aboutsummaryrefslogtreecommitdiff
path: root/sound/arm/sa11xx-uda1341.c
blob: faeddf3ecedb871835dbae30ddf74b07d85e42ad (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
/*
 *  Driver for Philips UDA1341TS on Compaq iPAQ H3600 soundcard
 *  Copyright (C) 2002 Tomas Kasparek <tomas.kasparek@seznam.cz>
 *
 *   This program is free software; you can redistribute it and/or modify
 *   it under the terms of the GNU General Public License.
 * 
 * History:
 *
 * 2002-03-13   Tomas Kasparek  initial release - based on h3600-uda1341.c from OSS
 * 2002-03-20   Tomas Kasparek  playback over ALSA is working
 * 2002-03-28   Tomas Kasparek  playback over OSS emulation is working
 * 2002-03-29   Tomas Kasparek  basic capture is working (native ALSA)
 * 2002-03-29   Tomas Kasparek  capture is working (OSS emulation)
 * 2002-04-04   Tomas Kasparek  better rates handling (allow non-standard rates)
 * 2003-02-14   Brian Avery     fixed full duplex mode, other updates
 * 2003-02-20   Tomas Kasparek  merged updates by Brian (except HAL)
 * 2003-04-19   Jaroslav Kysela recoded DMA stuff to follow 2.4.18rmk3-hh24 kernel
 *                              working suspend and resume
 * 2003-04-28   Tomas Kasparek  updated work by Jaroslav to compile it under 2.5.x again
 *                              merged HAL layer (patches from Brian)
 */

/***************************************************************************************************
*
* To understand what Alsa Drivers should be doing look at "Writing an Alsa Driver" by Takashi Iwai
* available in the Alsa doc section on the website		
* 
* A few notes to make things clearer. The UDA1341 is hooked up to Serial port 4 on the SA1100.
* We are using  SSP mode to talk to the UDA1341. The UDA1341 bit & wordselect clocks are generated
* by this UART. Unfortunately, the clock only runs if the transmit buffer has something in it.
* So, if we are just recording, we feed the transmit DMA stream a bunch of 0x0000 so that the
* transmit buffer is full and the clock keeps going. The zeroes come from FLUSH_BASE_PHYS which
* is a mem loc that always decodes to 0's w/ no off chip access.
*
* Some alsa terminology:
*	frame => num_channels * sample_size  e.g stereo 16 bit is 2 * 16 = 32 bytes
*	period => the least number of bytes that will generate an interrupt e.g. we have a 1024 byte
*             buffer and 4 periods in the runtime structure this means we'll get an int every 256
*             bytes or 4 times per buffer.
*             A number of the sizes are in frames rather than bytes, use frames_to_bytes and
*             bytes_to_frames to convert.  The easiest way to tell the units is to look at the
*             type i.e. runtime-> buffer_size is in frames and its type is snd_pcm_uframes_t
*             
*	Notes about the pointer fxn:
*	The pointer fxn needs to return the offset into the dma buffer in frames.
*	Interrupts must be blocked before calling the dma_get_pos fxn to avoid race with interrupts.
*
*	Notes about pause/resume
*	Implementing this would be complicated so it's skipped.  The problem case is:
*	A full duplex connection is going, then play is paused. At this point you need to start xmitting
*	0's to keep the record active which means you cant just freeze the dma and resume it later you'd
*	need to	save off the dma info, and restore it properly on a resume.  Yeach!
*
*	Notes about transfer methods:
*	The async write calls fail.  I probably need to implement something else to support them?
* 
***************************************************************************************************/

#include <linux/module.h>
#include <linux/moduleparam.h>
#include <linux/init.h>
#include <linux/err.h>
#include <linux/platform_device.h>
#include <linux/errno.h>
#include <linux/ioctl.h>
#include <linux/delay.h>
#include <linux/slab.h>

#ifdef CONFIG_PM
#include <linux/pm.h>
#endif

#include <asm/hardware.h>
#include <asm/arch/h3600.h>
#include <asm/mach-types.h>
#include <asm/dma.h>

#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/initval.h>

#include <linux/l3/l3.h>

#undef DEBUG_MODE
#undef DEBUG_FUNCTION_NAMES
#include <sound/uda1341.h>

/*
 * FIXME: Is this enough as autodetection of 2.4.X-rmkY-hhZ kernels?
 * We use DMA stuff from 2.4.18-rmk3-hh24 here to be able to compile this
 * module for Familiar 0.6.1
 */

/* {{{ Type definitions */

MODULE_AUTHOR("Tomas Kasparek <tomas.kasparek@seznam.cz>");
MODULE_LICENSE("GPL");
MODULE_DESCRIPTION("SA1100/SA1111 + UDA1341TS driver for ALSA");
MODULE_SUPPORTED_DEVICE("{{UDA1341,iPAQ H3600 UDA1341TS}}");

static char *id;	/* ID for this card */

module_param(id, charp, 0444);
MODULE_PARM_DESC(id, "ID string for SA1100/SA1111 + UDA1341TS soundcard.");

struct audio_stream {
	char *id;		/* identification string */
	int stream_id;		/* numeric identification */	
	dma_device_t dma_dev;	/* device identifier for DMA */
#ifdef HH_VERSION
	dmach_t dmach;		/* dma channel identification */
#else
	dma_regs_t *dma_regs;	/* points to our DMA registers */
#endif
	unsigned int active:1;	/* we are using this stream for transfer now */
	int period;		/* current transfer period */
	int periods;		/* current count of periods registerd in the DMA engine */
	int tx_spin;		/* are we recoding - flag used to do DMA trans. for sync */
	unsigned int old_offset;
	spinlock_t dma_lock;	/* for locking in DMA operations (see dma-sa1100.c in the kernel) */
	struct snd_pcm_substream *stream;
};

struct sa11xx_uda1341 {
	struct snd_card *card;
	struct l3_client *uda1341;
	struct snd_pcm *pcm;
	long samplerate;
	struct audio_stream s[2];	/* playback & capture */
};

static unsigned int rates[] = {
	8000,  10666, 10985, 14647,
	16000, 21970, 22050, 24000,
	29400, 32000, 44100, 48000,
};

static struct snd_pcm_hw_constraint_list hw_constraints_rates = {
	.count	= ARRAY_SIZE(rates),
	.list	= rates,
	.mask	= 0,
};

static struct platform_device *device;

/* }}} */

/* {{{ Clock and sample rate stuff */

/*
 * Stop-gap solution until rest of hh.org HAL stuff is merged.
 */
#define GPIO_H3600_CLK_SET0		GPIO_GPIO (12)
#define GPIO_H3600_CLK_SET1		GPIO_GPIO (13)

#ifdef CONFIG_SA1100_H3XXX
#define	clr_sa11xx_uda1341_egpio(x)	clr_h3600_egpio(x)
#define set_sa11xx_uda1341_egpio(x)	set_h3600_egpio(x)
#else
#error This driver could serve H3x00 handhelds only!
#endif

static void sa11xx_uda1341_set_audio_clock(long val)
{
	switch (val) {
	case 24000: case 32000: case 48000:	/* 00: 12.288 MHz */
		GPCR = GPIO_H3600_CLK_SET0 | GPIO_H3600_CLK_SET1;
		break;

	case 22050: case 29400: case 44100:	/* 01: 11.2896 MHz */
		GPSR = GPIO_H3600_CLK_SET0;
		GPCR = GPIO_H3600_CLK_SET1;
		break;

	case 8000: case 10666: case 16000:	/* 10: 4.096 MHz */
		GPCR = GPIO_H3600_CLK_SET0;
		GPSR = GPIO_H3600_CLK_SET1;
		break;

	case 10985: case 14647: case 21970:	/* 11: 5.6245 MHz */
		GPSR = GPIO_H3600_CLK_SET0 | GPIO_H3600_CLK_SET1;
		break;
	}
}

static void sa11xx_uda1341_set_samplerate(struct sa11xx_uda1341 *sa11xx_uda1341, long rate)
{
	int clk_div = 0;
	int clk=0;

	/* We don't want to mess with clocks when frames are in flight */
	Ser4SSCR0 &= ~SSCR0_SSE;
	/* wait for any frame to complete */
	udelay(125);

	/*
	 * We have the following clock sources:
	 * 4.096 MHz, 5.6245 MHz, 11.2896 MHz, 12.288 MHz
	 * Those can be divided either by 256, 384 or 512.
	 * This makes up 12 combinations for the following samplerates...
	 */
	if (rate >= 48000)
		rate = 48000;
	else if (rate >= 44100)
		rate = 44100;
	else if (rate >= 32000)
		rate = 32000;
	else if (rate >= 29400)
		rate = 29400;
	else if (rate >= 24000)
		rate = 24000;
	else if (rate >= 22050)
		rate = 22050;
	else if (rate >= 21970)
		rate = 21970;
	else if (rate >= 16000)
		rate = 16000;
	else if (rate >= 14647)
		rate = 14647;
	else if (rate >= 10985)
		rate = 10985;
	else if (rate >= 10666)
		rate = 10666;
	else
		rate = 8000;

	/* Set the external clock generator */
	
	sa11xx_uda1341_set_audio_clock(rate);

	/* Select the clock divisor */
	switch (rate) {
	case 8000:
	case 10985:
	case 22050:
	case 24000:
		clk = F512;
		clk_div = SSCR0_SerClkDiv(16);
		break;
	case 16000:
	case 21970:
	case 44100:
	case 48000:
		clk = F256;
		clk_div = SSCR0_SerClkDiv(8);
		break;
	case 10666:
	case 14647:
	case 29400:
	case 32000:
		clk = F384;
		clk_div = SSCR0_SerClkDiv(12);
		break;
	}

	/* FMT setting should be moved away when other FMTs are added (FIXME) */
	l3_command(sa11xx_uda1341->uda1341, CMD_FORMAT, (void *)LSB16);
	
	l3_command(sa11xx_uda1341->uda1341, CMD_FS, (void *)clk);        
	Ser4SSCR0 = (Ser4SSCR0 & ~0xff00) + clk_div + SSCR0_SSE;
	sa11xx_uda1341->samplerate = rate;
}

/* }}} */

/* {{{ HW init and shutdown */

static void sa11xx_uda1341_audio_init(struct sa11xx_uda1341 *sa11xx_uda1341)
{
	unsigned long flags;

	/* Setup DMA stuff */
	sa11xx_uda1341->s[SNDRV_PCM_STREAM_PLAYBACK].id = "UDA1341 out";
	sa11xx_uda1341->s[SNDRV_PCM_STREAM_PLAYBACK].stream_id = SNDRV_PCM_STREAM_PLAYBACK;
	sa11xx_uda1341->s[SNDRV_PCM_STREAM_PLAYBACK].dma_dev = DMA_Ser4SSPWr;

	sa11xx_uda1341->s[SNDRV_PCM_STREAM_CAPTURE].id = "UDA1341 in";
	sa11xx_uda1341->s[SNDRV_PCM_STREAM_CAPTURE].stream_id = SNDRV_PCM_STREAM_CAPTURE;
	sa11xx_uda1341->s[SNDRV_PCM_STREAM_CAPTURE].dma_dev = DMA_Ser4SSPRd;

	/* Initialize the UDA1341 internal state */
       
	/* Setup the uarts */
	local_irq_save(flags);
	GAFR |= (GPIO_SSP_CLK);
	GPDR &= ~(GPIO_SSP_CLK);
	Ser4SSCR0 = 0;
	Ser4SSCR0 = SSCR0_DataSize(16) + SSCR0_TI + SSCR0_SerClkDiv(8);
	Ser4SSCR1 = SSCR1_SClkIactL + SSCR1_SClk1P + SSCR1_ExtClk;
	Ser4SSCR0 |= SSCR0_SSE;
	local_irq_restore(flags);

	/* Enable the audio power */

	clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_CODEC_NRESET);
	set_sa11xx_uda1341_egpio(IPAQ_EGPIO_AUDIO_ON);
	set_sa11xx_uda1341_egpio(IPAQ_EGPIO_QMUTE);
 
	/* Wait for the UDA1341 to wake up */
	mdelay(1); //FIXME - was removed by Perex - Why?

	/* Initialize the UDA1341 internal state */
	l3_open(sa11xx_uda1341->uda1341);
	
	/* external clock configuration (after l3_open - regs must be initialized */
	sa11xx_uda1341_set_samplerate(sa11xx_uda1341, sa11xx_uda1341->samplerate);

	/* Wait for the UDA1341 to wake up */
	set_sa11xx_uda1341_egpio(IPAQ_EGPIO_CODEC_NRESET);
	mdelay(1);	

	/* make the left and right channels unswapped (flip the WS latch) */
	Ser4SSDR = 0;

	clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_QMUTE);
}

static void sa11xx_uda1341_audio_shutdown(struct sa11xx_uda1341 *sa11xx_uda1341)
{
	/* mute on */
	set_sa11xx_uda1341_egpio(IPAQ_EGPIO_QMUTE);
	
	/* disable the audio power and all signals leading to the audio chip */
	l3_close(sa11xx_uda1341->uda1341);
	Ser4SSCR0 = 0;
	clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_CODEC_NRESET);

	/* power off and mute off */
	/* FIXME - is muting off necesary??? */

	clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_AUDIO_ON);
	clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_QMUTE);
}

/* }}} */

/* {{{ DMA staff */

/*
 * these are the address and sizes used to fill the xmit buffer
 * so we can get a clock in record only mode
 */
#define FORCE_CLOCK_ADDR		(dma_addr_t)FLUSH_BASE_PHYS
#define FORCE_CLOCK_SIZE		4096 // was 2048

// FIXME Why this value exactly - wrote comment
#define DMA_BUF_SIZE	8176	/* <= MAX_DMA_SIZE from asm/arch-sa1100/dma.h */

#ifdef HH_VERSION

static int audio_dma_request(struct audio_stream *s, void (*callback)(void *, int))
{
	int ret;

	ret = sa1100_request_dma(&s->dmach, s->id, s->dma_dev);
	if (ret < 0) {
		printk(KERN_ERR "unable to grab audio dma 0x%x\n", s->dma_dev);
		return ret;
	}
	sa1100_dma_set_callback(s->dmach, callback);
	return 0;
}

static inline void audio_dma_free(struct audio_stream *s)
{
	sa1100_free_dma(s->dmach);
	s->dmach = -1;
}

#else

static int audio_dma_request(struct audio_stream *s, void (*callback)(void *))
{
	int ret;

	ret = sa1100_request_dma(s->dma_dev, s->id, callback, s, &s->dma_regs);
	if (ret < 0)
		printk(KERN_ERR "unable to grab audio dma 0x%x\n", s->dma_dev);
	return ret;
}

static void audio_dma_free(struct audio_stream *s)
{
	sa1100_free_dma(s->dma_regs);
	s->dma_regs = 0;
}

#endif

static u_int audio_get_dma_pos(struct audio_stream *s)
{
	struct snd_pcm_substream *substream = s->stream;
	struct snd_pcm_runtime *runtime = substream->runtime;
	unsigned int offset;
	unsigned long flags;
	dma_addr_t addr;
	
	// this must be called w/ interrupts locked out see dma-sa1100.c in the kernel
	spin_lock_irqsave(&s->dma_lock, flags);
#ifdef HH_VERSION	
	sa1100_dma_get_current(s->dmach, NULL, &addr);
#else
	addr = sa1100_get_dma_pos((s)->dma_regs);
#endif
	offset = addr - runtime->dma_addr;
	spin_unlock_irqrestore(&s->dma_lock, flags);
	
	offset = bytes_to_frames(runtime,offset);
	if (offset >= runtime->buffer_size)
		offset = 0;

	return offset;
}

/*
 * this stops the dma and clears the dma ptrs
 */
static void audio_stop_dma(struct audio_stream *s)
{
	unsigned long flags;

	spin_lock_irqsave(&s->dma_lock, flags);	
	s->active = 0;
	s->period = 0;
	/* this stops the dma channel and clears the buffer ptrs */
#ifdef HH_VERSION
	sa1100_dma_flush_all(s->dmach);
#else
	sa1100_clear_dma(s->dma_regs);	
#endif
	spin_unlock_irqrestore(&s->dma_lock, flags);
}

static void audio_process_dma(struct audio_stream *s)
{
	struct snd_pcm_substream *substream = s->stream;
	struct snd_pcm_runtime *runtime;
	unsigned int dma_size;		
	unsigned int offset;
	int ret;
                
	/* we are requested to process synchronization DMA transfer */
	if (s->tx_spin) {
		snd_assert(s->stream_id == SNDRV_PCM_STREAM_PLAYBACK, return);
		/* fill the xmit dma buffers and return */
#ifdef HH_VERSION
		sa1100_dma_set_spin(s->dmach, FORCE_CLOCK_ADDR, FORCE_CLOCK_SIZE);
#else
		while (1) {
			ret = sa1100_start_dma(s->dma_regs, FORCE_CLOCK_ADDR, FORCE_CLOCK_SIZE);
			if (ret)
				return;   
		}
#endif
		return;
	}

	/* must be set here - only valid for running streams, not for forced_clock dma fills  */
	runtime = substream->runtime;
	while (s->active && s->periods < runtime->periods) {
		dma_size = frames_to_bytes(runtime, runtime->period_size);
		if (s->old_offset) {
			/* a little trick, we need resume from old position */
			offset = frames_to_bytes(runtime, s->old_offset - 1);
			s->old_offset = 0;
			s->periods = 0;
			s->period = offset / dma_size;
			offset %= dma_size;
			dma_size = dma_size - offset;
			if (!dma_size)
				continue;		/* special case */
		} else {
			offset = dma_size * s->period;
			snd_assert(dma_size <= DMA_BUF_SIZE, );
		}
#ifdef HH_VERSION
		ret = sa1100_dma_queue_buffer(s->dmach, s, runtime->dma_addr + offset, dma_size);
		if (ret)
			return; //FIXME
#else
		ret = sa1100_start_dma((s)->dma_regs, runtime->dma_addr + offset, dma_size);
		if (ret) {
			printk(KERN_ERR "audio_process_dma: cannot queue DMA buffer (%i)\n", ret);
			return;
		}
#endif

		s->period++;
		s->period %= runtime->periods;
		s->periods++;
	}
}

#ifdef HH_VERSION
static void audio_dma_callback(void *data, int size)
#else
static void audio_dma_callback(void *data)
#endif
{
	struct audio_stream *s = data;
        
	/* 
	 * If we are getting a callback for an active stream then we inform
	 * the PCM middle layer we've finished a period
	 */
 	if (s->active)
		snd_pcm_period_elapsed(s->stream);

	spin_lock(&s->dma_lock);
	if (!s->tx_spin && s->periods > 0)
		s->periods--;
	audio_process_dma(s);
	spin_unlock(&s->dma_lock);
}

/* }}} */

/* {{{ PCM setting */

/* {{{ trigger & timer */

static int snd_sa11xx_uda1341_trigger(struct snd_pcm_substream *substream, int cmd)
{
	struct sa11xx_uda1341 *chip = snd_pcm_substream_chip(substream);
	int stream_id = substream->pstr->stream;
	struct audio_stream *s = &chip->s[stream_id];
	struct audio_stream *s1 = &chip->s[stream_id ^ 1];
	int err = 0;

	/* note local interrupts are already disabled in the midlevel code */
	spin_lock(&s->dma_lock);
	switch (cmd) {
	case SNDRV_PCM_TRIGGER_START:
		/* now we need to make sure a record only stream has a clock */
		if (stream_id == SNDRV_PCM_STREAM_CAPTURE && !s1->active) {
			/* we need to force fill the xmit DMA with zeros */
			s1->tx_spin = 1;
			audio_process_dma(s1);
		}
		/* this case is when you were recording then you turn on a
		 * playback stream so we stop (also clears it) the dma first,
		 * clear the sync flag and then we let it turned on
		 */		
		else {
 			s->tx_spin = 0;
 		}

		/* requested stream startup */
		s->active = 1;
		audio_process_dma(s);
		break;
	case SNDRV_PCM_TRIGGER_STOP:
		/* requested stream shutdown */
		audio_stop_dma(s);
		
		/*
		 * now we need to make sure a record only stream has a clock
		 * so if we're stopping a playback with an active capture
		 * we need to turn the 0 fill dma on for the xmit side
		 */
		if (stream_id == SNDRV_PCM_STREAM_PLAYBACK && s1->active) {
			/* we need to force fill the xmit DMA with zeros */
			s->tx_spin = 1;
			audio_process_dma(s);
		}
		/*
		 * we killed a capture only stream, so we should also kill
		 * the zero fill transmit
		 */
		else {
			if (s1->tx_spin) {
				s1->tx_spin = 0;
				audio_stop_dma(s1);
			}
		}
		
		break;
	case SNDRV_PCM_TRIGGER_SUSPEND:
		s->active = 0;
#ifdef HH_VERSION		
		sa1100_dma_stop(s->dmach);
#else
		//FIXME - DMA API
#endif		
		s->old_offset = audio_get_dma_pos(s) + 1;
#ifdef HH_VERSION		
		sa1100_dma_flush_all(s->dmach);
#else
		//FIXME - DMA API
#endif		
		s->periods = 0;
		break;
	case SNDRV_PCM_TRIGGER_RESUME:
		s->active = 1;
		s->tx_spin = 0;
		audio_process_dma(s);
		if (stream_id == SNDRV_PCM_STREAM_CAPTURE && !s1->active) {
			s1->tx_spin = 1;
			audio_process_dma(s1);
		}
		break;
	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
#ifdef HH_VERSION		
		sa1100_dma_stop(s->dmach);
#else
		//FIXME - DMA API
#endif
		s->active = 0;
		if (stream_id == SNDRV_PCM_STREAM_PLAYBACK) {
			if (s1->active) {
				s->tx_spin = 1;
				s->old_offset = audio_get_dma_pos(s) + 1;
#ifdef HH_VERSION				
				sa1100_dma_flush_all(s->dmach);
#else
				//FIXME - DMA API
#endif				
				audio_process_dma(s);
			}
		} else {
			if (s1->tx_spin) {
				s1->tx_spin = 0;
#ifdef HH_VERSION				
				sa1100_dma_flush_all(s1->dmach);
#else
				//FIXME - DMA API
#endif				
			}
		}
		break;
	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
		s->active = 1;
		if (s->old_offset) {
			s->tx_spin = 0;
			audio_process_dma(s);
			break;
		}
		if (stream_id == SNDRV_PCM_STREAM_CAPTURE && !s1->active) {
			s1->tx_spin = 1;
			audio_process_dma(s1);
		}
#ifdef HH_VERSION		
		sa1100_dma_resume(s->dmach);
#else
		//FIXME - DMA API
#endif
		break;
	default:
		err = -EINVAL;
		break;
	}
	spin_unlock(&s->dma_lock);	
	return err;
}

static int snd_sa11xx_uda1341_prepare(struct snd_pcm_substream *substream)
{
	struct sa11xx_uda1341 *chip = snd_pcm_substream_chip(substream);
	struct snd_pcm_runtime *runtime = substream->runtime;
	struct audio_stream *s = &chip->s[substream->pstr->stream];
        
	/* set requested samplerate */
	sa11xx_uda1341_set_samplerate(chip, runtime->rate);

	/* set requestd format when available */
	/* set FMT here !!! FIXME */

	s->period = 0;
	s->periods = 0;
        
	return 0;
}

static snd_pcm_uframes_t snd_sa11xx_uda1341_pointer(struct snd_pcm_substream *substream)
{
	struct sa11xx_uda1341 *chip = snd_pcm_substream_chip(substream);
	return audio_get_dma_pos(&chip->s[substream->pstr->stream]);
}

/* }}} */

static struct snd_pcm_hardware snd_sa11xx_uda1341_capture =
{
	.info			= (SNDRV_PCM_INFO_INTERLEAVED |
				   SNDRV_PCM_INFO_BLOCK_TRANSFER |
				   SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID |
				   SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME),
	.formats		= SNDRV_PCM_FMTBIT_S16_LE,
	.rates			= (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |\
				   SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 |\
				   SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |\
				   SNDRV_PCM_RATE_KNOT),
	.rate_min		= 8000,
	.rate_max		= 48000,
	.channels_min		= 2,
	.channels_max		= 2,
	.buffer_bytes_max	= 64*1024,
	.period_bytes_min	= 64,
	.period_bytes_max	= DMA_BUF_SIZE,
	.periods_min		= 2,
	.periods_max		= 255,
	.fifo_size		= 0,
};

static struct snd_pcm_hardware snd_sa11xx_uda1341_playback =
{
	.info			= (SNDRV_PCM_INFO_INTERLEAVED |
				   SNDRV_PCM_INFO_BLOCK_TRANSFER |
				   SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID |
				   SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME),
	.formats		= SNDRV_PCM_FMTBIT_S16_LE,
	.rates			= (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |\
                                   SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 |\
				   SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |\
				   SNDRV_PCM_RATE_KNOT),
	.rate_min		= 8000,
	.rate_max		= 48000,
	.channels_min		= 2,
	.channels_max		= 2,
	.buffer_bytes_max	= 64*1024,
	.period_bytes_min	= 64,
	.period_bytes_max	= DMA_BUF_SIZE,
	.periods_min		= 2,
	.periods_max		= 255,
	.fifo_size		= 0,
};

static int snd_card_sa11xx_uda1341_open(struct snd_pcm_substream *substream)
{
	struct sa11xx_uda1341 *chip = snd_pcm_substream_chip(substream);
	struct snd_pcm_runtime *runtime = substream->runtime;
	int stream_id = substream->pstr->stream;
	int err;

	chip->s[stream_id].stream = substream;

	if (stream_id == SNDRV_PCM_STREAM_PLAYBACK)
		runtime->hw = snd_sa11xx_uda1341_playback;
	else
		runtime->hw = snd_sa11xx_uda1341_capture;
	if ((err = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS)) < 0)
		return err;
	if ((err = snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &hw_constraints_rates)) < 0)
		return err;
        
	return 0;
}

static int snd_card_sa11xx_uda1341_close(struct snd_pcm_substream *substream)
{
	struct sa11xx_uda1341 *chip = snd_pcm_substream_chip(substream);

	chip->s[substream->pstr->stream].stream = NULL;
	return 0;
}

/* {{{ HW params & free */

static int snd_sa11xx_uda1341_hw_params(struct snd_pcm_substream *substream,
					struct snd_pcm_hw_params *hw_params)
{
        
	return snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params));
}

static int snd_sa11xx_uda1341_hw_free(struct snd_pcm_substream *substream)
{
	return snd_pcm_lib_free_pages(substream);
}

/* }}} */

static struct snd_pcm_ops snd_card_sa11xx_uda1341_playback_ops = {
	.open			= snd_card_sa11xx_uda1341_open,
	.close			= snd_card_sa11xx_uda1341_close,
	.ioctl			= snd_pcm_lib_ioctl,
	.hw_params	        = snd_sa11xx_uda1341_hw_params,
	.hw_free	        = snd_sa11xx_uda1341_hw_free,
	.prepare		= snd_sa11xx_uda1341_prepare,
	.trigger		= snd_sa11xx_uda1341_trigger,
	.pointer		= snd_sa11xx_uda1341_pointer,
};

static struct snd_pcm_ops snd_card_sa11xx_uda1341_capture_ops = {
	.open			= snd_card_sa11xx_uda1341_open,
	.close			= snd_card_sa11xx_uda1341_close,
	.ioctl			= snd_pcm_lib_ioctl,
	.hw_params	        = snd_sa11xx_uda1341_hw_params,
	.hw_free	        = snd_sa11xx_uda1341_hw_free,
	.prepare		= snd_sa11xx_uda1341_prepare,
	.trigger		= snd_sa11xx_uda1341_trigger,
	.pointer		= snd_sa11xx_uda1341_pointer,
};

static int __init snd_card_sa11xx_uda1341_pcm(struct sa11xx_uda1341 *sa11xx_uda1341, int device)
{
	struct snd_pcm *pcm;
	int err;

	if ((err = snd_pcm_new(sa11xx_uda1341->card, "UDA1341 PCM", device, 1, 1, &pcm)) < 0)
		return err;

	/*
	 * this sets up our initial buffers and sets the dma_type to isa.
	 * isa works but I'm not sure why (or if) it's the right choice
	 * this may be too large, trying it for now
	 */
	snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, 
					      snd_dma_isa_data(),
					      64*1024, 64*1024);

	snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_card_sa11xx_uda1341_playback_ops);
	snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_card_sa11xx_uda1341_capture_ops);
	pcm->private_data = sa11xx_uda1341;
	pcm->info_flags = 0;
	strcpy(pcm->name, "UDA1341 PCM");

	sa11xx_uda1341_audio_init(sa11xx_uda1341);

	/* setup DMA controller */
	audio_dma_request(&sa11xx_uda1341->s[SNDRV_PCM_STREAM_PLAYBACK], audio_dma_callback);
	audio_dma_request(&sa11xx_uda1341->s[SNDRV_PCM_STREAM_CAPTURE], audio_dma_callback);

	sa11xx_uda1341->pcm = pcm;

	return 0;
}

/* }}} */

/* {{{ module init & exit */

#ifdef CONFIG_PM

static int snd_sa11xx_uda1341_suspend(struct platform_device *devptr,
				      pm_message_t state)
{
	struct snd_card *card = platform_get_drvdata(devptr);
	struct sa11xx_uda1341 *chip = card->private_data;

	snd_power_change_state(card, SNDRV_CTL_POWER_D3hot);
	snd_pcm_suspend_all(chip->pcm);
#ifdef HH_VERSION
	sa1100_dma_sleep(chip->s[SNDRV_PCM_STREAM_PLAYBACK].dmach);
	sa1100_dma_sleep(chip->s[SNDRV_PCM_STREAM_CAPTURE].dmach);
#else
	//FIXME
#endif
	l3_command(chip->uda1341, CMD_SUSPEND, NULL);
	sa11xx_uda1341_audio_shutdown(chip);

	return 0;
}

static int snd_sa11xx_uda1341_resume(struct platform_device *devptr)
{
	struct snd_card *card = platform_get_drvdata(devptr);
	struct sa11xx_uda1341 *chip = card->private_data;

	sa11xx_uda1341_audio_init(chip);
	l3_command(chip->uda1341, CMD_RESUME, NULL);
#ifdef HH_VERSION	
	sa1100_dma_wakeup(chip->s[SNDRV_PCM_STREAM_PLAYBACK].dmach);
	sa1100_dma_wakeup(chip->s[SNDRV_PCM_STREAM_CAPTURE].dmach);
#else
	//FIXME
#endif
	snd_power_change_state(card, SNDRV_CTL_POWER_D0);
	return 0;
}
#endif /* COMFIG_PM */

void snd_sa11xx_uda1341_free(struct snd_card *card)
{
	struct sa11xx_uda1341 *chip = card->private_data;

	audio_dma_free(&chip->s[SNDRV_PCM_STREAM_PLAYBACK]);
	audio_dma_free(&chip->s[SNDRV_PCM_STREAM_CAPTURE]);
}

static int __init sa11xx_uda1341_probe(struct platform_device *devptr)
{
	int err;
	struct snd_card *card;
	struct sa11xx_uda1341 *chip;

	/* register the soundcard */
	card = snd_card_new(-1, id, THIS_MODULE, sizeof(struct sa11xx_uda1341));
	if (card == NULL)
		return -ENOMEM;

	chip = card->private_data;
	spin_lock_init(&chip->s[0].dma_lock);
	spin_lock_init(&chip->s[1].dma_lock);

	card->private_free = snd_sa11xx_uda1341_free;
	chip->card = card;
	chip->samplerate = AUDIO_RATE_DEFAULT;

	// mixer
	if ((err = snd_chip_uda1341_mixer_new(card, &chip->uda1341)))
		goto nodev;

	// PCM
	if ((err = snd_card_sa11xx_uda1341_pcm(chip, 0)) < 0)
		goto nodev;
        
	strcpy(card->driver, "UDA1341");
	strcpy(card->shortname, "H3600 UDA1341TS");
	sprintf(card->longname, "Compaq iPAQ H3600 with Philips UDA1341TS");
        
	snd_card_set_dev(card, &devptr->dev);

	if ((err = snd_card_register(card)) == 0) {
		printk( KERN_INFO "iPAQ audio support initialized\n" );
		platform_set_drvdata(devptr, card);
		return 0;
	}
        
 nodev:
	snd_card_free(card);
	return err;
}

static int __devexit sa11xx_uda1341_remove(struct platform_device *devptr)
{
	snd_card_free(platform_get_drvdata(devptr));
	platform_set_drvdata(devptr, NULL);
	return 0;
}

#define SA11XX_UDA1341_DRIVER	"sa11xx_uda1341"

static struct platform_driver sa11xx_uda1341_driver = {
	.probe		= sa11xx_uda1341_probe,
	.remove		= __devexit_p(sa11xx_uda1341_remove),
#ifdef CONFIG_PM
	.suspend	= snd_sa11xx_uda1341_suspend,
	.resume		= snd_sa11xx_uda1341_resume,
#endif
	.driver		= {
		.name	= SA11XX_UDA1341_DRIVER,
	},
};

static int __init sa11xx_uda1341_init(void)
{
	int err;

	if (!machine_is_h3xxx())
		return -ENODEV;
	if ((err = platform_driver_register(&sa11xx_uda1341_driver)) < 0)
		return err;
	device = platform_device_register_simple(SA11XX_UDA1341_DRIVER, -1, NULL, 0);
	if (!IS_ERR(device)) {
		if (platform_get_drvdata(device))
			return 0;
		platform_device_unregister(device);
		err = -ENODEV;
	} else
		err = PTR_ERR(device);
	platform_driver_unregister(&sa11xx_uda1341_driver);
	return err;
}

static void __exit sa11xx_uda1341_exit(void)
{
	platform_device_unregister(device);
	platform_driver_unregister(&sa11xx_uda1341_driver);
}

module_init(sa11xx_uda1341_init);
module_exit(sa11xx_uda1341_exit);

/* }}} */

/*
 * Local variables:
 * indent-tabs-mode: t
 * End:
 */