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