ASoC: Revert "LOCAL / temporary workaround for i2s/prepare_lock deadlock"
[platform/kernel/linux-exynos.git] / 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/interrupt.h>
29 #include <linux/dma-mapping.h>
30 #include <linux/platform_device.h>
31 #include <linux/io.h>
32 #include <linux/slab.h>
33 #include <linux/module.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 * const texts[3] = {
205                 "Cam Mic", "Mic", "Line"
206         };
207         return snd_ctl_enum_info(uinfo, 1, 3, texts);
208 }
209
210 static int sgio2audio_source_get(struct snd_kcontrol *kcontrol,
211                                struct snd_ctl_elem_value *ucontrol)
212 {
213         struct snd_sgio2audio *chip = snd_kcontrol_chip(kcontrol);
214
215         ucontrol->value.enumerated.item[0] = ad1843_get_recsrc(&chip->ad1843);
216         return 0;
217 }
218
219 static int sgio2audio_source_put(struct snd_kcontrol *kcontrol,
220                         struct snd_ctl_elem_value *ucontrol)
221 {
222         struct snd_sgio2audio *chip = snd_kcontrol_chip(kcontrol);
223         int newsrc, oldsrc;
224
225         oldsrc = ad1843_get_recsrc(&chip->ad1843);
226         newsrc = ad1843_set_recsrc(&chip->ad1843,
227                                    ucontrol->value.enumerated.item[0]);
228
229         return newsrc != oldsrc;
230 }
231
232 /* dac1/pcm0 mixer control */
233 static const struct snd_kcontrol_new sgio2audio_ctrl_pcm0 = {
234         .iface          = SNDRV_CTL_ELEM_IFACE_MIXER,
235         .name           = "PCM Playback Volume",
236         .index          = 0,
237         .access         = SNDRV_CTL_ELEM_ACCESS_READWRITE,
238         .private_value  = AD1843_GAIN_PCM_0,
239         .info           = sgio2audio_gain_info,
240         .get            = sgio2audio_gain_get,
241         .put            = sgio2audio_gain_put,
242 };
243
244 /* dac2/pcm1 mixer control */
245 static const struct snd_kcontrol_new sgio2audio_ctrl_pcm1 = {
246         .iface          = SNDRV_CTL_ELEM_IFACE_MIXER,
247         .name           = "PCM Playback Volume",
248         .index          = 1,
249         .access         = SNDRV_CTL_ELEM_ACCESS_READWRITE,
250         .private_value  = AD1843_GAIN_PCM_1,
251         .info           = sgio2audio_gain_info,
252         .get            = sgio2audio_gain_get,
253         .put            = sgio2audio_gain_put,
254 };
255
256 /* record level mixer control */
257 static const struct snd_kcontrol_new sgio2audio_ctrl_reclevel = {
258         .iface          = SNDRV_CTL_ELEM_IFACE_MIXER,
259         .name           = "Capture Volume",
260         .access         = SNDRV_CTL_ELEM_ACCESS_READWRITE,
261         .private_value  = AD1843_GAIN_RECLEV,
262         .info           = sgio2audio_gain_info,
263         .get            = sgio2audio_gain_get,
264         .put            = sgio2audio_gain_put,
265 };
266
267 /* record level source control */
268 static const struct snd_kcontrol_new sgio2audio_ctrl_recsource = {
269         .iface          = SNDRV_CTL_ELEM_IFACE_MIXER,
270         .name           = "Capture Source",
271         .access         = SNDRV_CTL_ELEM_ACCESS_READWRITE,
272         .info           = sgio2audio_source_info,
273         .get            = sgio2audio_source_get,
274         .put            = sgio2audio_source_put,
275 };
276
277 /* line mixer control */
278 static const struct snd_kcontrol_new sgio2audio_ctrl_line = {
279         .iface          = SNDRV_CTL_ELEM_IFACE_MIXER,
280         .name           = "Line Playback Volume",
281         .index          = 0,
282         .access         = SNDRV_CTL_ELEM_ACCESS_READWRITE,
283         .private_value  = AD1843_GAIN_LINE,
284         .info           = sgio2audio_gain_info,
285         .get            = sgio2audio_gain_get,
286         .put            = sgio2audio_gain_put,
287 };
288
289 /* cd mixer control */
290 static const struct snd_kcontrol_new sgio2audio_ctrl_cd = {
291         .iface          = SNDRV_CTL_ELEM_IFACE_MIXER,
292         .name           = "Line Playback Volume",
293         .index          = 1,
294         .access         = SNDRV_CTL_ELEM_ACCESS_READWRITE,
295         .private_value  = AD1843_GAIN_LINE_2,
296         .info           = sgio2audio_gain_info,
297         .get            = sgio2audio_gain_get,
298         .put            = sgio2audio_gain_put,
299 };
300
301 /* mic mixer control */
302 static const struct snd_kcontrol_new sgio2audio_ctrl_mic = {
303         .iface          = SNDRV_CTL_ELEM_IFACE_MIXER,
304         .name           = "Mic Playback Volume",
305         .access         = SNDRV_CTL_ELEM_ACCESS_READWRITE,
306         .private_value  = AD1843_GAIN_MIC,
307         .info           = sgio2audio_gain_info,
308         .get            = sgio2audio_gain_get,
309         .put            = sgio2audio_gain_put,
310 };
311
312
313 static int snd_sgio2audio_new_mixer(struct snd_sgio2audio *chip)
314 {
315         int err;
316
317         err = snd_ctl_add(chip->card,
318                           snd_ctl_new1(&sgio2audio_ctrl_pcm0, chip));
319         if (err < 0)
320                 return err;
321
322         err = snd_ctl_add(chip->card,
323                           snd_ctl_new1(&sgio2audio_ctrl_pcm1, chip));
324         if (err < 0)
325                 return err;
326
327         err = snd_ctl_add(chip->card,
328                           snd_ctl_new1(&sgio2audio_ctrl_reclevel, chip));
329         if (err < 0)
330                 return err;
331
332         err = snd_ctl_add(chip->card,
333                           snd_ctl_new1(&sgio2audio_ctrl_recsource, chip));
334         if (err < 0)
335                 return err;
336         err = snd_ctl_add(chip->card,
337                           snd_ctl_new1(&sgio2audio_ctrl_line, chip));
338         if (err < 0)
339                 return err;
340
341         err = snd_ctl_add(chip->card,
342                           snd_ctl_new1(&sgio2audio_ctrl_cd, chip));
343         if (err < 0)
344                 return err;
345
346         err = snd_ctl_add(chip->card,
347                           snd_ctl_new1(&sgio2audio_ctrl_mic, chip));
348         if (err < 0)
349                 return err;
350
351         return 0;
352 }
353
354 /* low-level audio interface DMA */
355
356 /* get data out of bounce buffer, count must be a multiple of 32 */
357 /* returns 1 if a period has elapsed */
358 static int snd_sgio2audio_dma_pull_frag(struct snd_sgio2audio *chip,
359                                         unsigned int ch, unsigned int count)
360 {
361         int ret;
362         unsigned long src_base, src_pos, dst_mask;
363         unsigned char *dst_base;
364         int dst_pos;
365         u64 *src;
366         s16 *dst;
367         u64 x;
368         unsigned long flags;
369         struct snd_pcm_runtime *runtime = chip->channel[ch].substream->runtime;
370
371         spin_lock_irqsave(&chip->channel[ch].lock, flags);
372
373         src_base = (unsigned long) chip->ring_base | (ch << CHANNEL_RING_SHIFT);
374         src_pos = readq(&mace->perif.audio.chan[ch].read_ptr);
375         dst_base = runtime->dma_area;
376         dst_pos = chip->channel[ch].pos;
377         dst_mask = frames_to_bytes(runtime, runtime->buffer_size) - 1;
378
379         /* check if a period has elapsed */
380         chip->channel[ch].size += (count >> 3); /* in frames */
381         ret = chip->channel[ch].size >= runtime->period_size;
382         chip->channel[ch].size %= runtime->period_size;
383
384         while (count) {
385                 src = (u64 *)(src_base + src_pos);
386                 dst = (s16 *)(dst_base + dst_pos);
387
388                 x = *src;
389                 dst[0] = (x >> CHANNEL_LEFT_SHIFT) & 0xffff;
390                 dst[1] = (x >> CHANNEL_RIGHT_SHIFT) & 0xffff;
391
392                 src_pos = (src_pos + sizeof(u64)) & CHANNEL_RING_MASK;
393                 dst_pos = (dst_pos + 2 * sizeof(s16)) & dst_mask;
394                 count -= sizeof(u64);
395         }
396
397         writeq(src_pos, &mace->perif.audio.chan[ch].read_ptr); /* in bytes */
398         chip->channel[ch].pos = dst_pos;
399
400         spin_unlock_irqrestore(&chip->channel[ch].lock, flags);
401         return ret;
402 }
403
404 /* put some DMA data in bounce buffer, count must be a multiple of 32 */
405 /* returns 1 if a period has elapsed */
406 static int snd_sgio2audio_dma_push_frag(struct snd_sgio2audio *chip,
407                                         unsigned int ch, unsigned int count)
408 {
409         int ret;
410         s64 l, r;
411         unsigned long dst_base, dst_pos, src_mask;
412         unsigned char *src_base;
413         int src_pos;
414         u64 *dst;
415         s16 *src;
416         unsigned long flags;
417         struct snd_pcm_runtime *runtime = chip->channel[ch].substream->runtime;
418
419         spin_lock_irqsave(&chip->channel[ch].lock, flags);
420
421         dst_base = (unsigned long)chip->ring_base | (ch << CHANNEL_RING_SHIFT);
422         dst_pos = readq(&mace->perif.audio.chan[ch].write_ptr);
423         src_base = runtime->dma_area;
424         src_pos = chip->channel[ch].pos;
425         src_mask = frames_to_bytes(runtime, runtime->buffer_size) - 1;
426
427         /* check if a period has elapsed */
428         chip->channel[ch].size += (count >> 3); /* in frames */
429         ret = chip->channel[ch].size >= runtime->period_size;
430         chip->channel[ch].size %= runtime->period_size;
431
432         while (count) {
433                 src = (s16 *)(src_base + src_pos);
434                 dst = (u64 *)(dst_base + dst_pos);
435
436                 l = src[0]; /* sign extend */
437                 r = src[1]; /* sign extend */
438
439                 *dst = ((l & 0x00ffffff) << CHANNEL_LEFT_SHIFT) |
440                         ((r & 0x00ffffff) << CHANNEL_RIGHT_SHIFT);
441
442                 dst_pos = (dst_pos + sizeof(u64)) & CHANNEL_RING_MASK;
443                 src_pos = (src_pos + 2 * sizeof(s16)) & src_mask;
444                 count -= sizeof(u64);
445         }
446
447         writeq(dst_pos, &mace->perif.audio.chan[ch].write_ptr); /* in bytes */
448         chip->channel[ch].pos = src_pos;
449
450         spin_unlock_irqrestore(&chip->channel[ch].lock, flags);
451         return ret;
452 }
453
454 static int snd_sgio2audio_dma_start(struct snd_pcm_substream *substream)
455 {
456         struct snd_sgio2audio *chip = snd_pcm_substream_chip(substream);
457         struct snd_sgio2audio_chan *chan = substream->runtime->private_data;
458         int ch = chan->idx;
459
460         /* reset DMA channel */
461         writeq(CHANNEL_CONTROL_RESET, &mace->perif.audio.chan[ch].control);
462         udelay(10);
463         writeq(0, &mace->perif.audio.chan[ch].control);
464
465         if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
466                 /* push a full buffer */
467                 snd_sgio2audio_dma_push_frag(chip, ch, CHANNEL_RING_SIZE - 32);
468         }
469         /* set DMA to wake on 50% empty and enable interrupt */
470         writeq(CHANNEL_DMA_ENABLE | CHANNEL_INT_THRESHOLD_50,
471                &mace->perif.audio.chan[ch].control);
472         return 0;
473 }
474
475 static int snd_sgio2audio_dma_stop(struct snd_pcm_substream *substream)
476 {
477         struct snd_sgio2audio_chan *chan = substream->runtime->private_data;
478
479         writeq(0, &mace->perif.audio.chan[chan->idx].control);
480         return 0;
481 }
482
483 static irqreturn_t snd_sgio2audio_dma_in_isr(int irq, void *dev_id)
484 {
485         struct snd_sgio2audio_chan *chan = dev_id;
486         struct snd_pcm_substream *substream;
487         struct snd_sgio2audio *chip;
488         int count, ch;
489
490         substream = chan->substream;
491         chip = snd_pcm_substream_chip(substream);
492         ch = chan->idx;
493
494         /* empty the ring */
495         count = CHANNEL_RING_SIZE -
496                 readq(&mace->perif.audio.chan[ch].depth) - 32;
497         if (snd_sgio2audio_dma_pull_frag(chip, ch, count))
498                 snd_pcm_period_elapsed(substream);
499
500         return IRQ_HANDLED;
501 }
502
503 static irqreturn_t snd_sgio2audio_dma_out_isr(int irq, void *dev_id)
504 {
505         struct snd_sgio2audio_chan *chan = dev_id;
506         struct snd_pcm_substream *substream;
507         struct snd_sgio2audio *chip;
508         int count, ch;
509
510         substream = chan->substream;
511         chip = snd_pcm_substream_chip(substream);
512         ch = chan->idx;
513         /* fill the ring */
514         count = CHANNEL_RING_SIZE -
515                 readq(&mace->perif.audio.chan[ch].depth) - 32;
516         if (snd_sgio2audio_dma_push_frag(chip, ch, count))
517                 snd_pcm_period_elapsed(substream);
518
519         return IRQ_HANDLED;
520 }
521
522 static irqreturn_t snd_sgio2audio_error_isr(int irq, void *dev_id)
523 {
524         struct snd_sgio2audio_chan *chan = dev_id;
525         struct snd_pcm_substream *substream;
526
527         substream = chan->substream;
528         snd_sgio2audio_dma_stop(substream);
529         snd_sgio2audio_dma_start(substream);
530         return IRQ_HANDLED;
531 }
532
533 /* PCM part */
534 /* PCM hardware definition */
535 static const struct snd_pcm_hardware snd_sgio2audio_pcm_hw = {
536         .info = (SNDRV_PCM_INFO_MMAP |
537                  SNDRV_PCM_INFO_MMAP_VALID |
538                  SNDRV_PCM_INFO_INTERLEAVED |
539                  SNDRV_PCM_INFO_BLOCK_TRANSFER),
540         .formats =          SNDRV_PCM_FMTBIT_S16_BE,
541         .rates =            SNDRV_PCM_RATE_8000_48000,
542         .rate_min =         8000,
543         .rate_max =         48000,
544         .channels_min =     2,
545         .channels_max =     2,
546         .buffer_bytes_max = 65536,
547         .period_bytes_min = 32768,
548         .period_bytes_max = 65536,
549         .periods_min =      1,
550         .periods_max =      1024,
551 };
552
553 /* PCM playback open callback */
554 static int snd_sgio2audio_playback1_open(struct snd_pcm_substream *substream)
555 {
556         struct snd_sgio2audio *chip = snd_pcm_substream_chip(substream);
557         struct snd_pcm_runtime *runtime = substream->runtime;
558
559         runtime->hw = snd_sgio2audio_pcm_hw;
560         runtime->private_data = &chip->channel[1];
561         return 0;
562 }
563
564 static int snd_sgio2audio_playback2_open(struct snd_pcm_substream *substream)
565 {
566         struct snd_sgio2audio *chip = snd_pcm_substream_chip(substream);
567         struct snd_pcm_runtime *runtime = substream->runtime;
568
569         runtime->hw = snd_sgio2audio_pcm_hw;
570         runtime->private_data = &chip->channel[2];
571         return 0;
572 }
573
574 /* PCM capture open callback */
575 static int snd_sgio2audio_capture_open(struct snd_pcm_substream *substream)
576 {
577         struct snd_sgio2audio *chip = snd_pcm_substream_chip(substream);
578         struct snd_pcm_runtime *runtime = substream->runtime;
579
580         runtime->hw = snd_sgio2audio_pcm_hw;
581         runtime->private_data = &chip->channel[0];
582         return 0;
583 }
584
585 /* PCM close callback */
586 static int snd_sgio2audio_pcm_close(struct snd_pcm_substream *substream)
587 {
588         struct snd_pcm_runtime *runtime = substream->runtime;
589
590         runtime->private_data = NULL;
591         return 0;
592 }
593
594
595 /* hw_params callback */
596 static int snd_sgio2audio_pcm_hw_params(struct snd_pcm_substream *substream,
597                                         struct snd_pcm_hw_params *hw_params)
598 {
599         return snd_pcm_lib_alloc_vmalloc_buffer(substream,
600                                                 params_buffer_bytes(hw_params));
601 }
602
603 /* hw_free callback */
604 static int snd_sgio2audio_pcm_hw_free(struct snd_pcm_substream *substream)
605 {
606         return snd_pcm_lib_free_vmalloc_buffer(substream);
607 }
608
609 /* prepare callback */
610 static int snd_sgio2audio_pcm_prepare(struct snd_pcm_substream *substream)
611 {
612         struct snd_sgio2audio *chip = snd_pcm_substream_chip(substream);
613         struct snd_pcm_runtime *runtime = substream->runtime;
614         struct snd_sgio2audio_chan *chan = substream->runtime->private_data;
615         int ch = chan->idx;
616         unsigned long flags;
617
618         spin_lock_irqsave(&chip->channel[ch].lock, flags);
619
620         /* Setup the pseudo-dma transfer pointers.  */
621         chip->channel[ch].pos = 0;
622         chip->channel[ch].size = 0;
623         chip->channel[ch].substream = substream;
624
625         /* set AD1843 format */
626         /* hardware format is always S16_LE */
627         switch (substream->stream) {
628         case SNDRV_PCM_STREAM_PLAYBACK:
629                 ad1843_setup_dac(&chip->ad1843,
630                                  ch - 1,
631                                  runtime->rate,
632                                  SNDRV_PCM_FORMAT_S16_LE,
633                                  runtime->channels);
634                 break;
635         case SNDRV_PCM_STREAM_CAPTURE:
636                 ad1843_setup_adc(&chip->ad1843,
637                                  runtime->rate,
638                                  SNDRV_PCM_FORMAT_S16_LE,
639                                  runtime->channels);
640                 break;
641         }
642         spin_unlock_irqrestore(&chip->channel[ch].lock, flags);
643         return 0;
644 }
645
646 /* trigger callback */
647 static int snd_sgio2audio_pcm_trigger(struct snd_pcm_substream *substream,
648                                       int cmd)
649 {
650         switch (cmd) {
651         case SNDRV_PCM_TRIGGER_START:
652                 /* start the PCM engine */
653                 snd_sgio2audio_dma_start(substream);
654                 break;
655         case SNDRV_PCM_TRIGGER_STOP:
656                 /* stop the PCM engine */
657                 snd_sgio2audio_dma_stop(substream);
658                 break;
659         default:
660                 return -EINVAL;
661         }
662         return 0;
663 }
664
665 /* pointer callback */
666 static snd_pcm_uframes_t
667 snd_sgio2audio_pcm_pointer(struct snd_pcm_substream *substream)
668 {
669         struct snd_sgio2audio *chip = snd_pcm_substream_chip(substream);
670         struct snd_sgio2audio_chan *chan = substream->runtime->private_data;
671
672         /* get the current hardware pointer */
673         return bytes_to_frames(substream->runtime,
674                                chip->channel[chan->idx].pos);
675 }
676
677 /* operators */
678 static const struct snd_pcm_ops snd_sgio2audio_playback1_ops = {
679         .open =        snd_sgio2audio_playback1_open,
680         .close =       snd_sgio2audio_pcm_close,
681         .ioctl =       snd_pcm_lib_ioctl,
682         .hw_params =   snd_sgio2audio_pcm_hw_params,
683         .hw_free =     snd_sgio2audio_pcm_hw_free,
684         .prepare =     snd_sgio2audio_pcm_prepare,
685         .trigger =     snd_sgio2audio_pcm_trigger,
686         .pointer =     snd_sgio2audio_pcm_pointer,
687         .page =        snd_pcm_lib_get_vmalloc_page,
688         .mmap =        snd_pcm_lib_mmap_vmalloc,
689 };
690
691 static const struct snd_pcm_ops snd_sgio2audio_playback2_ops = {
692         .open =        snd_sgio2audio_playback2_open,
693         .close =       snd_sgio2audio_pcm_close,
694         .ioctl =       snd_pcm_lib_ioctl,
695         .hw_params =   snd_sgio2audio_pcm_hw_params,
696         .hw_free =     snd_sgio2audio_pcm_hw_free,
697         .prepare =     snd_sgio2audio_pcm_prepare,
698         .trigger =     snd_sgio2audio_pcm_trigger,
699         .pointer =     snd_sgio2audio_pcm_pointer,
700         .page =        snd_pcm_lib_get_vmalloc_page,
701         .mmap =        snd_pcm_lib_mmap_vmalloc,
702 };
703
704 static const struct snd_pcm_ops snd_sgio2audio_capture_ops = {
705         .open =        snd_sgio2audio_capture_open,
706         .close =       snd_sgio2audio_pcm_close,
707         .ioctl =       snd_pcm_lib_ioctl,
708         .hw_params =   snd_sgio2audio_pcm_hw_params,
709         .hw_free =     snd_sgio2audio_pcm_hw_free,
710         .prepare =     snd_sgio2audio_pcm_prepare,
711         .trigger =     snd_sgio2audio_pcm_trigger,
712         .pointer =     snd_sgio2audio_pcm_pointer,
713         .page =        snd_pcm_lib_get_vmalloc_page,
714         .mmap =        snd_pcm_lib_mmap_vmalloc,
715 };
716
717 /*
718  *  definitions of capture are omitted here...
719  */
720
721 /* create a pcm device */
722 static int snd_sgio2audio_new_pcm(struct snd_sgio2audio *chip)
723 {
724         struct snd_pcm *pcm;
725         int err;
726
727         /* create first pcm device with one outputs and one input */
728         err = snd_pcm_new(chip->card, "SGI O2 Audio", 0, 1, 1, &pcm);
729         if (err < 0)
730                 return err;
731
732         pcm->private_data = chip;
733         strcpy(pcm->name, "SGI O2 DAC1");
734
735         /* set operators */
736         snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK,
737                         &snd_sgio2audio_playback1_ops);
738         snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE,
739                         &snd_sgio2audio_capture_ops);
740
741         /* create second  pcm device with one outputs and no input */
742         err = snd_pcm_new(chip->card, "SGI O2 Audio", 1, 1, 0, &pcm);
743         if (err < 0)
744                 return err;
745
746         pcm->private_data = chip;
747         strcpy(pcm->name, "SGI O2 DAC2");
748
749         /* set operators */
750         snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK,
751                         &snd_sgio2audio_playback2_ops);
752
753         return 0;
754 }
755
756 static struct {
757         int idx;
758         int irq;
759         irqreturn_t (*isr)(int, void *);
760         const char *desc;
761 } snd_sgio2_isr_table[] = {
762         {
763                 .idx = 0,
764                 .irq = MACEISA_AUDIO1_DMAT_IRQ,
765                 .isr = snd_sgio2audio_dma_in_isr,
766                 .desc = "Capture DMA Channel 0"
767         }, {
768                 .idx = 0,
769                 .irq = MACEISA_AUDIO1_OF_IRQ,
770                 .isr = snd_sgio2audio_error_isr,
771                 .desc = "Capture Overflow"
772         }, {
773                 .idx = 1,
774                 .irq = MACEISA_AUDIO2_DMAT_IRQ,
775                 .isr = snd_sgio2audio_dma_out_isr,
776                 .desc = "Playback DMA Channel 1"
777         }, {
778                 .idx = 1,
779                 .irq = MACEISA_AUDIO2_MERR_IRQ,
780                 .isr = snd_sgio2audio_error_isr,
781                 .desc = "Memory Error Channel 1"
782         }, {
783                 .idx = 2,
784                 .irq = MACEISA_AUDIO3_DMAT_IRQ,
785                 .isr = snd_sgio2audio_dma_out_isr,
786                 .desc = "Playback DMA Channel 2"
787         }, {
788                 .idx = 2,
789                 .irq = MACEISA_AUDIO3_MERR_IRQ,
790                 .isr = snd_sgio2audio_error_isr,
791                 .desc = "Memory Error Channel 2"
792         }
793 };
794
795 /* ALSA driver */
796
797 static int snd_sgio2audio_free(struct snd_sgio2audio *chip)
798 {
799         int i;
800
801         /* reset interface */
802         writeq(AUDIO_CONTROL_RESET, &mace->perif.audio.control);
803         udelay(1);
804         writeq(0, &mace->perif.audio.control);
805
806         /* release IRQ's */
807         for (i = 0; i < ARRAY_SIZE(snd_sgio2_isr_table); i++)
808                 free_irq(snd_sgio2_isr_table[i].irq,
809                          &chip->channel[snd_sgio2_isr_table[i].idx]);
810
811         dma_free_coherent(NULL, MACEISA_RINGBUFFERS_SIZE,
812                           chip->ring_base, chip->ring_base_dma);
813
814         /* release card data */
815         kfree(chip);
816         return 0;
817 }
818
819 static int snd_sgio2audio_dev_free(struct snd_device *device)
820 {
821         struct snd_sgio2audio *chip = device->device_data;
822
823         return snd_sgio2audio_free(chip);
824 }
825
826 static struct snd_device_ops ops = {
827         .dev_free = snd_sgio2audio_dev_free,
828 };
829
830 static int snd_sgio2audio_create(struct snd_card *card,
831                                  struct snd_sgio2audio **rchip)
832 {
833         struct snd_sgio2audio *chip;
834         int i, err;
835
836         *rchip = NULL;
837
838         /* check if a codec is attached to the interface */
839         /* (Audio or Audio/Video board present) */
840         if (!(readq(&mace->perif.audio.control) & AUDIO_CONTROL_CODEC_PRESENT))
841                 return -ENOENT;
842
843         chip = kzalloc(sizeof(struct snd_sgio2audio), GFP_KERNEL);
844         if (chip == NULL)
845                 return -ENOMEM;
846
847         chip->card = card;
848
849         chip->ring_base = dma_alloc_coherent(NULL, MACEISA_RINGBUFFERS_SIZE,
850                                              &chip->ring_base_dma, GFP_USER);
851         if (chip->ring_base == NULL) {
852                 printk(KERN_ERR
853                        "sgio2audio: could not allocate ring buffers\n");
854                 kfree(chip);
855                 return -ENOMEM;
856         }
857
858         spin_lock_init(&chip->ad1843_lock);
859
860         /* initialize channels */
861         for (i = 0; i < 3; i++) {
862                 spin_lock_init(&chip->channel[i].lock);
863                 chip->channel[i].idx = i;
864         }
865
866         /* allocate IRQs */
867         for (i = 0; i < ARRAY_SIZE(snd_sgio2_isr_table); i++) {
868                 if (request_irq(snd_sgio2_isr_table[i].irq,
869                                 snd_sgio2_isr_table[i].isr,
870                                 0,
871                                 snd_sgio2_isr_table[i].desc,
872                                 &chip->channel[snd_sgio2_isr_table[i].idx])) {
873                         snd_sgio2audio_free(chip);
874                         printk(KERN_ERR "sgio2audio: cannot allocate irq %d\n",
875                                snd_sgio2_isr_table[i].irq);
876                         return -EBUSY;
877                 }
878         }
879
880         /* reset the interface */
881         writeq(AUDIO_CONTROL_RESET, &mace->perif.audio.control);
882         udelay(1);
883         writeq(0, &mace->perif.audio.control);
884         msleep_interruptible(1); /* give time to recover */
885
886         /* set ring base */
887         writeq(chip->ring_base_dma, &mace->perif.ctrl.ringbase);
888
889         /* attach the AD1843 codec */
890         chip->ad1843.read = read_ad1843_reg;
891         chip->ad1843.write = write_ad1843_reg;
892         chip->ad1843.chip = chip;
893
894         /* initialize the AD1843 codec */
895         err = ad1843_init(&chip->ad1843);
896         if (err < 0) {
897                 snd_sgio2audio_free(chip);
898                 return err;
899         }
900
901         err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops);
902         if (err < 0) {
903                 snd_sgio2audio_free(chip);
904                 return err;
905         }
906         *rchip = chip;
907         return 0;
908 }
909
910 static int snd_sgio2audio_probe(struct platform_device *pdev)
911 {
912         struct snd_card *card;
913         struct snd_sgio2audio *chip;
914         int err;
915
916         err = snd_card_new(&pdev->dev, index, id, THIS_MODULE, 0, &card);
917         if (err < 0)
918                 return err;
919
920         err = snd_sgio2audio_create(card, &chip);
921         if (err < 0) {
922                 snd_card_free(card);
923                 return err;
924         }
925
926         err = snd_sgio2audio_new_pcm(chip);
927         if (err < 0) {
928                 snd_card_free(card);
929                 return err;
930         }
931         err = snd_sgio2audio_new_mixer(chip);
932         if (err < 0) {
933                 snd_card_free(card);
934                 return err;
935         }
936
937         strcpy(card->driver, "SGI O2 Audio");
938         strcpy(card->shortname, "SGI O2 Audio");
939         sprintf(card->longname, "%s irq %i-%i",
940                 card->shortname,
941                 MACEISA_AUDIO1_DMAT_IRQ,
942                 MACEISA_AUDIO3_MERR_IRQ);
943
944         err = snd_card_register(card);
945         if (err < 0) {
946                 snd_card_free(card);
947                 return err;
948         }
949         platform_set_drvdata(pdev, card);
950         return 0;
951 }
952
953 static int snd_sgio2audio_remove(struct platform_device *pdev)
954 {
955         struct snd_card *card = platform_get_drvdata(pdev);
956
957         snd_card_free(card);
958         return 0;
959 }
960
961 static struct platform_driver sgio2audio_driver = {
962         .probe  = snd_sgio2audio_probe,
963         .remove = snd_sgio2audio_remove,
964         .driver = {
965                 .name   = "sgio2audio",
966         }
967 };
968
969 module_platform_driver(sgio2audio_driver);