEC_PRESENT))
 798                 return -ENOENT;
 799
 800         chip = kzalloc(sizeof(*chip), GFP_KERNEL);
 801         if (chip == NULL)
 802                 return -ENOMEM;
 803
 804         chip->card = card;
 805
 806         chip->ring_base = dma_alloc_coherent(card->dev,
 807                                              MACEISA_RINGBUFFERS_SIZE,
 808                                              &chip->ring_base_dma, GFP_KERNEL);
 809         if (chip->ring_base == NULL) {
 810                 printk(KERN_ERR
 811                        "sgio2audio: could not allocate ring buffers\n");
 812                 kfree(chip);
 813                 return -ENOMEM;
 814         }
 815
 816         spin_lock_init(&chip->ad1843_lock);
 817
 818         /* initialize channels */
 819         for (i = 0; i < 3; i++) {
 820                 spin_lock_init(&chip->channel[i].lock);
 821                 chip->channel[i].idx = i;
 822         }
 823
 824         /* allocate IRQs */
 825         for (i = 0; i < ARRAY_SIZE(snd_sgio2_isr_table); i++) {
 826                 if (request_irq(snd_sgio2_isr_table[i].irq,
 827                                 snd_sgio2_isr_table[i].isr,
 828                                 0,
 829                                 snd_sgio2_isr_table[i].desc,
 830                                 &chip->channel[snd_sgio2_isr_table[i].idx])) {
 831                         snd_sgio2audio_free(chip);
 832                         printk(KERN_ERR "sgio2audio: cannot allocate irq %d\n",
 833                                snd_sgio2_isr_table[i].irq);
 834                         return -EBUSY;
 835                 }
 836         }
 837
 838         /* reset the interface */
 839         writeq(AUDIO_CONTROL_RESET, &mace->perif.audio.control);
 840         udelay(1);
 841         writeq(0, &mace->perif.audio.control);
 842         msleep_interruptible(1); /* give time to recover */
 843
 844         /* set ring base */
 845         writeq(chip->ring_base_dma, &mace->perif.ctrl.ringbase);
 846
 847         /* attach the AD1843 codec */
 848         chip->ad1843.read = read_ad1843_reg;
 849         chip->ad1843.write = write_ad1843_reg;
 850         chip->ad1843.chip = chip;
 851
 852         /* initialize the AD1843 codec */
 853         err = ad1843_init(&chip->ad1843);
 854         if (err < 0) {
 855                 snd_sgio2audio_free(chip);
 856                 return err;
 857         }
 858
 859         err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops);
 860         if (err < 0) {
 861                 snd_sgio2audio_free(chip);
 862                 return err;
 863         }
 864         *rchip = chip;
 865         return 0;
 866 }
 867
 868 static int snd_sgio2audio_probe(struct platform_device *pdev)
 869 {
 870         struct snd_card *card;
 871         struct snd_sgio2audio *chip;
 872         int err;
 873
 874         err = snd_card_new(&pdev->dev, index, id, THIS_MODULE, 0, &card);
 875         if (err < 0)
 876                 return err;
 877
 878         err = snd_sgio2audio_create(card, &chip);
 879         if (err < 0) {
 880                 snd_card_free(card);
 881                 return err;
 882         }
 883
 884         err = snd_sgio2audio_new_pcm(chip);
 885         if (err < 0) {
 886                 snd_card_free(card);
 887                 return err;
 888         }
 889         err = snd_sgio2audio_new_mixer(chip);
 890         if (err < 0) {
 891                 snd_card_free(card);
 892                 return err;
 893         }
 894
 895         strcpy(card->driver, "SGI O2 Audio");
 896         strcpy(card->shortname, "SGI O2 Audio");
 897         sprintf(card->longname, "%s irq %i-%i",
 898                 card->shortname,
 899                 MACEISA_AUDIO1_DMAT_IRQ,
 900                 MACEISA_AUDIO3_MERR_IRQ);
 901
 902         err = snd_card_register(card);
 903         if (err < 0) {
 904                 snd_card_free(card);
 905                 return err;
 906         }
 907         platform_set_drvdata(pdev, card);
 908         return 0;
 909 }
 910
 911 static int snd_sgio2audio_remove(struct platform_device *pdev)
 912 {
 913         struct snd_card *card = platform_get_drvdata(pdev);
 914
 915         snd_card_free(card);
 916         return 0;
 917 }
 918
 919 static struct platform_driver sgio2audio_driver = {
 920         .probe  = snd_sgio2audio_probe,
 921         .remove = snd_sgio2audio_remove,
 922         .driver = {
 923                 .name   = "sgio2audio",
 924         }
 925 };
 926
 927 module_platform_driver(sgio2audio_driver);