aboutsummaryrefslogtreecommitdiff
path: root/sound/mips/sgio2audio.c
blob: 717604c00f0add6bcab218e9b17e4de9dc6efcf7 (plain)
1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
127
128
129
130
131
132
133
134
135
136
137
138
139
140
141
142
143
144
145
146
147
148
149
150
151
152
153
154
155
156
157
158
159
160
161
162
163
164
165
166
167
168
169
170
171
172
173
174
175
176
177
178
179
180
181
182
183
184
185
186
187
188
189
190
191
192
193
194
195
196
197
198
199
200
201
202
203
204
205
206
207
208
209
210
211
212
213
214
215
216
217
218
219
220
221
222
223
224
225
226
227
228
229
230
231
232
233
234
235
236
237
238
239
240
241
242
243
244
245
246
247
248
249
250
251
252
253
254
255
256
257
258
259
260
261
262
263
264
265
266
267
268
269
270
271
272
273
274
275
276
277
278
279
280
281
282
283
284
285
286
287
288
289
290
291
292
293
294
295
296
297
298
299
300
301
302
303
304
305
306
307
308
309
310
311
312
313
314
315
316
317
318
319
320
321
322
323
324
325
326
327
328
329
330
331
332
333
334
335
336
337
338
339
340
341
342
343
344
345
346
347
348
349
350
351
352
353
354
355
356
357
358
359
360
361
362
363
364
365
366
367
368
369
370
371
372
373
374
375
376
377
378
379
380
381
382
383
384
385
386
387
388
389
390
391
392
393
394
395
396
397
398
399
400
401
402
403
404
405
406
407
408
409
410
411
412
413
414
415
416
417
418
419
420
421
422
423
424
425
426
427
428
429
430
431
432
433
434
435
436
437
438
439
440
441
442
443
444
445
446
447
448
449
450
451
452
453
454
455
456
457
458
459
460
461
462
463
464
465
466
467
468
469
470
471
472
473
474
475
476
477
478
479
480
481
482
483
484
485
486
487
488
489
490
491
492
493
494
495
496
497
498
499
500
501
502
503
504
505
506
507
508
509
510
511
512
513
514
515
516
517
518
519
520
521
522
523
524
525
526
527
528
529
530
531
532
533
534
535
536
537
538
539
540
541
542
543
544
545
546
547
548
549
550
551
552
553
554
555
556
557
558
559
560
561
562
563
564
565
566
567
568
569
570
571
572
573
574
575
576
577
578
579
580
581
582
583
584
585
586
587
588
589
590
591
592
593
594
595
596
597
598
599
600
601
602
603
604
605
606
607
608
609
610
611
612
613
614
615
616
617
618
619
620
621
622
623
624
625
626
627
628
629
630
631
632
633
634
635
636
637
638
639
640
641
642
643
644
645
646
647
648
649
650
651
652
653
654
655
656
657
658
659
660
661
662
663
664
665
666
667
668
669
670
671
672
673
674
675
676
677
678
679
680
681
682
683
684
685
686
687
688
689
690
691
692
693
694
695
696
697
698
699
700
701
702
703
704
705
706
707
708
709
710
711
712
713
714
715
716
717
718
719
720
721
722
723
724
725
726
727
728
729
730
731
732
733
734
735
736
737
738
739
740
741
742
743
744
745
746
747
748
749
750
751
752
753
754
755
756
757
758
759
760
761
762
763
764
765
766
767
768
769
770
771
772
773
774
775
776
777
778
779
780
781
782
783
784
785
786
787
788
789
790
791
792
793
794
795
796
797
798
799
800
801
802
803
804
805
806
807
808
809
810
811
812
813
814
815
816
817
818
819
820
821
822
823
824
825
826
827
828
829
830
831
832
833
834
835
836
837
838
839
840
841
842
843
844
845
846
847
848
849
850
851
852
853
854
855
856
857
858
859
860
861
862
863
864
865
866
867
868
869
870
871
872
873
874
875
876
877
878
879
880
881
882
883
884
885
886
887
888
889
890
891
892
893
894
895
896
897
898
899
900
901
902
903
904
905
906
907
908
909
910
911
912
913
914
915
916
917
918
919
920
921
922
923
924
925
926
927
928
929
930
931
932
933
934
935
936
937
938
939
940
941
942
943
944
945
946
947
948
949
950
951
952
953
954
955
956
957
958
959
960
961
962
963
964
965
966
967
968
969
970
971
972
973
974
975
976
977
978
979
980
981
982
983
984
985
986
987
988
989
/*
 *   Sound driver for Silicon Graphics O2 Workstations A/V board audio.
 *
 *   Copyright 2003 Vivien Chappelier <vivien.chappelier@linux-mips.org>
 *   Copyright 2008 Thomas Bogendoerfer <tsbogend@alpha.franken.de>
 *   Mxier part taken from mace_audio.c:
 *   Copyright 2007 Thorben Jändling <tj.trevelyan@gmail.com>
 *
 *   This program is free software; you can redistribute it and/or modify
 *   it under the terms of the GNU General Public License as published by
 *   the Free Software Foundation; either version 2 of the License, or
 *   (at your option) any later version.
 *
 *   This program is distributed in the hope that it will be useful,
 *   but WITHOUT ANY WARRANTY; without even the implied warranty of
 *   MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
 *   GNU General Public License for more details.
 *
 *   You should have received a copy of the GNU General Public License
 *   along with this program; if not, write to the Free Software
 *   Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307 USA
 *
 */

#include <linux/init.h>
#include <linux/delay.h>
#include <linux/spinlock.h>
#include <linux/interrupt.h>
#include <linux/dma-mapping.h>
#include <linux/platform_device.h>
#include <linux/io.h>
#include <linux/slab.h>

#include <asm/ip32/ip32_ints.h>
#include <asm/ip32/mace.h>

#include <sound/core.h>
#include <sound/control.h>
#include <sound/pcm.h>
#define SNDRV_GET_ID
#include <sound/initval.h>
#include <sound/ad1843.h>


MODULE_AUTHOR("Vivien Chappelier <vivien.chappelier@linux-mips.org>");
MODULE_DESCRIPTION("SGI O2 Audio");
MODULE_LICENSE("GPL");
MODULE_SUPPORTED_DEVICE("{{Silicon Graphics, O2 Audio}}");

static int index = SNDRV_DEFAULT_IDX1;  /* Index 0-MAX */
static char *id = SNDRV_DEFAULT_STR1;   /* ID for this card */

module_param(index, int, 0444);
MODULE_PARM_DESC(index, "Index value for SGI O2 soundcard.");
module_param(id, charp, 0444);
MODULE_PARM_DESC(id, "ID string for SGI O2 soundcard.");


#define AUDIO_CONTROL_RESET              BIT(0) /* 1: reset audio interface */
#define AUDIO_CONTROL_CODEC_PRESENT      BIT(1) /* 1: codec detected */

#define CODEC_CONTROL_WORD_SHIFT        0
#define CODEC_CONTROL_READ              BIT(16)
#define CODEC_CONTROL_ADDRESS_SHIFT     17

#define CHANNEL_CONTROL_RESET           BIT(10) /* 1: reset channel */
#define CHANNEL_DMA_ENABLE              BIT(9)  /* 1: enable DMA transfer */
#define CHANNEL_INT_THRESHOLD_DISABLED  (0 << 5) /* interrupt disabled */
#define CHANNEL_INT_THRESHOLD_25        (1 << 5) /* int on buffer >25% full */
#define CHANNEL_INT_THRESHOLD_50        (2 << 5) /* int on buffer >50% full */
#define CHANNEL_INT_THRESHOLD_75        (3 << 5) /* int on buffer >75% full */
#define CHANNEL_INT_THRESHOLD_EMPTY     (4 << 5) /* int on buffer empty */
#define CHANNEL_INT_THRESHOLD_NOT_EMPTY (5 << 5) /* int on buffer !empty */
#define CHANNEL_INT_THRESHOLD_FULL      (6 << 5) /* int on buffer empty */
#define CHANNEL_INT_THRESHOLD_NOT_FULL  (7 << 5) /* int on buffer !empty */

#define CHANNEL_RING_SHIFT              12
#define CHANNEL_RING_SIZE               (1 << CHANNEL_RING_SHIFT)
#define CHANNEL_RING_MASK               (CHANNEL_RING_SIZE - 1)

#define CHANNEL_LEFT_SHIFT 40
#define CHANNEL_RIGHT_SHIFT 8

struct snd_sgio2audio_chan {
	int idx;
	struct snd_pcm_substream *substream;
	int pos;
	snd_pcm_uframes_t size;
	spinlock_t lock;
};

/* definition of the chip-specific record */
struct snd_sgio2audio {
	struct snd_card *card;

	/* codec */
	struct snd_ad1843 ad1843;
	spinlock_t ad1843_lock;

	/* channels */
	struct snd_sgio2audio_chan channel[3];

	/* resources */
	void *ring_base;
	dma_addr_t ring_base_dma;
};

/* AD1843 access */

/*
 * read_ad1843_reg returns the current contents of a 16 bit AD1843 register.
 *
 * Returns unsigned register value on success, -errno on failure.
 */
static int read_ad1843_reg(void *priv, int reg)
{
	struct snd_sgio2audio *chip = priv;
	int val;
	unsigned long flags;

	spin_lock_irqsave(&chip->ad1843_lock, flags);

	writeq((reg << CODEC_CONTROL_ADDRESS_SHIFT) |
	       CODEC_CONTROL_READ, &mace->perif.audio.codec_control);
	wmb();
	val = readq(&mace->perif.audio.codec_control); /* flush bus */
	udelay(200);

	val = readq(&mace->perif.audio.codec_read);

	spin_unlock_irqrestore(&chip->ad1843_lock, flags);
	return val;
}

/*
 * write_ad1843_reg writes the specified value to a 16 bit AD1843 register.
 */
static int write_ad1843_reg(void *priv, int reg, int word)
{
	struct snd_sgio2audio *chip = priv;
	int val;
	unsigned long flags;

	spin_lock_irqsave(&chip->ad1843_lock, flags);

	writeq((reg << CODEC_CONTROL_ADDRESS_SHIFT) |
	       (word << CODEC_CONTROL_WORD_SHIFT),
	       &mace->perif.audio.codec_control);
	wmb();
	val = readq(&mace->perif.audio.codec_control); /* flush bus */
	udelay(200);

	spin_unlock_irqrestore(&chip->ad1843_lock, flags);
	return 0;
}

static int sgio2audio_gain_info(struct snd_kcontrol *kcontrol,
			       struct snd_ctl_elem_info *uinfo)
{
	struct snd_sgio2audio *chip = snd_kcontrol_chip(kcontrol);

	uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
	uinfo->count = 2;
	uinfo->value.integer.min = 0;
	uinfo->value.integer.max = ad1843_get_gain_max(&chip->ad1843,
					     (int)kcontrol->private_value);
	return 0;
}

static int sgio2audio_gain_get(struct snd_kcontrol *kcontrol,
			       struct snd_ctl_elem_value *ucontrol)
{
	struct snd_sgio2audio *chip = snd_kcontrol_chip(kcontrol);
	int vol;

	vol = ad1843_get_gain(&chip->ad1843, (int)kcontrol->private_value);

	ucontrol->value.integer.value[0] = (vol >> 8) & 0xFF;
	ucontrol->value.integer.value[1] = vol & 0xFF;

	return 0;
}

static int sgio2audio_gain_put(struct snd_kcontrol *kcontrol,
			struct snd_ctl_elem_value *ucontrol)
{
	struct snd_sgio2audio *chip = snd_kcontrol_chip(kcontrol);
	int newvol, oldvol;

	oldvol = ad1843_get_gain(&chip->ad1843, kcontrol->private_value);
	newvol = (ucontrol->value.integer.value[0] << 8) |
		ucontrol->value.integer.value[1];

	newvol = ad1843_set_gain(&chip->ad1843, kcontrol->private_value,
		newvol);

	return newvol != oldvol;
}

static int sgio2audio_source_info(struct snd_kcontrol *kcontrol,
			       struct snd_ctl_elem_info *uinfo)
{
	static const char *texts[3] = {
		"Cam Mic", "Mic", "Line"
	};
	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
	uinfo->count = 1;
	uinfo->value.enumerated.items = 3;
	if (uinfo->value.enumerated.item >= 3)
		uinfo->value.enumerated.item = 1;
	strcpy(uinfo->value.enumerated.name,
	       texts[uinfo->value.enumerated.item]);
	return 0;
}

static int sgio2audio_source_get(struct snd_kcontrol *kcontrol,
			       struct snd_ctl_elem_value *ucontrol)
{
	struct snd_sgio2audio *chip = snd_kcontrol_chip(kcontrol);

	ucontrol->value.enumerated.item[0] = ad1843_get_recsrc(&chip->ad1843);
	return 0;
}

static int sgio2audio_source_put(struct snd_kcontrol *kcontrol,
			struct snd_ctl_elem_value *ucontrol)
{
	struct snd_sgio2audio *chip = snd_kcontrol_chip(kcontrol);
	int newsrc, oldsrc;

	oldsrc = ad1843_get_recsrc(&chip->ad1843);
	newsrc = ad1843_set_recsrc(&chip->ad1843,
				   ucontrol->value.enumerated.item[0]);

	return newsrc != oldsrc;
}

/* dac1/pcm0 mixer control */
static struct snd_kcontrol_new sgio2audio_ctrl_pcm0 __devinitdata = {
	.iface          = SNDRV_CTL_ELEM_IFACE_MIXER,
	.name           = "PCM Playback Volume",
	.index          = 0,
	.access         = SNDRV_CTL_ELEM_ACCESS_READWRITE,
	.private_value  = AD1843_GAIN_PCM_0,
	.info           = sgio2audio_gain_info,
	.get            = sgio2audio_gain_get,
	.put            = sgio2audio_gain_put,
};

/* dac2/pcm1 mixer control */
static struct snd_kcontrol_new sgio2audio_ctrl_pcm1 __devinitdata = {
	.iface          = SNDRV_CTL_ELEM_IFACE_MIXER,
	.name           = "PCM Playback Volume",
	.index          = 1,
	.access         = SNDRV_CTL_ELEM_ACCESS_READWRITE,
	.private_value  = AD1843_GAIN_PCM_1,
	.info           = sgio2audio_gain_info,
	.get            = sgio2audio_gain_get,
	.put            = sgio2audio_gain_put,
};

/* record level mixer control */
static struct snd_kcontrol_new sgio2audio_ctrl_reclevel __devinitdata = {
	.iface          = SNDRV_CTL_ELEM_IFACE_MIXER,
	.name           = "Capture Volume",
	.access         = SNDRV_CTL_ELEM_ACCESS_READWRITE,
	.private_value  = AD1843_GAIN_RECLEV,
	.info           = sgio2audio_gain_info,
	.get            = sgio2audio_gain_get,
	.put            = sgio2audio_gain_put,
};

/* record level source control */
static struct snd_kcontrol_new sgio2audio_ctrl_recsource __devinitdata = {
	.iface          = SNDRV_CTL_ELEM_IFACE_MIXER,
	.name           = "Capture Source",
	.access         = SNDRV_CTL_ELEM_ACCESS_READWRITE,
	.info           = sgio2audio_source_info,
	.get            = sgio2audio_source_get,
	.put            = sgio2audio_source_put,
};

/* line mixer control */
static struct snd_kcontrol_new sgio2audio_ctrl_line __devinitdata = {
	.iface          = SNDRV_CTL_ELEM_IFACE_MIXER,
	.name           = "Line Playback Volume",
	.index          = 0,
	.access         = SNDRV_CTL_ELEM_ACCESS_READWRITE,
	.private_value  = AD1843_GAIN_LINE,
	.info           = sgio2audio_gain_info,
	.get            = sgio2audio_gain_get,
	.put            = sgio2audio_gain_put,
};

/* cd mixer control */
static struct snd_kcontrol_new sgio2audio_ctrl_cd __devinitdata = {
	.iface          = SNDRV_CTL_ELEM_IFACE_MIXER,
	.name           = "Line Playback Volume",
	.index          = 1,
	.access         = SNDRV_CTL_ELEM_ACCESS_READWRITE,
	.private_value  = AD1843_GAIN_LINE_2,
	.info           = sgio2audio_gain_info,
	.get            = sgio2audio_gain_get,
	.put            = sgio2audio_gain_put,
};

/* mic mixer control */
static struct snd_kcontrol_new sgio2audio_ctrl_mic __devinitdata = {
	.iface          = SNDRV_CTL_ELEM_IFACE_MIXER,
	.name           = "Mic Playback Volume",
	.access         = SNDRV_CTL_ELEM_ACCESS_READWRITE,
	.private_value  = AD1843_GAIN_MIC,
	.info           = sgio2audio_gain_info,
	.get            = sgio2audio_gain_get,
	.put            = sgio2audio_gain_put,
};


static int __devinit snd_sgio2audio_new_mixer(struct snd_sgio2audio *chip)
{
	int err;

	err = snd_ctl_add(chip->card,
			  snd_ctl_new1(&sgio2audio_ctrl_pcm0, chip));
	if (err < 0)
		return err;

	err = snd_ctl_add(chip->card,
			  snd_ctl_new1(&sgio2audio_ctrl_pcm1, chip));
	if (err < 0)
		return err;

	err = snd_ctl_add(chip->card,
			  snd_ctl_new1(&sgio2audio_ctrl_reclevel, chip));
	if (err < 0)
		return err;

	err = snd_ctl_add(chip->card,
			  snd_ctl_new1(&sgio2audio_ctrl_recsource, chip));
	if (err < 0)
		return err;
	err = snd_ctl_add(chip->card,
			  snd_ctl_new1(&sgio2audio_ctrl_line, chip));
	if (err < 0)
		return err;

	err = snd_ctl_add(chip->card,
			  snd_ctl_new1(&sgio2audio_ctrl_cd, chip));
	if (err < 0)
		return err;

	err = snd_ctl_add(chip->card,
			  snd_ctl_new1(&sgio2audio_ctrl_mic, chip));
	if (err < 0)
		return err;

	return 0;
}

/* low-level audio interface DMA */

/* get data out of bounce buffer, count must be a multiple of 32 */
/* returns 1 if a period has elapsed */
static int snd_sgio2audio_dma_pull_frag(struct snd_sgio2audio *chip,
					unsigned int ch, unsigned int count)
{
	int ret;
	unsigned long src_base, src_pos, dst_mask;
	unsigned char *dst_base;
	int dst_pos;
	u64 *src;
	s16 *dst;
	u64 x;
	unsigned long flags;
	struct snd_pcm_runtime *runtime = chip->channel[ch].substream->runtime;

	spin_lock_irqsave(&chip->channel[ch].lock, flags);

	src_base = (unsigned long) chip->ring_base | (ch << CHANNEL_RING_SHIFT);
	src_pos = readq(&mace->perif.audio.chan[ch].read_ptr);
	dst_base = runtime->dma_area;
	dst_pos = chip->channel[ch].pos;
	dst_mask = frames_to_bytes(runtime, runtime->buffer_size) - 1;

	/* check if a period has elapsed */
	chip->channel[ch].size += (count >> 3); /* in frames */
	ret = chip->channel[ch].size >= runtime->period_size;
	chip->channel[ch].size %= runtime->period_size;

	while (count) {
		src = (u64 *)(src_base + src_pos);
		dst = (s16 *)(dst_base + dst_pos);

		x = *src;
		dst[0] = (x >> CHANNEL_LEFT_SHIFT) & 0xffff;
		dst[1] = (x >> CHANNEL_RIGHT_SHIFT) & 0xffff;

		src_pos = (src_pos + sizeof(u64)) & CHANNEL_RING_MASK;
		dst_pos = (dst_pos + 2 * sizeof(s16)) & dst_mask;
		count -= sizeof(u64);
	}

	writeq(src_pos, &mace->perif.audio.chan[ch].read_ptr); /* in bytes */
	chip->channel[ch].pos = dst_pos;

	spin_unlock_irqrestore(&chip->channel[ch].lock, flags);
	return ret;
}

/* put some DMA data in bounce buffer, count must be a multiple of 32 */
/* returns 1 if a period has elapsed */
static int snd_sgio2audio_dma_push_frag(struct snd_sgio2audio *chip,
					unsigned int ch, unsigned int count)
{
	int ret;
	s64 l, r;
	unsigned long dst_base, dst_pos, src_mask;
	unsigned char *src_base;
	int src_pos;
	u64 *dst;
	s16 *src;
	unsigned long flags;
	struct snd_pcm_runtime *runtime = chip->channel[ch].substream->runtime;

	spin_lock_irqsave(&chip->channel[ch].lock, flags);

	dst_base = (unsigned long)chip->ring_base | (ch << CHANNEL_RING_SHIFT);
	dst_pos = readq(&mace->perif.audio.chan[ch].write_ptr);
	src_base = runtime->dma_area;
	src_pos = chip->channel[ch].pos;
	src_mask = frames_to_bytes(runtime, runtime->buffer_size) - 1;

	/* check if a period has elapsed */
	chip->channel[ch].size += (count >> 3); /* in frames */
	ret = chip->channel[ch].size >= runtime->period_size;
	chip->channel[ch].size %= runtime->period_size;

	while (count) {
		src = (s16 *)(src_base + src_pos);
		dst = (u64 *)(dst_base + dst_pos);

		l = src[0]; /* sign extend */
		r = src[1]; /* sign extend */

		*dst = ((l & 0x00ffffff) << CHANNEL_LEFT_SHIFT) |
			((r & 0x00ffffff) << CHANNEL_RIGHT_SHIFT);

		dst_pos = (dst_pos + sizeof(u64)) & CHANNEL_RING_MASK;
		src_pos = (src_pos + 2 * sizeof(s16)) & src_mask;
		count -= sizeof(u64);
	}

	writeq(dst_pos, &mace->perif.audio.chan[ch].write_ptr); /* in bytes */
	chip->channel[ch].pos = src_pos;

	spin_unlock_irqrestore(&chip->channel[ch].lock, flags);
	return ret;
}

static int snd_sgio2audio_dma_start(struct snd_pcm_substream *substream)
{
	struct snd_sgio2audio *chip = snd_pcm_substream_chip(substream);
	struct snd_sgio2audio_chan *chan = substream->runtime->private_data;
	int ch = chan->idx;

	/* reset DMA channel */
	writeq(CHANNEL_CONTROL_RESET, &mace->perif.audio.chan[ch].control);
	udelay(10);
	writeq(0, &mace->perif.audio.chan[ch].control);

	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
		/* push a full buffer */
		snd_sgio2audio_dma_push_frag(chip, ch, CHANNEL_RING_SIZE - 32);
	}
	/* set DMA to wake on 50% empty and enable interrupt */
	writeq(CHANNEL_DMA_ENABLE | CHANNEL_INT_THRESHOLD_50,
	       &mace->perif.audio.chan[ch].control);
	return 0;
}

static int snd_sgio2audio_dma_stop(struct snd_pcm_substream *substream)
{
	struct snd_sgio2audio_chan *chan = substream->runtime->private_data;

	writeq(0, &mace->perif.audio.chan[chan->idx].control);
	return 0;
}

static irqreturn_t snd_sgio2audio_dma_in_isr(int irq, void *dev_id)
{
	struct snd_sgio2audio_chan *chan = dev_id;
	struct snd_pcm_substream *substream;
	struct snd_sgio2audio *chip;
	int count, ch;

	substream = chan->substream;
	chip = snd_pcm_substream_chip(substream);
	ch = chan->idx;

	/* empty the ring */
	count = CHANNEL_RING_SIZE -
		readq(&mace->perif.audio.chan[ch].depth) - 32;
	if (snd_sgio2audio_dma_pull_frag(chip, ch, count))
		snd_pcm_period_elapsed(substream);

	return IRQ_HANDLED;
}

static irqreturn_t snd_sgio2audio_dma_out_isr(int irq, void *dev_id)
{
	struct snd_sgio2audio_chan *chan = dev_id;
	struct snd_pcm_substream *substream;
	struct snd_sgio2audio *chip;
	int count, ch;

	substream = chan->substream;
	chip = snd_pcm_substream_chip(substream);
	ch = chan->idx;
	/* fill the ring */
	count = CHANNEL_RING_SIZE -
		readq(&mace->perif.audio.chan[ch].depth) - 32;
	if (snd_sgio2audio_dma_push_frag(chip, ch, count))
		snd_pcm_period_elapsed(substream);

	return IRQ_HANDLED;
}

static irqreturn_t snd_sgio2audio_error_isr(int irq, void *dev_id)
{
	struct snd_sgio2audio_chan *chan = dev_id;
	struct snd_pcm_substream *substream;

	substream = chan->substream;
	snd_sgio2audio_dma_stop(substream);
	snd_sgio2audio_dma_start(substream);
	return IRQ_HANDLED;
}

/* PCM part */
/* PCM hardware definition */
static struct snd_pcm_hardware snd_sgio2audio_pcm_hw = {
	.info = (SNDRV_PCM_INFO_MMAP |
		 SNDRV_PCM_INFO_MMAP_VALID |
		 SNDRV_PCM_INFO_INTERLEAVED |
		 SNDRV_PCM_INFO_BLOCK_TRANSFER),
	.formats =          SNDRV_PCM_FMTBIT_S16_BE,
	.rates =            SNDRV_PCM_RATE_8000_48000,
	.rate_min =         8000,
	.rate_max =         48000,
	.channels_min =     2,
	.channels_max =     2,
	.buffer_bytes_max = 65536,
	.period_bytes_min = 32768,
	.period_bytes_max = 65536,
	.periods_min =      1,
	.periods_max =      1024,
};

/* PCM playback open callback */
static int snd_sgio2audio_playback1_open(struct snd_pcm_substream *substream)
{
	struct snd_sgio2audio *chip = snd_pcm_substream_chip(substream);
	struct snd_pcm_runtime *runtime = substream->runtime;

	runtime->hw = snd_sgio2audio_pcm_hw;
	runtime->private_data = &chip->channel[1];
	return 0;
}

static int snd_sgio2audio_playback2_open(struct snd_pcm_substream *substream)
{
	struct snd_sgio2audio *chip = snd_pcm_substream_chip(substream);
	struct snd_pcm_runtime *runtime = substream->runtime;

	runtime->hw = snd_sgio2audio_pcm_hw;
	runtime->private_data = &chip->channel[2];
	return 0;
}

/* PCM capture open callback */
static int snd_sgio2audio_capture_open(struct snd_pcm_substream *substream)
{
	struct snd_sgio2audio *chip = snd_pcm_substream_chip(substream);
	struct snd_pcm_runtime *runtime = substream->runtime;

	runtime->hw = snd_sgio2audio_pcm_hw;
	runtime->private_data = &chip->channel[0];
	return 0;
}

/* PCM close callback */
static int snd_sgio2audio_pcm_close(struct snd_pcm_substream *substream)
{
	struct snd_pcm_runtime *runtime = substream->runtime;

	runtime->private_data = NULL;
	return 0;
}


/* hw_params callback */
static int snd_sgio2audio_pcm_hw_params(struct snd_pcm_substream *substream,
					struct snd_pcm_hw_params *hw_params)
{
	return snd_pcm_lib_alloc_vmalloc_buffer(substream,
						params_buffer_bytes(hw_params));
}

/* hw_free callback */
static int snd_sgio2audio_pcm_hw_free(struct snd_pcm_substream *substream)
{
	return snd_pcm_lib_free_vmalloc_buffer(substream);
}

/* prepare callback */
static int snd_sgio2audio_pcm_prepare(struct snd_pcm_substream *substream)
{
	struct snd_sgio2audio *chip = snd_pcm_substream_chip(substream);
	struct snd_pcm_runtime *runtime = substream->runtime;
	struct snd_sgio2audio_chan *chan = substream->runtime->private_data;
	int ch = chan->idx;
	unsigned long flags;

	spin_lock_irqsave(&chip->channel[ch].lock, flags);

	/* Setup the pseudo-dma transfer pointers.  */
	chip->channel[ch].pos = 0;
	chip->channel[ch].size = 0;
	chip->channel[ch].substream = substream;

	/* set AD1843 format */
	/* hardware format is always S16_LE */
	switch (substream->stream) {
	case SNDRV_PCM_STREAM_PLAYBACK:
		ad1843_setup_dac(&chip->ad1843,
				 ch - 1,
				 runtime->rate,
				 SNDRV_PCM_FORMAT_S16_LE,
				 runtime->channels);
		break;
	case SNDRV_PCM_STREAM_CAPTURE:
		ad1843_setup_adc(&chip->ad1843,
				 runtime->rate,
				 SNDRV_PCM_FORMAT_S16_LE,
				 runtime->channels);
		break;
	}
	spin_unlock_irqrestore(&chip->channel[ch].lock, flags);
	return 0;
}

/* trigger callback */
static int snd_sgio2audio_pcm_trigger(struct snd_pcm_substream *substream,
				      int cmd)
{
	switch (cmd) {
	case SNDRV_PCM_TRIGGER_START:
		/* start the PCM engine */
		snd_sgio2audio_dma_start(substream);
		break;
	case SNDRV_PCM_TRIGGER_STOP:
		/* stop the PCM engine */
		snd_sgio2audio_dma_stop(substream);
		break;
	default:
		return -EINVAL;
	}
	return 0;
}

/* pointer callback */
static snd_pcm_uframes_t
snd_sgio2audio_pcm_pointer(struct snd_pcm_substream *substream)
{
	struct snd_sgio2audio *chip = snd_pcm_substream_chip(substream);
	struct snd_sgio2audio_chan *chan = substream->runtime->private_data;

	/* get the current hardware pointer */
	return bytes_to_frames(substream->runtime,
			       chip->channel[chan->idx].pos);
}

/* operators */
static struct snd_pcm_ops snd_sgio2audio_playback1_ops = {
	.open =        snd_sgio2audio_playback1_open,
	.close =       snd_sgio2audio_pcm_close,
	.ioctl =       snd_pcm_lib_ioctl,
	.hw_params =   snd_sgio2audio_pcm_hw_params,
	.hw_free =     snd_sgio2audio_pcm_hw_free,
	.prepare =     snd_sgio2audio_pcm_prepare,
	.trigger =     snd_sgio2audio_pcm_trigger,
	.pointer =     snd_sgio2audio_pcm_pointer,
	.page =        snd_pcm_lib_get_vmalloc_page,
	.mmap =        snd_pcm_lib_mmap_vmalloc,
};

static struct snd_pcm_ops snd_sgio2audio_playback2_ops = {
	.open =        snd_sgio2audio_playback2_open,
	.close =       snd_sgio2audio_pcm_close,
	.ioctl =       snd_pcm_lib_ioctl,
	.hw_params =   snd_sgio2audio_pcm_hw_params,
	.hw_free =     snd_sgio2audio_pcm_hw_free,
	.prepare =     snd_sgio2audio_pcm_prepare,
	.trigger =     snd_sgio2audio_pcm_trigger,
	.pointer =     snd_sgio2audio_pcm_pointer,
	.page =        snd_pcm_lib_get_vmalloc_page,
	.mmap =        snd_pcm_lib_mmap_vmalloc,
};

static struct snd_pcm_ops snd_sgio2audio_capture_ops = {
	.open =        snd_sgio2audio_capture_open,
	.close =       snd_sgio2audio_pcm_close,
	.ioctl =       snd_pcm_lib_ioctl,
	.hw_params =   snd_sgio2audio_pcm_hw_params,
	.hw_free =     snd_sgio2audio_pcm_hw_free,
	.prepare =     snd_sgio2audio_pcm_prepare,
	.trigger =     snd_sgio2audio_pcm_trigger,
	.pointer =     snd_sgio2audio_pcm_pointer,
	.page =        snd_pcm_lib_get_vmalloc_page,
	.mmap =        snd_pcm_lib_mmap_vmalloc,
};

/*
 *  definitions of capture are omitted here...
 */

/* create a pcm device */
static int __devinit snd_sgio2audio_new_pcm(struct snd_sgio2audio *chip)
{
	struct snd_pcm *pcm;
	int err;

	/* create first pcm device with one outputs and one input */
	err = snd_pcm_new(chip->card, "SGI O2 Audio", 0, 1, 1, &pcm);
	if (err < 0)
		return err;

	pcm->private_data = chip;
	strcpy(pcm->name, "SGI O2 DAC1");

	/* set operators */
	snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK,
			&snd_sgio2audio_playback1_ops);
	snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE,
			&snd_sgio2audio_capture_ops);

	/* create second  pcm device with one outputs and no input */
	err = snd_pcm_new(chip->card, "SGI O2 Audio", 1, 1, 0, &pcm);
	if (err < 0)
		return err;

	pcm->private_data = chip;
	strcpy(pcm->name, "SGI O2 DAC2");

	/* set operators */
	snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK,
			&snd_sgio2audio_playback2_ops);

	return 0;
}

static struct {
	int idx;
	int irq;
	irqreturn_t (*isr)(int, void *);
	const char *desc;
} snd_sgio2_isr_table[] = {
	{
		.idx = 0,
		.irq = MACEISA_AUDIO1_DMAT_IRQ,
		.isr = snd_sgio2audio_dma_in_isr,
		.desc = "Capture DMA Channel 0"
	}, {
		.idx = 0,
		.irq = MACEISA_AUDIO1_OF_IRQ,
		.isr = snd_sgio2audio_error_isr,
		.desc = "Capture Overflow"
	}, {
		.idx = 1,
		.irq = MACEISA_AUDIO2_DMAT_IRQ,
		.isr = snd_sgio2audio_dma_out_isr,
		.desc = "Playback DMA Channel 1"
	}, {
		.idx = 1,
		.irq = MACEISA_AUDIO2_MERR_IRQ,
		.isr = snd_sgio2audio_error_isr,
		.desc = "Memory Error Channel 1"
	}, {
		.idx = 2,
		.irq = MACEISA_AUDIO3_DMAT_IRQ,
		.isr = snd_sgio2audio_dma_out_isr,
		.desc = "Playback DMA Channel 2"
	}, {
		.idx = 2,
		.irq = MACEISA_AUDIO3_MERR_IRQ,
		.isr = snd_sgio2audio_error_isr,
		.desc = "Memory Error Channel 2"
	}
};

/* ALSA driver */

static int snd_sgio2audio_free(struct snd_sgio2audio *chip)
{
	int i;

	/* reset interface */
	writeq(AUDIO_CONTROL_RESET, &mace->perif.audio.control);
	udelay(1);
	writeq(0, &mace->perif.audio.control);

	/* release IRQ's */
	for (i = 0; i < ARRAY_SIZE(snd_sgio2_isr_table); i++)
		free_irq(snd_sgio2_isr_table[i].irq,
			 &chip->channel[snd_sgio2_isr_table[i].idx]);

	dma_free_coherent(NULL, MACEISA_RINGBUFFERS_SIZE,
			  chip->ring_base, chip->ring_base_dma);

	/* release card data */
	kfree(chip);
	return 0;
}

static int snd_sgio2audio_dev_free(struct snd_device *device)
{
	struct snd_sgio2audio *chip = device->device_data;

	return snd_sgio2audio_free(chip);
}

static struct snd_device_ops ops = {
	.dev_free = snd_sgio2audio_dev_free,
};

static int __devinit snd_sgio2audio_create(struct snd_card *card,
					   struct snd_sgio2audio **rchip)
{
	struct snd_sgio2audio *chip;
	int i, err;

	*rchip = NULL;

	/* check if a codec is attached to the interface */
	/* (Audio or Audio/Video board present) */
	if (!(readq(&mace->perif.audio.control) & AUDIO_CONTROL_CODEC_PRESENT))
		return -ENOENT;

	chip = kzalloc(sizeof(struct snd_sgio2audio), GFP_KERNEL);
	if (chip == NULL)
		return -ENOMEM;

	chip->card = card;

	chip->ring_base = dma_alloc_coherent(NULL, MACEISA_RINGBUFFERS_SIZE,
					     &chip->ring_base_dma, GFP_USER);
	if (chip->ring_base == NULL) {
		printk(KERN_ERR
		       "sgio2audio: could not allocate ring buffers\n");
		kfree(chip);
		return -ENOMEM;
	}

	spin_lock_init(&chip->ad1843_lock);

	/* initialize channels */
	for (i = 0; i < 3; i++) {
		spin_lock_init(&chip->channel[i].lock);
		chip->channel[i].idx = i;
	}

	/* allocate IRQs */
	for (i = 0; i < ARRAY_SIZE(snd_sgio2_isr_table); i++) {
		if (request_irq(snd_sgio2_isr_table[i].irq,
				snd_sgio2_isr_table[i].isr,
				0,
				snd_sgio2_isr_table[i].desc,
				&chip->channel[snd_sgio2_isr_table[i].idx])) {
			snd_sgio2audio_free(chip);
			printk(KERN_ERR "sgio2audio: cannot allocate irq %d\n",
			       snd_sgio2_isr_table[i].irq);
			return -EBUSY;
		}
	}

	/* reset the interface */
	writeq(AUDIO_CONTROL_RESET, &mace->perif.audio.control);
	udelay(1);
	writeq(0, &mace->perif.audio.control);
	msleep_interruptible(1); /* give time to recover */

	/* set ring base */
	writeq(chip->ring_base_dma, &mace->perif.ctrl.ringbase);

	/* attach the AD1843 codec */
	chip->ad1843.read = read_ad1843_reg;
	chip->ad1843.write = write_ad1843_reg;
	chip->ad1843.chip = chip;

	/* initialize the AD1843 codec */
	err = ad1843_init(&chip->ad1843);
	if (err < 0) {
		snd_sgio2audio_free(chip);
		return err;
	}

	err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops);
	if (err < 0) {
		snd_sgio2audio_free(chip);
		return err;
	}
	*rchip = chip;
	return 0;
}

static int __devinit snd_sgio2audio_probe(struct platform_device *pdev)
{
	struct snd_card *card;
	struct snd_sgio2audio *chip;
	int err;

	err = snd_card_create(index, id, THIS_MODULE, 0, &card);
	if (err < 0)
		return err;

	err = snd_sgio2audio_create(card, &chip);
	if (err < 0) {
		snd_card_free(card);
		return err;
	}
	snd_card_set_dev(card, &pdev->dev);

	err = snd_sgio2audio_new_pcm(chip);
	if (err < 0) {
		snd_card_free(card);
		return err;
	}
	err = snd_sgio2audio_new_mixer(chip);
	if (err < 0) {
		snd_card_free(card);
		return err;
	}

	strcpy(card->driver, "SGI O2 Audio");
	strcpy(card->shortname, "SGI O2 Audio");
	sprintf(card->longname, "%s irq %i-%i",
		card->shortname,
		MACEISA_AUDIO1_DMAT_IRQ,
		MACEISA_AUDIO3_MERR_IRQ);

	err = snd_card_register(card);
	if (err < 0) {
		snd_card_free(card);
		return err;
	}
	platform_set_drvdata(pdev, card);
	return 0;
}

static int __devexit snd_sgio2audio_remove(struct platform_device *pdev)
{
	struct snd_card *card = platform_get_drvdata(pdev);

	snd_card_free(card);
	platform_set_drvdata(pdev, NULL);
	return 0;
}

static struct platform_driver sgio2audio_driver = {
	.probe	= snd_sgio2audio_probe,
	.remove	= __devexit_p(snd_sgio2audio_remove),
	.driver = {
		.name	= "sgio2audio",
		.owner	= THIS_MODULE,
	}
};

static int __init alsa_card_sgio2audio_init(void)
{
	return platform_driver_register(&sgio2audio_driver);
}

static void __exit alsa_card_sgio2audio_exit(void)
{
	platform_driver_unregister(&sgio2audio_driver);
}

module_init(alsa_card_sgio2audio_init)
module_exit(alsa_card_sgio2audio_exit)