summaryrefslogtreecommitdiff
path: root/sound/soc/omap/ams-delta.c
blob: b40095a198835dd9f2dc38a28ea40a49b4d84949 (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
/*
 * ams-delta.c  --  SoC audio for Amstrad E3 (Delta) videophone
 *
 * Copyright (C) 2009 Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
 *
 * Initially based on sound/soc/omap/osk5912.x
 * Copyright (C) 2008 Mistral Solutions
 *
 * This program is free software; you can redistribute it and/or
 * modify it under the terms of the GNU General Public License
 * version 2 as published by the Free Software Foundation.
 *
 * 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., 51 Franklin St, Fifth Floor, Boston, MA
 * 02110-1301 USA
 *
 */

#include <linux/gpio.h>
#include <linux/spinlock.h>
#include <linux/tty.h>

#include <sound/soc.h>
#include <sound/jack.h>

#include <asm/mach-types.h>

#include <plat/board-ams-delta.h>
#include <plat/mcbsp.h>

#include "omap-mcbsp.h"
#include "omap-pcm.h"
#include "../codecs/cx20442.h"


/* Board specific DAPM widgets */
static const struct snd_soc_dapm_widget ams_delta_dapm_widgets[] = {
	/* Handset */
	SND_SOC_DAPM_MIC("Mouthpiece", NULL),
	SND_SOC_DAPM_HP("Earpiece", NULL),
	/* Handsfree/Speakerphone */
	SND_SOC_DAPM_MIC("Microphone", NULL),
	SND_SOC_DAPM_SPK("Speaker", NULL),
};

/* How they are connected to codec pins */
static const struct snd_soc_dapm_route ams_delta_audio_map[] = {
	{"TELIN", NULL, "Mouthpiece"},
	{"Earpiece", NULL, "TELOUT"},

	{"MIC", NULL, "Microphone"},
	{"Speaker", NULL, "SPKOUT"},
};

/*
 * Controls, functional after the modem line discipline is activated.
 */

/* Virtual switch: audio input/output constellations */
static const char *ams_delta_audio_mode[] =
	{"Mixed", "Handset", "Handsfree", "Speakerphone"};

/* Selection <-> pin translation */
#define AMS_DELTA_MOUTHPIECE	0
#define AMS_DELTA_EARPIECE	1
#define AMS_DELTA_MICROPHONE	2
#define AMS_DELTA_SPEAKER	3
#define AMS_DELTA_AGC		4

#define AMS_DELTA_MIXED		((1 << AMS_DELTA_EARPIECE) | \
						(1 << AMS_DELTA_MICROPHONE))
#define AMS_DELTA_HANDSET	((1 << AMS_DELTA_MOUTHPIECE) | \
						(1 << AMS_DELTA_EARPIECE))
#define AMS_DELTA_HANDSFREE	((1 << AMS_DELTA_MICROPHONE) | \
						(1 << AMS_DELTA_SPEAKER))
#define AMS_DELTA_SPEAKERPHONE	(AMS_DELTA_HANDSFREE | (1 << AMS_DELTA_AGC))

static const unsigned short ams_delta_audio_mode_pins[] = {
	AMS_DELTA_MIXED,
	AMS_DELTA_HANDSET,
	AMS_DELTA_HANDSFREE,
	AMS_DELTA_SPEAKERPHONE,
};

static unsigned short ams_delta_audio_agc;

static int ams_delta_set_audio_mode(struct snd_kcontrol *kcontrol,
					struct snd_ctl_elem_value *ucontrol)
{
	struct snd_soc_codec *codec =  snd_kcontrol_chip(kcontrol);
	struct snd_soc_dapm_context *dapm = &codec->dapm;
	struct soc_enum *control = (struct soc_enum *)kcontrol->private_value;
	unsigned short pins;
	int pin, changed = 0;

	/* Refuse any mode changes if we are not able to control the codec. */
	if (!codec->hw_write)
		return -EUNATCH;

	if (ucontrol->value.enumerated.item[0] >= control->max)
		return -EINVAL;

	mutex_lock(&codec->mutex);

	/* Translate selection to bitmap */
	pins = ams_delta_audio_mode_pins[ucontrol->value.enumerated.item[0]];

	/* Setup pins after corresponding bits if changed */
	pin = !!(pins & (1 << AMS_DELTA_MOUTHPIECE));
	if (pin != snd_soc_dapm_get_pin_status(dapm, "Mouthpiece")) {
		changed = 1;
		if (pin)
			snd_soc_dapm_enable_pin(dapm, "Mouthpiece");
		else
			snd_soc_dapm_disable_pin(dapm, "Mouthpiece");
	}
	pin = !!(pins & (1 << AMS_DELTA_EARPIECE));
	if (pin != snd_soc_dapm_get_pin_status(dapm, "Earpiece")) {
		changed = 1;
		if (pin)
			snd_soc_dapm_enable_pin(dapm, "Earpiece");
		else
			snd_soc_dapm_disable_pin(dapm, "Earpiece");
	}
	pin = !!(pins & (1 << AMS_DELTA_MICROPHONE));
	if (pin != snd_soc_dapm_get_pin_status(dapm, "Microphone")) {
		changed = 1;
		if (pin)
			snd_soc_dapm_enable_pin(dapm, "Microphone");
		else
			snd_soc_dapm_disable_pin(dapm, "Microphone");
	}
	pin = !!(pins & (1 << AMS_DELTA_SPEAKER));
	if (pin != snd_soc_dapm_get_pin_status(dapm, "Speaker")) {
		changed = 1;
		if (pin)
			snd_soc_dapm_enable_pin(dapm, "Speaker");
		else
			snd_soc_dapm_disable_pin(dapm, "Speaker");
	}
	pin = !!(pins & (1 << AMS_DELTA_AGC));
	if (pin != ams_delta_audio_agc) {
		ams_delta_audio_agc = pin;
		changed = 1;
		if (pin)
			snd_soc_dapm_enable_pin(dapm, "AGCIN");
		else
			snd_soc_dapm_disable_pin(dapm, "AGCIN");
	}
	if (changed)
		snd_soc_dapm_sync(dapm);

	mutex_unlock(&codec->mutex);

	return changed;
}

static int ams_delta_get_audio_mode(struct snd_kcontrol *kcontrol,
					struct snd_ctl_elem_value *ucontrol)
{
	struct snd_soc_codec *codec =  snd_kcontrol_chip(kcontrol);
	struct snd_soc_dapm_context *dapm = &codec->dapm;
	unsigned short pins, mode;

	pins = ((snd_soc_dapm_get_pin_status(dapm, "Mouthpiece") <<
							AMS_DELTA_MOUTHPIECE) |
			(snd_soc_dapm_get_pin_status(dapm, "Earpiece") <<
							AMS_DELTA_EARPIECE));
	if (pins)
		pins |= (snd_soc_dapm_get_pin_status(dapm, "Microphone") <<
							AMS_DELTA_MICROPHONE);
	else
		pins = ((snd_soc_dapm_get_pin_status(dapm, "Microphone") <<
							AMS_DELTA_MICROPHONE) |
			(snd_soc_dapm_get_pin_status(dapm, "Speaker") <<
							AMS_DELTA_SPEAKER) |
			(ams_delta_audio_agc << AMS_DELTA_AGC));

	for (mode = 0; mode < ARRAY_SIZE(ams_delta_audio_mode); mode++)
		if (pins == ams_delta_audio_mode_pins[mode])
			break;

	if (mode >= ARRAY_SIZE(ams_delta_audio_mode))
		return -EINVAL;

	ucontrol->value.enumerated.item[0] = mode;

	return 0;
}

static const struct soc_enum ams_delta_audio_enum[] = {
	SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(ams_delta_audio_mode),
						ams_delta_audio_mode),
};

static const struct snd_kcontrol_new ams_delta_audio_controls[] = {
	SOC_ENUM_EXT("Audio Mode", ams_delta_audio_enum[0],
			ams_delta_get_audio_mode, ams_delta_set_audio_mode),
};

/* Hook switch */
static struct snd_soc_jack ams_delta_hook_switch;
static struct snd_soc_jack_gpio ams_delta_hook_switch_gpios[] = {
	{
		.gpio = 4,
		.name = "hook_switch",
		.report = SND_JACK_HEADSET,
		.invert = 1,
		.debounce_time = 150,
	}
};

/* After we are able to control the codec over the modem,
 * the hook switch can be used for dynamic DAPM reconfiguration. */
static struct snd_soc_jack_pin ams_delta_hook_switch_pins[] = {
	/* Handset */
	{
		.pin = "Mouthpiece",
		.mask = SND_JACK_MICROPHONE,
	},
	{
		.pin = "Earpiece",
		.mask = SND_JACK_HEADPHONE,
	},
	/* Handsfree */
	{
		.pin = "Microphone",
		.mask = SND_JACK_MICROPHONE,
		.invert = 1,
	},
	{
		.pin = "Speaker",
		.mask = SND_JACK_HEADPHONE,
		.invert = 1,
	},
};


/*
 * Modem line discipline, required for making above controls functional.
 * Activated from userspace with ldattach, possibly invoked from udev rule.
 */

/* To actually apply any modem controlled configuration changes to the codec,
 * we must connect codec DAI pins to the modem for a moment.  Be careful not
 * to interfere with our digital mute function that shares the same hardware. */
static struct timer_list cx81801_timer;
static bool cx81801_cmd_pending;
static bool ams_delta_muted;
static DEFINE_SPINLOCK(ams_delta_lock);

static void cx81801_timeout(unsigned long data)
{
	int muted;

	spin_lock(&ams_delta_lock);
	cx81801_cmd_pending = 0;
	muted = ams_delta_muted;
	spin_unlock(&ams_delta_lock);

	/* Reconnect the codec DAI back from the modem to the CPU DAI
	 * only if digital mute still off */
	if (!muted)
		ams_delta_latch2_write(AMS_DELTA_LATCH2_MODEM_CODEC, 0);
}

/*
 * Used for passing a codec structure pointer
 * from the board initialization code to the tty line discipline.
 */
static struct snd_soc_codec *cx20442_codec;

/* Line discipline .open() */
static int cx81801_open(struct tty_struct *tty)
{
	int ret;

	if (!cx20442_codec)
		return -ENODEV;

	/*
	 * Pass the codec structure pointer for use by other ldisc callbacks,
	 * both the card and the codec specific parts.
	 */
	tty->disc_data = cx20442_codec;

	ret = v253_ops.open(tty);

	if (ret < 0)
		tty->disc_data = NULL;

	return ret;
}

/* Line discipline .close() */
static void cx81801_close(struct tty_struct *tty)
{
	struct snd_soc_codec *codec = tty->disc_data;
	struct snd_soc_dapm_context *dapm = &codec->dapm;

	del_timer_sync(&cx81801_timer);

	/* Prevent the hook switch from further changing the DAPM pins */
	INIT_LIST_HEAD(&ams_delta_hook_switch.pins);

	if (!codec)
		return;

	v253_ops.close(tty);

	/* Revert back to default audio input/output constellation */
	snd_soc_dapm_disable_pin(dapm, "Mouthpiece");
	snd_soc_dapm_enable_pin(dapm, "Earpiece");
	snd_soc_dapm_enable_pin(dapm, "Microphone");
	snd_soc_dapm_disable_pin(dapm, "Speaker");
	snd_soc_dapm_disable_pin(dapm, "AGCIN");
	snd_soc_dapm_sync(dapm);
}

/* Line discipline .hangup() */
static int cx81801_hangup(struct tty_struct *tty)
{
	cx81801_close(tty);
	return 0;
}

/* Line discipline .recieve_buf() */
static void cx81801_receive(struct tty_struct *tty,
				const unsigned char *cp, char *fp, int count)
{
	struct snd_soc_codec *codec = tty->disc_data;
	const unsigned char *c;
	int apply, ret;

	if (!codec)
		return;

	if (!codec->hw_write) {
		/* First modem response, complete setup procedure */

		/* Initialize timer used for config pulse generation */
		setup_timer(&cx81801_timer, cx81801_timeout, 0);

		v253_ops.receive_buf(tty, cp, fp, count);

		/* Link hook switch to DAPM pins */
		ret = snd_soc_jack_add_pins(&ams_delta_hook_switch,
					ARRAY_SIZE(ams_delta_hook_switch_pins),
					ams_delta_hook_switch_pins);
		if (ret)
			dev_warn(codec->dev,
				"Failed to link hook switch to DAPM pins, "
				"will continue with hook switch unlinked.\n");

		return;
	}

	v253_ops.receive_buf(tty, cp, fp, count);

	for (c = &cp[count - 1]; c >= cp; c--) {
		if (*c != '\r')
			continue;
		/* Complete modem response received, apply config to codec */

		spin_lock_bh(&ams_delta_lock);
		mod_timer(&cx81801_timer, jiffies + msecs_to_jiffies(150));
		apply = !ams_delta_muted && !cx81801_cmd_pending;
		cx81801_cmd_pending = 1;
		spin_unlock_bh(&ams_delta_lock);

		/* Apply config pulse by connecting the codec to the modem
		 * if not already done */
		if (apply)
			ams_delta_latch2_write(AMS_DELTA_LATCH2_MODEM_CODEC,
						AMS_DELTA_LATCH2_MODEM_CODEC);
		break;
	}
}

/* Line discipline .write_wakeup() */
static void cx81801_wakeup(struct tty_struct *tty)
{
	v253_ops.write_wakeup(tty);
}

static struct tty_ldisc_ops cx81801_ops = {
	.magic = TTY_LDISC_MAGIC,
	.name = "cx81801",
	.owner = THIS_MODULE,
	.open = cx81801_open,
	.close = cx81801_close,
	.hangup = cx81801_hangup,
	.receive_buf = cx81801_receive,
	.write_wakeup = cx81801_wakeup,
};


/*
 * Even if not very useful, the sound card can still work without any of the
 * above functonality activated.  You can still control its audio input/output
 * constellation and speakerphone gain from userspace by issuing AT commands
 * over the modem port.
 */

static int ams_delta_hw_params(struct snd_pcm_substream *substream,
			 struct snd_pcm_hw_params *params)
{
	struct snd_soc_pcm_runtime *rtd = substream->private_data;

	/* Set cpu DAI configuration */
	return snd_soc_dai_set_fmt(rtd->cpu_dai,
				   SND_SOC_DAIFMT_DSP_A |
				   SND_SOC_DAIFMT_NB_NF |
				   SND_SOC_DAIFMT_CBM_CFM);
}

static struct snd_soc_ops ams_delta_ops = {
	.hw_params = ams_delta_hw_params,
};


/* Board specific codec bias level control */
static int ams_delta_set_bias_level(struct snd_soc_card *card,
				    struct snd_soc_dapm_context *dapm,
				    enum snd_soc_bias_level level)
{
	struct snd_soc_codec *codec = card->rtd->codec;

	switch (level) {
	case SND_SOC_BIAS_ON:
	case SND_SOC_BIAS_PREPARE:
	case SND_SOC_BIAS_STANDBY:
		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF)
			ams_delta_latch2_write(AMS_DELTA_LATCH2_MODEM_NRESET,
						AMS_DELTA_LATCH2_MODEM_NRESET);
		break;
	case SND_SOC_BIAS_OFF:
		if (codec->dapm.bias_level != SND_SOC_BIAS_OFF)
			ams_delta_latch2_write(AMS_DELTA_LATCH2_MODEM_NRESET,
						0);
	}
	codec->dapm.bias_level = level;

	return 0;
}

/* Digital mute implemented using modem/CPU multiplexer.
 * Shares hardware with codec config pulse generation */
static bool ams_delta_muted = 1;

static int ams_delta_digital_mute(struct snd_soc_dai *dai, int mute)
{
	int apply;

	if (ams_delta_muted == mute)
		return 0;

	spin_lock_bh(&ams_delta_lock);
	ams_delta_muted = mute;
	apply = !cx81801_cmd_pending;
	spin_unlock_bh(&ams_delta_lock);

	if (apply)
		ams_delta_latch2_write(AMS_DELTA_LATCH2_MODEM_CODEC,
				mute ? AMS_DELTA_LATCH2_MODEM_CODEC : 0);
	return 0;
}

/* Our codec DAI probably doesn't have its own .ops structure */
static struct snd_soc_dai_ops ams_delta_dai_ops = {
	.digital_mute = ams_delta_digital_mute,
};

/* Will be used if the codec ever has its own digital_mute function */
static int ams_delta_startup(struct snd_pcm_substream *substream)
{
	return ams_delta_digital_mute(NULL, 0);
}

static void ams_delta_shutdown(struct snd_pcm_substream *substream)
{
	ams_delta_digital_mute(NULL, 1);
}


/*
 * Card initialization
 */

static int ams_delta_cx20442_init(struct snd_soc_pcm_runtime *rtd)
{
	struct snd_soc_codec *codec = rtd->codec;
	struct snd_soc_dapm_context *dapm = &codec->dapm;
	struct snd_soc_dai *codec_dai = rtd->codec_dai;
	struct snd_soc_card *card = rtd->card;
	int ret;
	/* Codec is ready, now add/activate board specific controls */

	/* Store a pointer to the codec structure for tty ldisc use */
	cx20442_codec = codec;

	/* Set up digital mute if not provided by the codec */
	if (!codec_dai->driver->ops) {
		codec_dai->driver->ops = &ams_delta_dai_ops;
	} else {
		ams_delta_ops.startup = ams_delta_startup;
		ams_delta_ops.shutdown = ams_delta_shutdown;
	}

	/* Set codec bias level */
	ams_delta_set_bias_level(card, SND_SOC_BIAS_STANDBY);

	/* Add hook switch - can be used to control the codec from userspace
	 * even if line discipline fails */
	ret = snd_soc_jack_new(rtd->codec, "hook_switch",
				SND_JACK_HEADSET, &ams_delta_hook_switch);
	if (ret)
		dev_warn(card->dev,
				"Failed to allocate resources for hook switch, "
				"will continue without one.\n");
	else {
		ret = snd_soc_jack_add_gpios(&ams_delta_hook_switch,
					ARRAY_SIZE(ams_delta_hook_switch_gpios),
					ams_delta_hook_switch_gpios);
		if (ret)
			dev_warn(card->dev,
				"Failed to set up hook switch GPIO line, "
				"will continue with hook switch inactive.\n");
	}

	/* Register optional line discipline for over the modem control */
	ret = tty_register_ldisc(N_V253, &cx81801_ops);
	if (ret) {
		dev_warn(card->dev,
				"Failed to register line discipline, "
				"will continue without any controls.\n");
		return 0;
	}

	/* Add board specific DAPM widgets and routes */
	ret = snd_soc_dapm_new_controls(dapm, ams_delta_dapm_widgets,
					ARRAY_SIZE(ams_delta_dapm_widgets));
	if (ret) {
		dev_warn(card->dev,
				"Failed to register DAPM controls, "
				"will continue without any.\n");
		return 0;
	}

	ret = snd_soc_dapm_add_routes(dapm, ams_delta_audio_map,
					ARRAY_SIZE(ams_delta_audio_map));
	if (ret) {
		dev_warn(card->dev,
				"Failed to set up DAPM routes, "
				"will continue with codec default map.\n");
		return 0;
	}

	/* Set up initial pin constellation */
	snd_soc_dapm_disable_pin(dapm, "Mouthpiece");
	snd_soc_dapm_enable_pin(dapm, "Earpiece");
	snd_soc_dapm_enable_pin(dapm, "Microphone");
	snd_soc_dapm_disable_pin(dapm, "Speaker");
	snd_soc_dapm_disable_pin(dapm, "AGCIN");
	snd_soc_dapm_disable_pin(dapm, "AGCOUT");
	snd_soc_dapm_sync(dapm);

	/* Add virtual switch */
	ret = snd_soc_add_controls(codec, ams_delta_audio_controls,
					ARRAY_SIZE(ams_delta_audio_controls));
	if (ret)
		dev_warn(card->dev,
				"Failed to register audio mode control, "
				"will continue without it.\n");

	return 0;
}

/* DAI glue - connects codec <--> CPU */
static struct snd_soc_dai_link ams_delta_dai_link = {
	.name = "CX20442",
	.stream_name = "CX20442",
	.cpu_dai_name ="omap-mcbsp-dai.0",
	.codec_dai_name = "cx20442-voice",
	.init = ams_delta_cx20442_init,
	.platform_name = "omap-pcm-audio",
	.codec_name = "cx20442-codec",
	.ops = &ams_delta_ops,
};

/* Audio card driver */
static struct snd_soc_card ams_delta_audio_card = {
	.name = "AMS_DELTA",
	.dai_link = &ams_delta_dai_link,
	.num_links = 1,
	.set_bias_level = ams_delta_set_bias_level,
};

/* Module init/exit */
static struct platform_device *ams_delta_audio_platform_device;
static struct platform_device *cx20442_platform_device;

static int __init ams_delta_module_init(void)
{
	int ret;

	if (!(machine_is_ams_delta()))
		return -ENODEV;

	ams_delta_audio_platform_device =
			platform_device_alloc("soc-audio", -1);
	if (!ams_delta_audio_platform_device)
		return -ENOMEM;

	platform_set_drvdata(ams_delta_audio_platform_device,
				&ams_delta_audio_card);

	ret = platform_device_add(ams_delta_audio_platform_device);
	if (ret)
		goto err;

	/*
	 * Codec platform device could be registered from elsewhere (board?),
	 * but I do it here as it makes sense only if used with the card.
	 */
	cx20442_platform_device =
		platform_device_register_simple("cx20442-codec", -1, NULL, 0);
	return 0;
err:
	platform_device_put(ams_delta_audio_platform_device);
	return ret;
}
module_init(ams_delta_module_init);

static void __exit ams_delta_module_exit(void)
{
	if (tty_unregister_ldisc(N_V253) != 0)
		dev_warn(&ams_delta_audio_platform_device->dev,
			"failed to unregister V253 line discipline\n");

	snd_soc_jack_free_gpios(&ams_delta_hook_switch,
			ARRAY_SIZE(ams_delta_hook_switch_gpios),
			ams_delta_hook_switch_gpios);

	/* Keep modem power on */
	ams_delta_set_bias_level(&ams_delta_audio_card, SND_SOC_BIAS_STANDBY);

	platform_device_unregister(cx20442_platform_device);
	platform_device_unregister(ams_delta_audio_platform_device);
}
module_exit(ams_delta_module_exit);

MODULE_AUTHOR("Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>");
MODULE_DESCRIPTION("ALSA SoC driver for Amstrad E3 (Delta) videophone");
MODULE_LICENSE("GPL");