xref: /openbmc/linux/sound/mips/sgio2audio.c (revision 9aa2cba7)
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 void snd_sgio2audio_remove(struct platform_device *pdev)
912 {
913 	struct snd_card *card = platform_get_drvdata(pdev);
914 
915 	snd_card_free(card);
916 }
917 
918 static struct platform_driver sgio2audio_driver = {
919 	.probe	= snd_sgio2audio_probe,
920 	.remove_new = snd_sgio2audio_remove,
921 	.driver = {
922 		.name	= "sgio2audio",
923 	}
924 };
925 
926 module_platform_driver(sgio2audio_driver);
927