summaryrefslogtreecommitdiffstats
path: root/sound/oss/dmasound/dmasound_paula.c
blob: 87910e9921332d770e5cd07d60434dd670b891fa (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
/*
 *  linux/sound/oss/dmasound/dmasound_paula.c
 *
 *  Amiga `Paula' DMA Sound Driver
 *
 *  See linux/sound/oss/dmasound/dmasound_core.c for copyright and credits
 *  prior to 28/01/2001
 *
 *  28/01/2001 [0.1] Iain Sandoe
 *		     - added versioning
 *		     - put in and populated the hardware_afmts field.
 *             [0.2] - put in SNDCTL_DSP_GETCAPS value.
 *	       [0.3] - put in constraint on state buffer usage.
 *	       [0.4] - put in default hard/soft settings
*/


#include <linux/module.h>
#include <linux/mm.h>
#include <linux/init.h>
#include <linux/ioport.h>
#include <linux/soundcard.h>
#include <linux/interrupt.h>
#include <linux/platform_device.h>

#include <asm/uaccess.h>
#include <asm/setup.h>
#include <asm/amigahw.h>
#include <asm/amigaints.h>
#include <asm/machdep.h>

#include "dmasound.h"

#define DMASOUND_PAULA_REVISION 0
#define DMASOUND_PAULA_EDITION 4

#define custom amiga_custom
   /*
    *	The minimum period for audio depends on htotal (for OCS/ECS/AGA)
    *	(Imported from arch/m68k/amiga/amisound.c)
    */

extern volatile u_short amiga_audio_min_period;


   /*
    *	amiga_mksound() should be able to restore the period after beeping
    *	(Imported from arch/m68k/amiga/amisound.c)
    */

extern u_short amiga_audio_period;


   /*
    *	Audio DMA masks
    */

#define AMI_AUDIO_OFF	(DMAF_AUD0 | DMAF_AUD1 | DMAF_AUD2 | DMAF_AUD3)
#define AMI_AUDIO_8	(DMAF_SETCLR | DMAF_MASTER | DMAF_AUD0 | DMAF_AUD1)
#define AMI_AUDIO_14	(AMI_AUDIO_8 | DMAF_AUD2 | DMAF_AUD3)


    /*
     *  Helper pointers for 16(14)-bit sound
     */

static int write_sq_block_size_half, write_sq_block_size_quarter;


/*** Low level stuff *********************************************************/


static void *AmiAlloc(unsigned int size, gfp_t flags);
static void AmiFree(void *obj, unsigned int size);
static int AmiIrqInit(void);
#ifdef MODULE
static void AmiIrqCleanUp(void);
#endif
static void AmiSilence(void);
static void AmiInit(void);
static int AmiSetFormat(int format);
static int AmiSetVolume(int volume);
static int AmiSetTreble(int treble);
static void AmiPlayNextFrame(int index);
static void AmiPlay(void);
static irqreturn_t AmiInterrupt(int irq, void *dummy);

#ifdef CONFIG_HEARTBEAT

    /*
     *  Heartbeat interferes with sound since the 7 kHz low-pass filter and the
     *  power LED are controlled by the same line.
     */

static void (*saved_heartbeat)(int) = NULL;

static inline void disable_heartbeat(void)
{
	if (mach_heartbeat) {
	    saved_heartbeat = mach_heartbeat;
	    mach_heartbeat = NULL;
	}
	AmiSetTreble(dmasound.treble);
}

static inline void enable_heartbeat(void)
{
	if (saved_heartbeat)
	    mach_heartbeat = saved_heartbeat;
}
#else /* !CONFIG_HEARTBEAT */
#define disable_heartbeat()	do { } while (0)
#define enable_heartbeat()	do { } while (0)
#endif /* !CONFIG_HEARTBEAT */


/*** Mid level stuff *********************************************************/

static void AmiMixerInit(void);
static int AmiMixerIoctl(u_int cmd, u_long arg);
static int AmiWriteSqSetup(void);
static int AmiStateInfo(char *buffer, size_t space);


/*** Translations ************************************************************/

/* ++TeSche: radically changed for new expanding purposes...
 *
 * These two routines now deal with copying/expanding/translating the samples
 * from user space into our buffer at the right frequency. They take care about
 * how much data there's actually to read, how much buffer space there is and
 * to convert samples into the right frequency/encoding. They will only work on
 * complete samples so it may happen they leave some bytes in the input stream
 * if the user didn't write a multiple of the current sample size. They both
 * return the number of bytes they've used from both streams so you may detect
 * such a situation. Luckily all programs should be able to cope with that.
 *
 * I think I've optimized anything as far as one can do in plain C, all
 * variables should fit in registers and the loops are really short. There's
 * one loop for every possible situation. Writing a more generalized and thus
 * parameterized loop would only produce slower code. Feel free to optimize
 * this in assembler if you like. :)
 *
 * I think these routines belong here because they're not yet really hardware
 * independent, especially the fact that the Falcon can play 16bit samples
 * only in stereo is hardcoded in both of them!
 *
 * ++geert: split in even more functions (one per format)
 */


    /*
     *  Native format
     */

static ssize_t ami_ct_s8(const u_char __user *userPtr, size_t userCount,
			 u_char frame[], ssize_t *frameUsed, ssize_t frameLeft)
{
	ssize_t count, used;

	if (!dmasound.soft.stereo) {
		void *p = &frame[*frameUsed];
		count = min_t(unsigned long, userCount, frameLeft) & ~1;
		used = count;
		if (copy_from_user(p, userPtr, count))
			return -EFAULT;
	} else {
		u_char *left = &frame[*frameUsed>>1];
		u_char *right = left+write_sq_block_size_half;
		count = min_t(unsigned long, userCount, frameLeft)>>1 & ~1;
		used = count*2;
		while (count > 0) {
			if (get_user(*left++, userPtr++)
			    || get_user(*right++, userPtr++))
				return -EFAULT;
			count--;
		}
	}
	*frameUsed += used;
	return used;
}


    /*
     *  Copy and convert 8 bit data
     */

#define GENERATE_AMI_CT8(funcname, convsample)				\
static ssize_t funcname(const u_char __user *userPtr, size_t userCount,	\
			u_char frame[], ssize_t *frameUsed,		\
			ssize_t frameLeft)				\
{									\
	ssize_t count, used;						\
									\
	if (!dmasound.soft.stereo) {					\
		u_char *p = &frame[*frameUsed];				\
		count = min_t(size_t, userCount, frameLeft) & ~1;	\
		used = count;						\
		while (count > 0) {					\
			u_char data;					\
			if (get_user(data, userPtr++))			\
				return -EFAULT;				\
			*p++ = convsample(data);			\
			count--;					\
		}							\
	} else {							\
		u_char *left = &frame[*frameUsed>>1];			\
		u_char *right = left+write_sq_block_size_half;		\
		count = min_t(size_t, userCount, frameLeft)>>1 & ~1;	\
		used = count*2;						\
		while (count > 0) {					\
			u_char data;					\
			if (get_user(data, userPtr++))			\
				return -EFAULT;				\
			*left++ = convsample(data);			\
			if (get_user(data, userPtr++))			\
				return -EFAULT;				\
			*right++ = convsample(data);			\
			count--;					\
		}							\
	}								\
	*frameUsed += used;						\
	return used;							\
}

#define AMI_CT_ULAW(x)	(dmasound_ulaw2dma8[(x)])
#define AMI_CT_ALAW(x)	(dmasound_alaw2dma8[(x)])
#define AMI_CT_U8(x)	((x) ^ 0x80)

GENERATE_AMI_CT8(ami_ct_ulaw, AMI_CT_ULAW)
GENERATE_AMI_CT8(ami_ct_alaw, AMI_CT_ALAW)
GENERATE_AMI_CT8(ami_ct_u8, AMI_CT_U8)


    /*
     *  Copy and convert 16 bit data
     */

#define GENERATE_AMI_CT_16(funcname, convsample)			\
static ssize_t funcname(const u_char __user *userPtr, size_t userCount,	\
			u_char frame[], ssize_t *frameUsed,		\
			ssize_t frameLeft)				\
{									\
	const u_short __user *ptr = (const u_short __user *)userPtr;	\
	ssize_t count, used;						\
	u_short data;							\
									\
	if (!dmasound.soft.stereo) {					\
		u_char *high = &frame[*frameUsed>>1];			\
		u_char *low = high+write_sq_block_size_half;		\
		count = min_t(size_t, userCount, frameLeft)>>1 & ~1;	\
		used = count*2;						\
		while (count > 0) {					\
			if (get_user(data, ptr++))			\
				return -EFAULT;				\
			data = convsample(data);			\
			*high++ = data>>8;				\
			*low++ = (data>>2) & 0x3f;			\
			count--;					\
		}							\
	} else {							\
		u_char *lefth = &frame[*frameUsed>>2];			\
		u_char *leftl = lefth+write_sq_block_size_quarter;	\
		u_char *righth = lefth+write_sq_block_size_half;	\
		u_char *rightl = righth+write_sq_block_size_quarter;	\
		count = min_t(size_t, userCount, frameLeft)>>2 & ~1;	\
		used = count*4;						\
		while (count > 0) {					\
			if (get_user(data, ptr++))			\
				return -EFAULT;				\
			data = convsample(data);			\
			*lefth++ = data>>8;				\
			*leftl++ = (data>>2) & 0x3f;			\
			if (get_user(data, ptr++))			\
				return -EFAULT;				\
			data = convsample(data);			\
			*righth++ = data>>8;				\
			*rightl++ = (data>>2) & 0x3f;			\
			count--;					\
		}							\
	}								\
	*frameUsed += used;						\
	return used;							\
}

#define AMI_CT_S16BE(x)	(x)
#define AMI_CT_U16BE(x)	((x) ^ 0x8000)
#define AMI_CT_S16LE(x)	(le2be16((x)))
#define AMI_CT_U16LE(x)	(le2be16((x)) ^ 0x8000)

GENERATE_AMI_CT_16(ami_ct_s16be, AMI_CT_S16BE)
GENERATE_AMI_CT_16(ami_ct_u16be, AMI_CT_U16BE)
GENERATE_AMI_CT_16(ami_ct_s16le, AMI_CT_S16LE)
GENERATE_AMI_CT_16(ami_ct_u16le, AMI_CT_U16LE)


static TRANS transAmiga = {
	.ct_ulaw	= ami_ct_ulaw,
	.ct_alaw	= ami_ct_alaw,
	.ct_s8		= ami_ct_s8,
	.ct_u8		= ami_ct_u8,
	.ct_s16be	= ami_ct_s16be,
	.ct_u16be	= ami_ct_u16be,
	.ct_s16le	= ami_ct_s16le,
	.ct_u16le	= ami_ct_u16le,
};

/*** Low level stuff *********************************************************/

static inline void StopDMA(void)
{
	custom.aud[0].audvol = custom.aud[1].audvol = 0;
	custom.aud[2].audvol = custom.aud[3].audvol = 0;
	custom.dmacon = AMI_AUDIO_OFF;
	enable_heartbeat();
}

static void *AmiAlloc(unsigned int size, gfp_t flags)
{
	return amiga_chip_alloc((long)size, "dmasound [Paula]");
}

static void AmiFree(void *obj, unsigned int size)
{
	amiga_chip_free (obj);
}

static int __init AmiIrqInit(void)
{
	/* turn off DMA for audio channels */
	StopDMA();

	/* Register interrupt handler. */
	if (request_irq(IRQ_AMIGA_AUD0, AmiInterrupt, 0, "DMA sound",
			AmiInterrupt))
		return 0;
	return 1;
}

#ifdef MODULE
static void AmiIrqCleanUp(void)
{
	/* turn off DMA for audio channels */
	StopDMA();
	/* release the interrupt */
	free_irq(IRQ_AMIGA_AUD0, AmiInterrupt);
}
#endif /* MODULE */

static void AmiSilence(void)
{
	/* turn off DMA for audio channels */
	StopDMA();
}


static void AmiInit(void)
{
	int period, i;

	AmiSilence();

	if (dmasound.soft.speed)
		period = amiga_colorclock/dmasound.soft.speed-1;
	else
		period = amiga_audio_min_period;
	dmasound.hard = dmasound.soft;
	dmasound.trans_write = &transAmiga;

	if (period < amiga_audio_min_period) {
		/* we would need to squeeze the sound, but we won't do that */
		period = amiga_audio_min_period;
	} else if (period > 65535) {
		period = 65535;
	}
	dmasound.hard.speed = amiga_colorclock/(period+1);

	for (i = 0; i < 4; i++)
		custom.aud[i].audper = period;
	amiga_audio_period = period;
}


static int AmiSetFormat(int format)
{
	int size;

	/* Amiga sound DMA supports 8bit and 16bit (pseudo 14 bit) modes */

	switch (format) {
	case AFMT_QUERY:
		return dmasound.soft.format;
	case AFMT_MU_LAW:
	case AFMT_A_LAW:
	case AFMT_U8:
	case AFMT_S8:
		size = 8;
		break;
	case AFMT_S16_BE:
	case AFMT_U16_BE:
	case AFMT_S16_LE:
	case AFMT_U16_LE:
		size = 16;
		break;
	default: /* :-) */
		size = 8;
		format = AFMT_S8;
	}

	dmasound.soft.format = format;
	dmasound.soft.size = size;
	if (dmasound.minDev == SND_DEV_DSP) {
		dmasound.dsp.format = format;
		dmasound.dsp.size = dmasound.soft.size;
	}
	AmiInit();

	return format;
}


#define VOLUME_VOXWARE_TO_AMI(v) \
	(((v) < 0) ? 0 : ((v) > 100) ? 64 : ((v) * 64)/100)
#define VOLUME_AMI_TO_VOXWARE(v) ((v)*100/64)

static int AmiSetVolume(int volume)
{
	dmasound.volume_left = VOLUME_VOXWARE_TO_AMI(volume & 0xff);
	custom.aud[0].audvol = dmasound.volume_left;
	dmasound.volume_right = VOLUME_VOXWARE_TO_AMI((volume & 0xff00) >> 8);
	custom.aud[1].audvol = dmasound.volume_right;
	if (dmasound.hard.size == 16) {
		if (dmasound.volume_left == 64 && dmasound.volume_right == 64) {
			custom.aud[2].audvol = 1;
			custom.aud[3].audvol = 1;
		} else {
			custom.aud[2].audvol = 0;
			custom.aud[3].audvol = 0;
		}
	}
	return VOLUME_AMI_TO_VOXWARE(dmasound.volume_left) |
	       (VOLUME_AMI_TO_VOXWARE(dmasound.volume_right) << 8);
}

static int AmiSetTreble(int treble)
{
	dmasound.treble = treble;
	if (treble < 50)
		ciaa.pra &= ~0x02;
	else
		ciaa.pra |= 0x02;
	return treble;
}


#define AMI_PLAY_LOADED		1
#define AMI_PLAY_PLAYING	2
#define AMI_PLAY_MASK		3


static void AmiPlayNextFrame(int index)
{
	u_char *start, *ch0, *ch1, *ch2, *ch3;
	u_long size;

	/* used by AmiPlay() if all doubts whether there really is something
	 * to be played are already wiped out.
	 */
	start = write_sq.buffers[write_sq.front];
	size = (write_sq.count == index ? write_sq.rear_size
					: write_sq.block_size)>>1;

	if (dmasound.hard.stereo) {
		ch0 = start;
		ch1 = start+write_sq_block_size_half;
		size >>= 1;
	} else {
		ch0 = start;
		ch1 = start;
	}

	disable_heartbeat();
	custom.aud[0].audvol = dmasound.volume_left;
	custom.aud[1].audvol = dmasound.volume_right;
	if (dmasound.hard.size == 8) {
		custom.aud[0].audlc = (u_short *)ZTWO_PADDR(ch0);
		custom.aud[0].audlen = size;
		custom.aud[1].audlc = (u_short *)ZTWO_PADDR(ch1);
		custom.aud[1].audlen = size;
		custom.dmacon = AMI_AUDIO_8;
	} else {
		size >>= 1;
		custom.aud[0].audlc = (u_short *)ZTWO_PADDR(ch0);
		custom.aud[0].audlen = size;
		custom.aud[1].audlc = (u_short *)ZTWO_PADDR(ch1);
		custom.aud[1].audlen = size;
		if (dmasound.volume_left == 64 && dmasound.volume_right == 64) {
			/* We can play pseudo 14-bit only with the maximum volume */
			ch3 = ch0+write_sq_block_size_quarter;
			ch2 = ch1+write_sq_block_size_quarter;
			custom.aud[2].audvol = 1;  /* we are being affected by the beeps */
			custom.aud[3].audvol = 1;  /* restoring volume here helps a bit */
			custom.aud[2].audlc = (u_short *)ZTWO_PADDR(ch2);
			custom.aud[2].audlen = size;
			custom.aud[3].audlc = (u_short *)ZTWO_PADDR(ch3);
			custom.aud[3].audlen = size;
			custom.dmacon = AMI_AUDIO_14;
		} else {
			custom.aud[2].audvol = 0;
			custom.aud[3].audvol = 0;
			custom.dmacon = AMI_AUDIO_8;
		}
	}
	write_sq.front = (write_sq.front+1) % write_sq.max_count;
	write_sq.active |= AMI_PLAY_LOADED;
}


static void AmiPlay(void)
{
	int minframes = 1;

	custom.intena = IF_AUD0;

	if (write_sq.active & AMI_PLAY_LOADED) {
		/* There's already a frame loaded */
		custom.intena = IF_SETCLR | IF_AUD0;
		return;
	}

	if (write_sq.active & AMI_PLAY_PLAYING)
		/* Increase threshold: frame 1 is already being played */
		minframes = 2;

	if (write_sq.count < minframes) {
		/* Nothing to do */
		custom.intena = IF_SETCLR | IF_AUD0;
		return;
	}

	if (write_sq.count <= minframes &&
	    write_sq.rear_size < write_sq.block_size && !write_sq.syncing) {
		/* hmmm, the only existing frame is not
		 * yet filled and we're not syncing?
		 */
		custom.intena = IF_SETCLR | IF_AUD0;
		return;
	}

	AmiPlayNextFrame(minframes);

	custom.intena = IF_SETCLR | IF_AUD0;
}


static irqreturn_t AmiInterrupt(int irq, void *dummy)
{
	int minframes = 1;

	custom.intena = IF_AUD0;

	if (!write_sq.active) {
		/* Playing was interrupted and sq_reset() has already cleared
		 * the sq variables, so better don't do anything here.
		 */
		WAKE_UP(write_sq.sync_queue);
		return IRQ_HANDLED;
	}

	if (write_sq.active & AMI_PLAY_PLAYING) {
		/* We've just finished a frame */
		write_sq.count--;
		WAKE_UP(write_sq.action_queue);
	}

	if (write_sq.active & AMI_PLAY_LOADED)
		/* Increase threshold: frame 1 is already being played */
		minframes = 2;

	/* Shift the flags */
	write_sq.active = (write_sq.active<<1) & AMI_PLAY_MASK;

	if (!write_sq.active)
		/* No frame is playing, disable audio DMA */
		StopDMA();

	custom.intena = IF_SETCLR | IF_AUD0;

	if (write_sq.count >= minframes)
		/* Try to play the next frame */
		AmiPlay();

	if (!write_sq.active)
		/* Nothing to play anymore.
		   Wake up a process waiting for audio output to drain. */
		WAKE_UP(write_sq.sync_queue);
	return IRQ_HANDLED;
}

/*** Mid level stuff *********************************************************/


/*
 * /dev/mixer abstraction
 */

static void __init AmiMixerInit(void)
{
	dmasound.volume_left = 64;
	dmasound.volume_right = 64;
	custom.aud[0].audvol = dmasound.volume_left;
	custom.aud[3].audvol = 1;	/* For pseudo 14bit */
	custom.aud[1].audvol = dmasound.volume_right;
	custom.aud[2].audvol = 1;	/* For pseudo 14bit */
	dmasound.treble = 50;
}

static int AmiMixerIoctl(u_int cmd, u_long arg)
{
	int data;
	switch (cmd) {
	    case SOUND_MIXER_READ_DEVMASK:
		    return IOCTL_OUT(arg, SOUND_MASK_VOLUME | SOUND_MASK_TREBLE);
	    case SOUND_MIXER_READ_RECMASK:
		    return IOCTL_OUT(arg, 0);
	    case SOUND_MIXER_READ_STEREODEVS:
		    return IOCTL_OUT(arg, SOUND_MASK_VOLUME);
	    case SOUND_MIXER_READ_VOLUME:
		    return IOCTL_OUT(arg,
			    VOLUME_AMI_TO_VOXWARE(dmasound.volume_left) |
			    VOLUME_AMI_TO_VOXWARE(dmasound.volume_right) << 8);
	    case SOUND_MIXER_WRITE_VOLUME:
		    IOCTL_IN(arg, data);
		    return IOCTL_OUT(arg, dmasound_set_volume(data));
	    case SOUND_MIXER_READ_TREBLE:
		    return IOCTL_OUT(arg, dmasound.treble);
	    case SOUND_MIXER_WRITE_TREBLE:
		    IOCTL_IN(arg, data);
		    return IOCTL_OUT(arg, dmasound_set_treble(data));
	}
	return -EINVAL;
}


static int AmiWriteSqSetup(void)
{
	write_sq_block_size_half = write_sq.block_size>>1;
	write_sq_block_size_quarter = write_sq_block_size_half>>1;
	return 0;
}


static int AmiStateInfo(char *buffer, size_t space)
{
	int len = 0;
	len += sprintf(buffer+len, "\tsound.volume_left = %d [0...64]\n",
		       dmasound.volume_left);
	len += sprintf(buffer+len, "\tsound.volume_right = %d [0...64]\n",
		       dmasound.volume_right);
	if (len >= space) {
		printk(KERN_ERR "dmasound_paula: overflowed state buffer alloc.\n") ;
		len = space ;
	}
	return len;
}


/*** Machine definitions *****************************************************/

static SETTINGS def_hard = {
	.format	= AFMT_S8,
	.stereo	= 0,
	.size	= 8,
	.speed	= 8000
} ;

static SETTINGS def_soft = {
	.format	= AFMT_U8,
	.stereo	= 0,
	.size	= 8,
	.speed	= 8000
} ;

static MACHINE machAmiga = {
	.name		= "Amiga",
	.name2		= "AMIGA",
	.owner		= THIS_MODULE,
	.dma_alloc	= AmiAlloc,
	.dma_free	= AmiFree,
	.irqinit	= AmiIrqInit,
#ifdef MODULE
	.irqcleanup	= AmiIrqCleanUp,
#endif /* MODULE */
	.init		= AmiInit,
	.silence	= AmiSilence,
	.setFormat	= AmiSetFormat,
	.setVolume	= AmiSetVolume,
	.setTreble	= AmiSetTreble,
	.play		= AmiPlay,
	.mixer_init	= AmiMixerInit,
	.mixer_ioctl	= AmiMixerIoctl,
	.write_sq_setup	= AmiWriteSqSetup,
	.state_info	= AmiStateInfo,
	.min_dsp_speed	= 8000,
	.version	= ((DMASOUND_PAULA_REVISION<<8) | DMASOUND_PAULA_EDITION),
	.hardware_afmts	= (AFMT_S8 | AFMT_S16_BE), /* h'ware-supported formats *only* here */
	.capabilities	= DSP_CAP_BATCH          /* As per SNDCTL_DSP_GETCAPS */
};


/*** Config & Setup **********************************************************/


static int __init amiga_audio_probe(struct platform_device *pdev)
{
	dmasound.mach = machAmiga;
	dmasound.mach.default_hard = def_hard ;
	dmasound.mach.default_soft = def_soft ;
	return dmasound_init();
}

static int __exit amiga_audio_remove(struct platform_device *pdev)
{
	dmasound_deinit();
	return 0;
}

static struct platform_driver amiga_audio_driver = {
	.remove = __exit_p(amiga_audio_remove),
	.driver   = {
		.name	= "amiga-audio",
		.owner	= THIS_MODULE,
	},
};

static int __init amiga_audio_init(void)
{
	return platform_driver_probe(&amiga_audio_driver, amiga_audio_probe);
}

module_init(amiga_audio_init);

static void __exit amiga_audio_exit(void)
{
	platform_driver_unregister(&amiga_audio_driver);
}

module_exit(amiga_audio_exit);

MODULE_LICENSE("GPL");
MODULE_ALIAS("platform:amiga-audio");