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 MODULE_SUPPORTED_DEVICE("{{Silicon Graphics, O2 Audio}}");
  36
  37 static int index = SNDRV_DEFAULT_IDX1;  /* Index 0-MAX */
  38 static char *id = SNDRV_DEFAULT_STR1;   /* ID for this card */
  39
  40 module_param(index, int, 0444);
  41 MODULE_PARM_DESC(index, "Index value for SGI O2 soundcard.");
  42 module_param(id, charp, 0444);
  43 MODULE_PARM_DESC(id, "ID string for SGI O2 soundcard.");
  44
  45
  46 #define AUDIO_CONTROL_RESET              BIT(0) /* 1: reset audio interface */
  47 #define AUDIO_CONTROL_CODEC_PRESENT      BIT(1) /* 1: codec detected */
  48
  49 #define CODEC_CONTROL_WORD_SHIFT        0
  50 #define CODEC_CONTROL_READ              BIT(16)
  51 #define CODEC_CONTROL_ADDRESS_SHIFT     17
  52
  53 #define CHANNEL_CONTROL_RESET           BIT(10) /* 1: reset channel */
  54 #define CHANNEL_DMA_ENABLE              BIT(9)  /* 1: enable DMA transfer */
  55 #define CHANNEL_INT_THRESHOLD_DISABLED  (0 << 5) /* interrupt disabled */
  56 #define CHANNEL_INT_THRESHOLD_25        (1 << 5) /* int on buffer >25% full */
  57 #define CHANNEL_INT_THRESHOLD_50        (2 << 5) /* int on buffer >50% full */
  58 #define CHANNEL_INT_THRESHOLD_75        (3 << 5) /* int on buffer >75% full */
  59 #define CHANNEL_INT_THRESHOLD_EMPTY     (4 << 5) /* int on buffer empty */
  60 #define CHANNEL_INT_THRESHOLD_NOT_EMPTY (5 << 5) /* int on buffer !empty */
  61 #define CHANNEL_INT_THRESHOLD_FULL      (6 << 5) /* int on buffer empty */
  62 #define CHANNEL_INT_THRESHOLD_NOT_FULL  (7 << 5) /* int on buffer !empty */
  63
  64 #define CHANNEL_RING_SHIFT              12
  65 #define CHANNEL_RING_SIZE               (1 << CHANNEL_RING_SHIFT)
  66 #define CHANNEL_RING_MASK               (CHANNEL_RING_SIZE - 1)
  67
  68 #define CHANNEL_LEFT_SHIFT 40
  69 #define CHANNEL_RIGHT_SHIFT 8
  70
  71 struct snd_sgio2audio_chan {
  72         int idx;
  73         struct snd_pcm_substream *substream;
  74         int pos;
  75         snd_pcm_uframes_t size;
  76         spinlock_t lock;
  77 };
  78
  79 /* definition of the chip-specific record */
  80 struct snd_sgio2audio {
  81         struct snd_card *card;
  82
  83         /* codec */
  84         struct snd_ad1843 ad1843;
  85         spinlock_t ad1843_lock;
  86
  87         /* channels */
  88         struct snd_sgio2audio_chan channel[3];
  89
  90         /* resources */
  91         void *ring_base;
  92         dma_addr_t ring_base_dma;
  93 };
  94
  95 /* AD1843 access */
  96
  97 /*
  98  * read_ad1843_reg returns the current contents of a 16 bit AD1843 register.
  99  *
 100  * Returns unsigned register value on success, -errno on failure.
 101  */
 102 static int read_ad1843_reg(void *priv, int reg)
 103 {
 104         struct snd_sgio2audio *chip = priv;
 105         int val;
 106         unsigned long flags;
 107
 108         spin_lock_irqsave(&chip->ad1843_lock, flags);
 109
 110         writeq((reg << CODEC_CONTROL_ADDRESS_SHIFT) |
 111                CODEC_CONTROL_READ, &mace->perif.audio.codec_control);
 112         wmb();
 113         val = readq(&mace->perif.audio.codec_control); /* flush bus */
 114         udelay(200);
 115
 116         val = readq(&mace->perif.audio.codec_read);
 117
 118         spin_unlock_irqrestore(&chip->ad1843_lock, flags);
 119         return val;
 120 }
 121
 122 /*
 123  * write_ad1843_reg writes the specified value to a 16 bit AD1843 register.
 124  */
 125 static int write_ad1843_reg(void *priv, int reg, int word)
 126 {
 127         struct snd_sgio2audio *chip = priv;
 128         int val;
 129         unsigned long flags;
 130
 131         spin_lock_irqsave(&chip->ad1843_lock, flags);
 132
 133         writeq((reg << CODEC_CONTROL_ADDRESS_SHIFT) |
 134                (word << CODEC_CONTROL_WORD_SHIFT),
 135                &mace->perif.audio.codec_control);
 136         wmb();
 137         val = readq(&mace->perif.audio.codec_control); /* flush bus */
 138         udelay(200);
 139
 140         spin_unlock_irqrestore(&chip->ad1843_lock, flags);
 141         return 0;
 142 }
 143
 144 static int sgio2audio_gain_info(struct snd_kcontrol *kcontrol,
 145                                struct snd_ctl_elem_info *uinfo)
 146 {
 147         struct snd_sgio2audio *chip = snd_kcontrol_chip(kcontrol);
 148
 149         uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
 150         uinfo->count = 2;
 151         uinfo->value.integer.min = 0;
 152         uinfo->value.integer.max = ad1843_get_gain_max(&chip->ad1843,
 153                                              (int)kcontrol->private_value);
 154         return 0;
 155 }
 156
 157 static int sgio2audio_gain_get(struct snd_kcontrol *kcontrol,
 158                                struct snd_ctl_elem_value *ucontrol)
 159 {
 160         struct snd_sgio2audio *chip = snd_kcontrol_chip(kcontrol);
 161         int vol;
 162
 163         vol = ad1843_get_gain(&chip->ad1843, (int)kcontrol->private_value);
 164
 165         ucontrol->value.integer.value[0] = (vol >> 8) & 0xFF;
 166         ucontrol->value.integer.value[1] = vol & 0xFF;
 167
 168         return 0;
 169 }
 170
 171 static int sgio2audio_gain_put(struct snd_kcontrol *kcontrol,
 172                         struct snd_ctl_elem_value *ucontrol)
 173 {
 174         struct snd_sgio2audio *chip = snd_kcontrol_chip(kcontrol);
 175         int newvol, oldvol;
 176
 177         oldvol = ad1843_get_gain(&chip->ad1843, kcontrol->private_value);
 178         newvol = (ucontrol->value.integer.value[0] << 8) |
 179                 ucontrol->value.integer.value[1];
 180
 181         newvol = ad1843_set_gain(&chip->ad1843, kcontrol->private_value,
 182                 newvol);
 183
 184         return newvol != oldvol;
 185 }
 186
 187 static int sgio2audio_source_info(struct snd_kcontrol *kcontrol,
 188                                struct snd_ctl_elem_info *uinfo)
 189 {
 190         static const char * const texts[3] = {
 191                 "Cam Mic", "Mic", "Line"
 192         };
 193         return snd_ctl_enum_info(uinfo, 1, 3, texts);
 194 }
 195
 196 static int sgio2audio_source_get(struct snd_kcontrol *kcontrol,
 197                                struct snd_ctl_elem_value *ucontrol)
 198 {
 199         struct snd_sgio2audio *chip = snd_kcontrol_chip(kcontrol);
 200
 201         ucontrol->value.enumerated.item[0] = ad1843_get_recsrc(&chip->ad1843);
 202         return 0;
 203 }
 204
 205 static int sgio2audio_source_put(struct snd_kcontrol *kcontrol,
 206                         struct snd_ctl_elem_value *ucontrol)
 207 {
 208         struct snd_sgio2audio *chip = snd_kcontrol_chip(kcontrol);
 209         int newsrc, oldsrc;
 210
 211         oldsrc = ad1843_get_recsrc(&chip->ad1843);
 212         newsrc = ad1843_set_recsrc(&chip->ad1843,
 213                                    ucontrol->value.enumerated.item[0]);
 214
 215         return newsrc != oldsrc;
 216 }
 217
 218 /* dac1/pcm0 mixer control */
 219 static const struct snd_kcontrol_new sgio2audio_ctrl_pcm0 = {
 220         .iface          = SNDRV_CTL_ELEM_IFACE_MIXER,
 221         .name           = "PCM Playback Volume",
 222         .index          = 0,
 223         .access         = SNDRV_CTL_ELEM_ACCESS_READWRITE,
 224         .private_value  = AD1843_GAIN_PCM_0,
 225         .info           = sgio2audio_gain_info,
 226         .get            = sgio2audio_gain_get,
 227         .put            = sgio2audio_gain_put,
 228 };
 229
 230 /* dac2/pcm1 mixer control */
 231 static const struct snd_kcontrol_new sgio2audio_ctrl_pcm1 = {
 232         .iface          = SNDRV_CTL_ELEM_IFACE_MIXER,
 233         .name           = "PCM Playback Volume",
 234         .index          = 1,
 235         .access         = SNDRV_CTL_ELEM_ACCESS_READWRITE,
 236         .private_value  = AD1843_GAIN_PCM_1,
 237         .info           = sgio2audio_gain_info,
 238         .get            = sgio2audio_gain_get,
 239         .put            = sgio2audio_gain_put,
 240 };
 241
 242 /* record level mixer control */
 243 static const struct snd_kcontrol_new sgio2audio_ctrl_reclevel = {
 244         .iface          = SNDRV_CTL_ELEM_IFACE_MIXER,
 245         .name           = "Capture Volume",
 246         .access         = SNDRV_CTL_ELEM_ACCESS_READWRITE,
 247         .private_value  = AD1843_GAIN_RECLEV,
 248         .info           = sgio2audio_gain_info,
 249         .get            = sgio2audio_gain_get,
 250         .put            = sgio2audio_gain_put,
 251 };
 252
 253 /* record level source control */
 254 static const struct snd_kcontrol_new sgio2audio_ctrl_recsource = {
 255         .iface          = SNDRV_CTL_ELEM_IFACE_MIXER,
 256         .name           = "Capture Source",
 257         .access         = SNDRV_CTL_ELEM_ACCESS_READWRITE,
 258         .info           = sgio2audio_source_info,
 259         .get            = sgio2audio_source_get,
 260         .put            = sgio2audio_source_put,
 261 };
 262
 263 /* line mixer control */
 264 static const struct snd_kcontrol_new sgio2audio_ctrl_line = {
 265         .iface          = SNDRV_CTL_ELEM_IFACE_MIXER,
 266         .name           = "Line Playback Volume",
 267         .index          = 0,
 268         .access         = SNDRV_CTL_ELEM_ACCESS_READWRITE,
 269         .private_value  = AD1843_GAIN_LINE,
 270         .info           = sgio2audio_gain_info,
 271         .get            = sgio2audio_gain_get,
 272         .put            = sgio2audio_gain_put,
 273 };
 274
 275 /* cd mixer control */
 276 static const struct snd_kcontrol_new sgio2audio_ctrl_cd = {
 277         .iface          = SNDRV_CTL_ELEM_IFACE_MIXER,
 278         .name           = "Line Playback Volume",
 279         .index          = 1,
 280         .access         = SNDRV_CTL_ELEM_ACCESS_READWRITE,
 281         .private_value  = AD1843_GAIN_LINE_2,
 282         .info           = sgio2audio_gain_info,
 283         .get            = sgio2audio_gain_get,
 284         .put            = sgio2audio_gain_put,
 285 };
 286
 287 /* mic mixer control */
 288 static const struct snd_kcontrol_new sgio2audio_ctrl_mic = {
 289         .iface          = SNDRV_CTL_ELEM_IFACE_MIXER,
 290         .name           = "Mic Playback Volume",
 291         .access         = SNDRV_CTL_ELEM_ACCESS_READWRITE,
 292         .private_value  = AD1843_GAIN_MIC,
 293         .info           = sgio2audio_gain_info,
 294         .get            = sgio2audio_gain_get,
 295         .put            = sgio2audio_gain_put,
 296 };
 297
 298
 299 static int snd_sgio2audio_new_mixer(struct snd_sgio2audio *chip)
 300 {
 301         int err;
 302
 303         err = snd_ctl_add(chip->card,
 304                           snd_ctl_new1(&sgio2audio_ctrl_pcm0, chip));
 305         if (err < 0)
 306                 return err;
 307
 308         err = snd_ctl_add(chip->card,
 309                           snd_ctl_new1(&sgio2audio_ctrl_pcm1, chip));
 310         if (err < 0)
 311                 return err;
 312
 313         err = snd_ctl_add(chip->card,
 314                           snd_ctl_new1(&sgio2audio_ctrl_reclevel, chip));
 315         if (err < 0)
 316                 return err;
 317
 318         err = snd_ctl_add(chip->card,
 319                           snd_ctl_new1(&sgio2audio_ctrl_recsource, chip));
 320         if (err < 0)
 321                 return err;
 322         err = snd_ctl_add(chip->card,
 323                           snd_ctl_new1(&sgio2audio_ctrl_line, chip));
 324         if (err < 0)
 325                 return err;
 326
 327         err = snd_ctl_add(chip->card,
 328                           snd_ctl_new1(&sgio2audio_ctrl_cd, chip));
 329         if (err < 0)
 330                 return err;
 331
 332         err = snd_ctl_add(chip->card,
 333                           snd_ctl_new1(&sgio2audio_ctrl_mic, chip));
 334         if (err < 0)
 335                 return err;
 336
 337         return 0;
 338 }
 339
 340 /* low-level audio interface DMA */
 341
 342 /* get data out of bounce buffer, count must be a multiple of 32 */
 343 /* returns 1 if a period has elapsed */
 344 static int snd_sgio2audio_dma_pull_frag(struct snd_sgio2audio *chip,
 345                                         unsigned int ch, unsigned int count)
 346 {
 347         int ret;
 348         unsigned long src_base, src_pos, dst_mask;
 349         unsigned char *dst_base;
 350         int dst_pos;
 351         u64 *src;
 352         s16 *dst;
 353         u64 x;
 354         unsigned long flags;
 355         struct snd_pcm_runtime *runtime = chip->channel[ch].substream->runtime;
 356
 357         spin_lock_irqsave(&chip->channel[ch].lock, flags);
 358
 359         src_base = (unsigned long) chip->ring_base | (ch << CHANNEL_RING_SHIFT);
 360         src_pos = readq(&mace->perif.audio.chan[ch].read_ptr);
 361         dst_base = runtime->dma_area;
 362         dst_pos = chip->channel[ch].pos;
 363         dst_mask = frames_to_bytes(runtime, runtime->buffer_size) - 1;
 364
 365         /* check if a period has elapsed */
 366         chip->channel[ch].size += (count >> 3); /* in frames */
 367         ret = chip->channel[ch].size >= runtime->period_size;
 368         chip->channel[ch].size %= runtime->period_size;
 369
 370         while (count) {
 371                 src = (u64 *)(src_base + src_pos);
 372                 dst = (s16 *)(dst_base + dst_pos);
 373
 374                 x = *src;
 375                 dst[0] = (x >> CHANNEL_LEFT_SHIFT) & 0xffff;
 376                 dst[1] = (x >> CHANNEL_RIGHT_SHIFT) & 0xffff;
 377
 378                 src_pos = (src_pos + sizeof(u64)) & CHANNEL_RING_MASK;
 379                 dst_pos = (dst_pos + 2 * sizeof(s16)) & dst_mask;
 380                 count -= sizeof(u64);
 381         }
 382
 383         writeq(src_pos, &mace->perif.audio.chan[ch].read_ptr); /* in bytes */
 384         chip->channel[ch].pos = dst_pos;
 385
 386         spin_unlock_irqrestore(&chip->channel[ch].lock, flags);
 387         return ret;
 388 }
 389
 390 /* put some DMA data in bounce buffer, count must be a multiple of 32 */
 391 /* returns 1 if a period has elapsed */
 392 static int snd_sgio2audio_dma_push_frag(struct snd_sgio2audio *chip,
 393                                         unsigned int ch, unsigned int count)
 394 {
 395         int ret;
 396         s64 l, r;
 397         unsigned long dst_base, dst_pos, src_mask;
 398         unsigned char *src_base;
 399         int src_pos;
 400         u64 *dst;
 401         s16 *src;
 402         unsigned long flags;
 403         struct snd_pcm_runtime *runtime = chip->channel[ch].substream->runtime;
 404
 405         spin_lock_irqsave(&chip->channel[ch].lock, flags);
 406
 407         dst_base = (unsigned long)chip->ring_base | (ch << CHANNEL_RING_SHIFT);
 408         dst_pos = readq(&mace->perif.audio.chan[ch].write_ptr);
 409         src_base = runtime->dma_area;
 410         src_pos = chip->channel[ch].pos;
 411         src_mask = frames_to_bytes(runtime, runtime->buffer_size) - 1;
 412
 413         /* check if a period has elapsed */
 414         chip->channel[ch].size += (count >> 3); /* in frames */
 415         ret = chip->channel[ch].size >= runtime->period_size;
 416         chip->channel[ch].size %= runtime->period_size;
 417
 418         while (count) {
 419                 src = (s16 *)(src_base + src_pos);
 420                 dst = (u64 *)(dst_base + dst_pos);
 421
 422                 l = src[0]; /* sign extend */
 423                 r = src[1]; /* sign extend */
 424
 425                 *dst = ((l & 0x00ffffff) << CHANNEL_LEFT_SHIFT) |
 426                         ((r & 0x00ffffff) << CHANNEL_RIGHT_SHIFT);
 427
 428                 dst_pos = (dst_pos + sizeof(u64)) & CHANNEL_RING_MASK;
 429                 src_pos = (src_pos + 2 * sizeof(s16)) & src_mask;
 430                 count -= sizeof(u64);
 431         }
 432
 433         writeq(dst_pos, &mace->perif.audio.chan[ch].write_ptr); /* in bytes */
 434         chip->channel[ch].pos = src_pos;
 435
 436         spin_unlock_irqrestore(&chip->channel[ch].lock, flags);
 437         return ret;
 438 }
 439
 440 static int snd_sgio2audio_dma_start(struct snd_pcm_substream *substream)
 441 {
 442         struct snd_sgio2audio *chip = snd_pcm_substream_chip(substream);
 443         struct snd_sgio2audio_chan *chan = substream->runtime->private_data;
 444         int ch = chan->idx;
 445
 446         /* reset DMA channel */
 447         writeq(CHANNEL_CONTROL_RESET, &mace->perif.audio.chan[ch].control);
 448         udelay(10);
 449         writeq(0, &mace->perif.audio.chan[ch].control);
 450
 451         if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
 452                 /* push a full buffer */
 453                 snd_sgio2audio_dma_push_frag(chip, ch, CHANNEL_RING_SIZE - 32);
 454         }
 455         /* set DMA to wake on 50% empty and enable interrupt */
 456         writeq(CHANNEL_DMA_ENABLE | CHANNEL_INT_THRESHOLD_50,
 457                &mace->perif.audio.chan[ch].control);
 458         return 0;
 459 }
 460
 461 static int snd_sgio2audio_dma_stop(struct snd_pcm_substream *substream)
 462 {
 463         struct snd_sgio2audio_chan *chan = substream->runtime->private_data;
 464
 465         writeq(0, &mace->perif.audio.chan[chan->idx].control);
 466         return 0;
 467 }
 468
 469 static irqreturn_t snd_sgio2audio_dma_in_isr(int irq, void *dev_id)
 470 {
 471         struct snd_sgio2audio_chan *chan = dev_id;
 472         struct snd_pcm_substream *substream;
 473         struct snd_sgio2audio *chip;
 474         int count, ch;
 475
 476         substream = chan->substream;
 477         chip = snd_pcm_substream_chip(substream);
 478         ch = chan->idx;
 479
 480         /* empty the ring */
 481         count = CHANNEL_RING_SIZE -
 482                 readq(&mace->perif.audio.chan[ch].depth) - 32;
 483         if (snd_sgio2audio_dma_pull_frag(chip, ch, count))
 484                 snd_pcm_period_elapsed(substream);
 485
 486         return IRQ_HANDLED;
 487 }
 488
 489 static irqreturn_t snd_sgio2audio_dma_out_isr(int irq, void *dev_id)
 490 {
 491         struct snd_sgio2audio_chan *chan = dev_id;
 492         struct snd_pcm_substream *substream;
 493         struct snd_sgio2audio *chip;
 494         int count, ch;
 495
 496         substream = chan->substream;
 497         chip = snd_pcm_substream_chip(substream);
 498         ch = chan->idx;
 499         /* fill the ring */
 500         count = CHANNEL_RING_SIZE -
 501                 readq(&mace->perif.audio.chan[ch].depth) - 32;
 502         if (snd_sgio2audio_dma_push_frag(chip, ch, count))
 503                 snd_pcm_period_elapsed(substream);
 504
 505         return IRQ_HANDLED;
 506 }
 507
 508 static irqreturn_t snd_sgio2audio_error_isr(int irq, void *dev_id)
 509 {
 510         struct snd_sgio2audio_chan *chan = dev_id;
 511         struct snd_pcm_substream *substream;
 512
 513         substream = chan->substream;
 514         snd_sgio2audio_dma_stop(substream);
 515         snd_sgio2audio_dma_start(substream);
 516         return IRQ_HANDLED;
 517 }
 518
 519 /* PCM part */
 520 /* PCM hardware definition */
 521 static const struct snd_pcm_hardware snd_sgio2audio_pcm_hw = {
 522         .info = (SNDRV_PCM_INFO_MMAP |
 523                  SNDRV_PCM_INFO_MMAP_VALID |
 524                  SNDRV_PCM_INFO_INTERLEAVED |
 525                  SNDRV_PCM_INFO_BLOCK_TRANSFER),
 526         .formats =          SNDRV_PCM_FMTBIT_S16_BE,
 527         .rates =            SNDRV_PCM_RATE_8000_48000,
 528         .rate_min =         8000,
 529         .rate_max =         48000,
 530         .channels_min =     2,
 531         .channels_max =     2,
 532         .buffer_bytes_max = 65536,
 533         .period_bytes_min = 32768,
 534         .period_bytes_max = 65536,
 535         .periods_min =      1,
 536         .periods_max =      1024,
 537 };
 538
 539 /* PCM playback open callback */
 540 static int snd_sgio2audio_playback1_open(struct snd_pcm_substream *substream)
 541 {
 542         struct snd_sgio2audio *chip = snd_pcm_substream_chip(substream);
 543         struct snd_pcm_runtime *runtime = substream->runtime;
 544
 545         runtime->hw = snd_sgio2audio_pcm_hw;
 546         runtime->private_data = &chip->channel[1];
 547         return 0;
 548 }
 549
 550 static int snd_sgio2audio_playback2_open(struct snd_pcm_substream *substream)
 551 {
 552         struct snd_sgio2audio *chip = snd_pcm_substream_chip(substream);
 553         struct snd_pcm_runtime *runtime = substream->runtime;
 554
 555         runtime->hw = snd_sgio2audio_pcm_hw;
 556         runtime->private_data = &chip->channel[2];
 557         return 0;
 558 }
 559
 560 /* PCM capture open callback */
 561 static int snd_sgio2audio_capture_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[0];
 568         return 0;
 569 }
 570
 571 /* PCM close callback */
 572 static int snd_sgio2audio_pcm_close(struct snd_pcm_substream *substream)
 573 {
 574         struct snd_pcm_runtime *runtime = substream->runtime;
 575
 576         runtime->private_data = NULL;
 577         return 0;
 578 }
 579
 580
 581 /* hw_params callback */
 582 static int snd_sgio2audio_pcm_hw_params(struct snd_pcm_substream *substream,
 583                                         struct snd_pcm_hw_params *hw_params)
 584 {
 585         return snd_pcm_lib_alloc_vmalloc_buffer(substream,
 586                                                 params_buffer_bytes(hw_params));
 587 }
 588
 589 /* hw_free callback */
 590 static int snd_sgio2audio_pcm_hw_free(struct snd_pcm_substream *substream)
 591 {
 592         return snd_pcm_lib_free_vmalloc_buffer(substream);
 593 }
 594
 595 /* prepare callback */
 596 static int snd_sgio2audio_pcm_prepare(struct snd_pcm_substream *substream)
 597 {
 598         struct snd_sgio2audio *chip = snd_pcm_substream_chip(substream);
 599         struct snd_pcm_runtime *runtime = substream->runtime;
 600         struct snd_sgio2audio_chan *chan = substream->runtime->private_data;
 601         int ch = chan->idx;
 602         unsigned long flags;
 603
 604         spin_lock_irqsave(&chip->channel[ch].lock, flags);
 605
 606         /* Setup the pseudo-dma transfer pointers.  */
 607         chip->channel[ch].pos = 0;
 608         chip->channel[ch].size = 0;
 609         chip->channel[ch].substream = substream;
 610
 611         /* set AD1843 format */
 612         /* hardware format is always S16_LE */
 613         switch (substream->stream) {
 614         case SNDRV_PCM_STREAM_PLAYBACK:
 615                 ad1843_setup_dac(&chip->ad1843,
 616                                  ch - 1,
 617                                  runtime->rate,
 618                                  SNDRV_PCM_FORMAT_S16_LE,
 619                                  runtime->channels);
 620                 break;
 621         case SNDRV_PCM_STREAM_CAPTURE:
 622                 ad1843_setup_adc(&chip->ad1843,
 623                                  runtime->rate,
 624                                  SNDRV_PCM_FORMAT_S16_LE,
 625                                  runtime->channels);
 626                 break;
 627         }
 628         spin_unlock_irqrestore(&chip->channel[ch].lock, flags);
 629         return 0;
 630 }
 631
 632 /* trigger callback */
 633 static int snd_sgio2audio_pcm_trigger(struct snd_pcm_substream *substream,
 634                                       int cmd)
 635 {
 636         switch (cmd) {
 637         case SNDRV_PCM_TRIGGER_START:
 638                 /* start the PCM engine */
 639                 snd_sgio2audio_dma_start(substream);
 640                 break;
 641         case SNDRV_PCM_TRIGGER_STOP:
 642                 /* stop the PCM engine */
 643                 snd_sgio2audio_dma_stop(substream);
 644                 break;
 645         default:
 646                 return -EINVAL;
 647         }
 648         return 0;
 649 }
 650
 651 /* pointer callback */
 652 static snd_pcm_uframes_t
 653 snd_sgio2audio_pcm_pointer(struct snd_pcm_substream *substream)
 654 {
 655         struct snd_sgio2audio *chip = snd_pcm_substream_chip(substream);
 656         struct snd_sgio2audio_chan *chan = substream->runtime->private_data;
 657
 658         /* get the current hardware pointer */
 659         return bytes_to_frames(substream->runtime,
 660                                chip->channel[chan->idx].pos);
 661 }
 662
 663 /* operators */
 664 static const struct snd_pcm_ops snd_sgio2audio_playback1_ops = {
 665         .open =        snd_sgio2audio_playback1_open,
 666         .close =       snd_sgio2audio_pcm_close,
 667         .ioctl =       snd_pcm_lib_ioctl,
 668         .hw_params =   snd_sgio2audio_pcm_hw_params,
 669         .hw_free =     snd_sgio2audio_pcm_hw_free,
 670         .prepare =     snd_sgio2audio_pcm_prepare,
 671         .trigger =     snd_sgio2audio_pcm_trigger,
 672         .pointer =     snd_sgio2audio_pcm_pointer,
 673         .page =        snd_pcm_lib_get_vmalloc_page,
 674 };
 675
 676 static const struct snd_pcm_ops snd_sgio2audio_playback2_ops = {
 677         .open =        snd_sgio2audio_playback2_open,
 678         .close =       snd_sgio2audio_pcm_close,
 679         .ioctl =       snd_pcm_lib_ioctl,
 680         .hw_params =   snd_sgio2audio_pcm_hw_params,
 681         .hw_free =     snd_sgio2audio_pcm_hw_free,
 682         .prepare =     snd_sgio2audio_pcm_prepare,
 683         .trigger =     snd_sgio2audio_pcm_trigger,
 684         .pointer =     snd_sgio2audio_pcm_pointer,
 685         .page =        snd_pcm_lib_get_vmalloc_page,
 686 };
 687
 688 static const struct snd_pcm_ops snd_sgio2audio_capture_ops = {
 689         .open =        snd_sgio2audio_capture_open,
 690         .close =       snd_sgio2audio_pcm_close,
 691         .ioctl =       snd_pcm_lib_ioctl,
 692         .hw_params =   snd_sgio2audio_pcm_hw_params,
 693         .hw_free =     snd_sgio2audio_pcm_hw_free,
 694         .prepare =     snd_sgio2audio_pcm_prepare,
 695         .trigger =     snd_sgio2audio_pcm_trigger,
 696         .pointer =     snd_sgio2audio_pcm_pointer,
 697         .page =        snd_pcm_lib_get_vmalloc_page,
 698 };
 699
 700 /*
 701  *  definitions of capture are omitted here...
 702  */
 703
 704 /* create a pcm device */
 705 static int snd_sgio2audio_new_pcm(struct snd_sgio2audio *chip)
 706 {
 707         struct snd_pcm *pcm;
 708         int err;
 709
 710         /* create first pcm device with one outputs and one input */
 711         err = snd_pcm_new(chip->card, "SGI O2 Audio", 0, 1, 1, &pcm);
 712         if (err < 0)
 713                 return err;
 714
 715         pcm->private_data = chip;
 716         strcpy(pcm->name, "SGI O2 DAC1");
 717
 718         /* set operators */
 719         snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK,
 720                         &snd_sgio2audio_playback1_ops);
 721         snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE,
 722                         &snd_sgio2audio_capture_ops);
 723
 724         /* create second  pcm device with one outputs and no input */
 725         err = snd_pcm_new(chip->card, "SGI O2 Audio", 1, 1, 0, &pcm);
 726         if (err < 0)
 727                 return err;
 728
 729         pcm->private_data = chip;
 730         strcpy(pcm->name, "SGI O2 DAC2");
 731
 732         /* set operators */
 733         snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK,
 734                         &snd_sgio2audio_playback2_ops);
 735
 736         return 0;
 737 }
 738
 739 static struct {
 740         int idx;
 741         int irq;
 742         irqreturn_t (*isr)(int, void *);
 743         const char *desc;
 744 } snd_sgio2_isr_table[] = {
 745         {
 746                 .idx = 0,
 747                 .irq = MACEISA_AUDIO1_DMAT_IRQ,
 748                 .isr = snd_sgio2audio_dma_in_isr,
 749                 .desc = "Capture DMA Channel 0"
 750         }, {
 751                 .idx = 0,
 752                 .irq = MACEISA_AUDIO1_OF_IRQ,
 753                 .isr = snd_sgio2audio_error_isr,
 754                 .desc = "Capture Overflow"
 755         }, {
 756                 .idx = 1,
 757                 .irq = MACEISA_AUDIO2_DMAT_IRQ,
 758                 .isr = snd_sgio2audio_dma_out_isr,
 759                 .desc = "Playback DMA Channel 1"
 760         }, {
 761                 .idx = 1,
 762                 .irq = MACEISA_AUDIO2_MERR_IRQ,
 763                 .isr = snd_sgio2audio_error_isr,
 764                 .desc = "Memory Error Channel 1"
 765         }, {
 766                 .idx = 2,
 767                 .irq = MACEISA_AUDIO3_DMAT_IRQ,
 768                 .isr = snd_sgio2audio_dma_out_isr,
 769                 .desc = "Playback DMA Channel 2"
 770         }, {
 771                 .idx = 2,
 772                 .irq = MACEISA_AUDIO3_MERR_IRQ,
 773                 .isr = snd_sgio2audio_error_isr,
 774                 .desc = "Memory Error Channel 2"
 775         }
 776 };
 777
 778 /* ALSA driver */
 779
 780 static int snd_sgio2audio_free(struct snd_sgio2audio *chip)
 781 {
 782         int i;
 783
 784         /* reset interface */
 785         writeq(AUDIO_CONTROL_RESET, &mace->perif.audio.control);
 786         udelay(1);
 787         writeq(0, &mace->perif.audio.control);
 788
 789         /* release IRQ's */
 790         for (i = 0; i < ARRAY_SIZE(snd_sgio2_isr_table); i++)
 791                 free_irq(snd_sgio2_isr_table[i].irq,
 792                          &chip->channel[snd_sgio2_isr_table[i].idx]);
 793
 794         dma_free_coherent(chip->card->dev, MACEISA_RINGBUFFERS_SIZE,
 795                           chip->ring_base, chip->ring_base_dma);
 796
 797         /* release card data */
 798         kfree(chip);
 799         return 0;
 800 }
 801
 802 static int snd_sgio2audio_dev_free(struct snd_device *device)
 803 {
 804         struct snd_sgio2audio *chip = device->device_data;
 805
 806         return snd_sgio2audio_free(chip);
 807 }
 808
 809 static struct snd_device_ops ops = {
 810         .dev_free = snd_sgio2audio_dev_free,
 811 };
 812
 813 static int snd_sgio2audio_create(struct snd_card *card,
 814                                  struct snd_sgio2audio **rchip)
 815 {
 816         struct snd_sgio2audio *chip;
 817         int i, err;
 818
 819         *rchip = NULL;
 820
 821         /* check if a codec is attached to the interface */
 822         /* (Audio or Audio/Video board present) */
 823         if (!(readq(&mace->perif.audio.control) & AUDIO_CONTROL_CODEC_PRESENT))
 824                 return -ENOENT;
 825
 826         chip = kzalloc(sizeof(*chip), GFP_KERNEL);
 827         if (chip == NULL)
 828                 return -ENOMEM;
 829
 830         chip->card = card;
 831
 832         chip->ring_base = dma_alloc_coherent(card->dev,
 833                                              MACEISA_RINGBUFFERS_SIZE,
 834                                              &chip->ring_base_dma, GFP_KERNEL);
 835         if (chip->ring_base == NULL) {
 836                 printk(KERN_ERR
 837                        "sgio2audio: could not allocate ring buffers\n");
 838                 kfree(chip);
 839                 return -ENOMEM;
 840         }
 841
 842         spin_lock_init(&chip->ad1843_lock);
 843
 844         /* initialize channels */
 845         for (i = 0; i < 3; i++) {
 846                 spin_lock_init(&chip->channel[i].lock);
 847                 chip->channel[i].idx = i;
 848         }
 849
 850         /* allocate IRQs */
 851         for (i = 0; i < ARRAY_SIZE(snd_sgio2_isr_table); i++) {
 852                 if (request_irq(snd_sgio2_isr_table[i].irq,
 853                                 snd_sgio2_isr_table[i].isr,
 854                                 0,
 855                                 snd_sgio2_isr_table[i].desc,
 856                                 &chip->channel[snd_sgio2_isr_table[i].idx])) {
 857                         snd_sgio2audio_free(chip);
 858                         printk(KERN_ERR "sgio2audio: cannot allocate irq %d\n",
 859                                snd_sgio2_isr_table[i].irq);
 860                         return -EBUSY;
 861                 }
 862         }
 863
 864         /* reset the interface */
 865         writeq(AUDIO_CONTROL_RESET, &mace->perif.audio.control);
 866         udelay(1);
 867         writeq(0, &mace->perif.audio.control);
 868         msleep_interruptible(1); /* give time to recover */
 869
 870         /* set ring base */
 871         writeq(chip->ring_base_dma, &mace->perif.ctrl.ringbase);
 872
 873         /* attach the AD1843 codec */
 874         chip->ad1843.read = read_ad1843_reg;
 875         chip->ad1843.write = write_ad1843_reg;
 876         chip->ad1843.chip = chip;
 877
 878         /* initialize the AD1843 codec */
 879         err = ad1843_init(&chip->ad1843);
 880         if (err < 0) {
 881                 snd_sgio2audio_free(chip);
 882                 return err;
 883         }
 884
 885         err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops);
 886         if (err < 0) {
 887                 snd_sgio2audio_free(chip);
 888                 return err;
 889         }
 890         *rchip = chip;
 891         return 0;
 892 }
 893
 894 static int snd_sgio2audio_probe(struct platform_device *pdev)
 895 {
 896         struct snd_card *card;
 897         struct snd_sgio2audio *chip;
 898         int err;
 899
 900         err = snd_card_new(&pdev->dev, index, id, THIS_MODULE, 0, &card);
 901         if (err < 0)
 902                 return err;
 903
 904         err = snd_sgio2audio_create(card, &chip);
 905         if (err < 0) {
 906                 snd_card_free(card);
 907                 return err;
 908         }
 909
 910         err = snd_sgio2audio_new_pcm(chip);
 911         if (err < 0) {
 912                 snd_card_free(card);
 913                 return err;
 914         }
 915         err = snd_sgio2audio_new_mixer(chip);
 916         if (err < 0) {
 917                 snd_card_free(card);
 918                 return err;
 919         }
 920
 921         strcpy(card->driver, "SGI O2 Audio");
 922         strcpy(card->shortname, "SGI O2 Audio");
 923         sprintf(card->longname, "%s irq %i-%i",
 924                 card->shortname,
 925                 MACEISA_AUDIO1_DMAT_IRQ,
 926                 MACEISA_AUDIO3_MERR_IRQ);
 927
 928         err = snd_card_register(card);
 929         if (err < 0) {
 930                 snd_card_free(card);
 931                 return err;
 932         }
 933         platform_set_drvdata(pdev, card);
 934         return 0;
 935 }
 936
 937 static int snd_sgio2audio_remove(struct platform_device *pdev)
 938 {
 939         struct snd_card *card = platform_get_drvdata(pdev);
 940
 941         snd_card_free(card);
 942         return 0;
 943 }
 944
 945 static struct platform_driver sgio2audio_driver = {
 946         .probe  = snd_sgio2audio_probe,
 947         .remove = snd_sgio2audio_remove,
 948         .driver = {
 949                 .name   = "sgio2audio",
 950         }
 951 };
 952
 953 module_platform_driver(sgio2audio_driver);