x86: Fix S4 regression
[linux-btrfs-devel.git] / sound / soc / omap / ams-delta.c
blob0aa475f92efaac9f01ad36d7fc2d334acdc1d0df
1 /*
2 * ams-delta.c -- SoC audio for Amstrad E3 (Delta) videophone
4 * Copyright (C) 2009 Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
6 * Initially based on sound/soc/omap/osk5912.x
7 * Copyright (C) 2008 Mistral Solutions
9 * This program is free software; you can redistribute it and/or
10 * modify it under the terms of the GNU General Public License
11 * version 2 as published by the Free Software Foundation.
13 * This program is distributed in the hope that it will be useful, but
14 * WITHOUT ANY WARRANTY; without even the implied warranty of
15 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
16 * General Public License for more details.
18 * You should have received a copy of the GNU General Public License
19 * along with this program; if not, write to the Free Software
20 * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
21 * 02110-1301 USA
25 #include <linux/gpio.h>
26 #include <linux/spinlock.h>
27 #include <linux/tty.h>
29 #include <sound/soc.h>
30 #include <sound/jack.h>
32 #include <asm/mach-types.h>
34 #include <plat/board-ams-delta.h>
35 #include <plat/mcbsp.h>
37 #include "omap-mcbsp.h"
38 #include "omap-pcm.h"
39 #include "../codecs/cx20442.h"
42 /* Board specific DAPM widgets */
43 static const struct snd_soc_dapm_widget ams_delta_dapm_widgets[] = {
44 /* Handset */
45 SND_SOC_DAPM_MIC("Mouthpiece", NULL),
46 SND_SOC_DAPM_HP("Earpiece", NULL),
47 /* Handsfree/Speakerphone */
48 SND_SOC_DAPM_MIC("Microphone", NULL),
49 SND_SOC_DAPM_SPK("Speaker", NULL),
52 /* How they are connected to codec pins */
53 static const struct snd_soc_dapm_route ams_delta_audio_map[] = {
54 {"TELIN", NULL, "Mouthpiece"},
55 {"Earpiece", NULL, "TELOUT"},
57 {"MIC", NULL, "Microphone"},
58 {"Speaker", NULL, "SPKOUT"},
62 * Controls, functional after the modem line discipline is activated.
65 /* Virtual switch: audio input/output constellations */
66 static const char *ams_delta_audio_mode[] =
67 {"Mixed", "Handset", "Handsfree", "Speakerphone"};
69 /* Selection <-> pin translation */
70 #define AMS_DELTA_MOUTHPIECE 0
71 #define AMS_DELTA_EARPIECE 1
72 #define AMS_DELTA_MICROPHONE 2
73 #define AMS_DELTA_SPEAKER 3
74 #define AMS_DELTA_AGC 4
76 #define AMS_DELTA_MIXED ((1 << AMS_DELTA_EARPIECE) | \
77 (1 << AMS_DELTA_MICROPHONE))
78 #define AMS_DELTA_HANDSET ((1 << AMS_DELTA_MOUTHPIECE) | \
79 (1 << AMS_DELTA_EARPIECE))
80 #define AMS_DELTA_HANDSFREE ((1 << AMS_DELTA_MICROPHONE) | \
81 (1 << AMS_DELTA_SPEAKER))
82 #define AMS_DELTA_SPEAKERPHONE (AMS_DELTA_HANDSFREE | (1 << AMS_DELTA_AGC))
84 static const unsigned short ams_delta_audio_mode_pins[] = {
85 AMS_DELTA_MIXED,
86 AMS_DELTA_HANDSET,
87 AMS_DELTA_HANDSFREE,
88 AMS_DELTA_SPEAKERPHONE,
91 static unsigned short ams_delta_audio_agc;
93 static int ams_delta_set_audio_mode(struct snd_kcontrol *kcontrol,
94 struct snd_ctl_elem_value *ucontrol)
96 struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
97 struct snd_soc_dapm_context *dapm = &codec->dapm;
98 struct soc_enum *control = (struct soc_enum *)kcontrol->private_value;
99 unsigned short pins;
100 int pin, changed = 0;
102 /* Refuse any mode changes if we are not able to control the codec. */
103 if (!codec->hw_write)
104 return -EUNATCH;
106 if (ucontrol->value.enumerated.item[0] >= control->max)
107 return -EINVAL;
109 mutex_lock(&codec->mutex);
111 /* Translate selection to bitmap */
112 pins = ams_delta_audio_mode_pins[ucontrol->value.enumerated.item[0]];
114 /* Setup pins after corresponding bits if changed */
115 pin = !!(pins & (1 << AMS_DELTA_MOUTHPIECE));
116 if (pin != snd_soc_dapm_get_pin_status(dapm, "Mouthpiece")) {
117 changed = 1;
118 if (pin)
119 snd_soc_dapm_enable_pin(dapm, "Mouthpiece");
120 else
121 snd_soc_dapm_disable_pin(dapm, "Mouthpiece");
123 pin = !!(pins & (1 << AMS_DELTA_EARPIECE));
124 if (pin != snd_soc_dapm_get_pin_status(dapm, "Earpiece")) {
125 changed = 1;
126 if (pin)
127 snd_soc_dapm_enable_pin(dapm, "Earpiece");
128 else
129 snd_soc_dapm_disable_pin(dapm, "Earpiece");
131 pin = !!(pins & (1 << AMS_DELTA_MICROPHONE));
132 if (pin != snd_soc_dapm_get_pin_status(dapm, "Microphone")) {
133 changed = 1;
134 if (pin)
135 snd_soc_dapm_enable_pin(dapm, "Microphone");
136 else
137 snd_soc_dapm_disable_pin(dapm, "Microphone");
139 pin = !!(pins & (1 << AMS_DELTA_SPEAKER));
140 if (pin != snd_soc_dapm_get_pin_status(dapm, "Speaker")) {
141 changed = 1;
142 if (pin)
143 snd_soc_dapm_enable_pin(dapm, "Speaker");
144 else
145 snd_soc_dapm_disable_pin(dapm, "Speaker");
147 pin = !!(pins & (1 << AMS_DELTA_AGC));
148 if (pin != ams_delta_audio_agc) {
149 ams_delta_audio_agc = pin;
150 changed = 1;
151 if (pin)
152 snd_soc_dapm_enable_pin(dapm, "AGCIN");
153 else
154 snd_soc_dapm_disable_pin(dapm, "AGCIN");
156 if (changed)
157 snd_soc_dapm_sync(dapm);
159 mutex_unlock(&codec->mutex);
161 return changed;
164 static int ams_delta_get_audio_mode(struct snd_kcontrol *kcontrol,
165 struct snd_ctl_elem_value *ucontrol)
167 struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
168 struct snd_soc_dapm_context *dapm = &codec->dapm;
169 unsigned short pins, mode;
171 pins = ((snd_soc_dapm_get_pin_status(dapm, "Mouthpiece") <<
172 AMS_DELTA_MOUTHPIECE) |
173 (snd_soc_dapm_get_pin_status(dapm, "Earpiece") <<
174 AMS_DELTA_EARPIECE));
175 if (pins)
176 pins |= (snd_soc_dapm_get_pin_status(dapm, "Microphone") <<
177 AMS_DELTA_MICROPHONE);
178 else
179 pins = ((snd_soc_dapm_get_pin_status(dapm, "Microphone") <<
180 AMS_DELTA_MICROPHONE) |
181 (snd_soc_dapm_get_pin_status(dapm, "Speaker") <<
182 AMS_DELTA_SPEAKER) |
183 (ams_delta_audio_agc << AMS_DELTA_AGC));
185 for (mode = 0; mode < ARRAY_SIZE(ams_delta_audio_mode); mode++)
186 if (pins == ams_delta_audio_mode_pins[mode])
187 break;
189 if (mode >= ARRAY_SIZE(ams_delta_audio_mode))
190 return -EINVAL;
192 ucontrol->value.enumerated.item[0] = mode;
194 return 0;
197 static const struct soc_enum ams_delta_audio_enum[] = {
198 SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(ams_delta_audio_mode),
199 ams_delta_audio_mode),
202 static const struct snd_kcontrol_new ams_delta_audio_controls[] = {
203 SOC_ENUM_EXT("Audio Mode", ams_delta_audio_enum[0],
204 ams_delta_get_audio_mode, ams_delta_set_audio_mode),
207 /* Hook switch */
208 static struct snd_soc_jack ams_delta_hook_switch;
209 static struct snd_soc_jack_gpio ams_delta_hook_switch_gpios[] = {
211 .gpio = 4,
212 .name = "hook_switch",
213 .report = SND_JACK_HEADSET,
214 .invert = 1,
215 .debounce_time = 150,
219 /* After we are able to control the codec over the modem,
220 * the hook switch can be used for dynamic DAPM reconfiguration. */
221 static struct snd_soc_jack_pin ams_delta_hook_switch_pins[] = {
222 /* Handset */
224 .pin = "Mouthpiece",
225 .mask = SND_JACK_MICROPHONE,
228 .pin = "Earpiece",
229 .mask = SND_JACK_HEADPHONE,
231 /* Handsfree */
233 .pin = "Microphone",
234 .mask = SND_JACK_MICROPHONE,
235 .invert = 1,
238 .pin = "Speaker",
239 .mask = SND_JACK_HEADPHONE,
240 .invert = 1,
246 * Modem line discipline, required for making above controls functional.
247 * Activated from userspace with ldattach, possibly invoked from udev rule.
250 /* To actually apply any modem controlled configuration changes to the codec,
251 * we must connect codec DAI pins to the modem for a moment. Be careful not
252 * to interfere with our digital mute function that shares the same hardware. */
253 static struct timer_list cx81801_timer;
254 static bool cx81801_cmd_pending;
255 static bool ams_delta_muted;
256 static DEFINE_SPINLOCK(ams_delta_lock);
258 static void cx81801_timeout(unsigned long data)
260 int muted;
262 spin_lock(&ams_delta_lock);
263 cx81801_cmd_pending = 0;
264 muted = ams_delta_muted;
265 spin_unlock(&ams_delta_lock);
267 /* Reconnect the codec DAI back from the modem to the CPU DAI
268 * only if digital mute still off */
269 if (!muted)
270 ams_delta_latch2_write(AMS_DELTA_LATCH2_MODEM_CODEC, 0);
274 * Used for passing a codec structure pointer
275 * from the board initialization code to the tty line discipline.
277 static struct snd_soc_codec *cx20442_codec;
279 /* Line discipline .open() */
280 static int cx81801_open(struct tty_struct *tty)
282 int ret;
284 if (!cx20442_codec)
285 return -ENODEV;
288 * Pass the codec structure pointer for use by other ldisc callbacks,
289 * both the card and the codec specific parts.
291 tty->disc_data = cx20442_codec;
293 ret = v253_ops.open(tty);
295 if (ret < 0)
296 tty->disc_data = NULL;
298 return ret;
301 /* Line discipline .close() */
302 static void cx81801_close(struct tty_struct *tty)
304 struct snd_soc_codec *codec = tty->disc_data;
305 struct snd_soc_dapm_context *dapm = &codec->dapm;
307 del_timer_sync(&cx81801_timer);
309 /* Prevent the hook switch from further changing the DAPM pins */
310 INIT_LIST_HEAD(&ams_delta_hook_switch.pins);
312 if (!codec)
313 return;
315 v253_ops.close(tty);
317 /* Revert back to default audio input/output constellation */
318 snd_soc_dapm_disable_pin(dapm, "Mouthpiece");
319 snd_soc_dapm_enable_pin(dapm, "Earpiece");
320 snd_soc_dapm_enable_pin(dapm, "Microphone");
321 snd_soc_dapm_disable_pin(dapm, "Speaker");
322 snd_soc_dapm_disable_pin(dapm, "AGCIN");
323 snd_soc_dapm_sync(dapm);
326 /* Line discipline .hangup() */
327 static int cx81801_hangup(struct tty_struct *tty)
329 cx81801_close(tty);
330 return 0;
333 /* Line discipline .receive_buf() */
334 static void cx81801_receive(struct tty_struct *tty,
335 const unsigned char *cp, char *fp, int count)
337 struct snd_soc_codec *codec = tty->disc_data;
338 const unsigned char *c;
339 int apply, ret;
341 if (!codec)
342 return;
344 if (!codec->hw_write) {
345 /* First modem response, complete setup procedure */
347 /* Initialize timer used for config pulse generation */
348 setup_timer(&cx81801_timer, cx81801_timeout, 0);
350 v253_ops.receive_buf(tty, cp, fp, count);
352 /* Link hook switch to DAPM pins */
353 ret = snd_soc_jack_add_pins(&ams_delta_hook_switch,
354 ARRAY_SIZE(ams_delta_hook_switch_pins),
355 ams_delta_hook_switch_pins);
356 if (ret)
357 dev_warn(codec->dev,
358 "Failed to link hook switch to DAPM pins, "
359 "will continue with hook switch unlinked.\n");
361 return;
364 v253_ops.receive_buf(tty, cp, fp, count);
366 for (c = &cp[count - 1]; c >= cp; c--) {
367 if (*c != '\r')
368 continue;
369 /* Complete modem response received, apply config to codec */
371 spin_lock_bh(&ams_delta_lock);
372 mod_timer(&cx81801_timer, jiffies + msecs_to_jiffies(150));
373 apply = !ams_delta_muted && !cx81801_cmd_pending;
374 cx81801_cmd_pending = 1;
375 spin_unlock_bh(&ams_delta_lock);
377 /* Apply config pulse by connecting the codec to the modem
378 * if not already done */
379 if (apply)
380 ams_delta_latch2_write(AMS_DELTA_LATCH2_MODEM_CODEC,
381 AMS_DELTA_LATCH2_MODEM_CODEC);
382 break;
386 /* Line discipline .write_wakeup() */
387 static void cx81801_wakeup(struct tty_struct *tty)
389 v253_ops.write_wakeup(tty);
392 static struct tty_ldisc_ops cx81801_ops = {
393 .magic = TTY_LDISC_MAGIC,
394 .name = "cx81801",
395 .owner = THIS_MODULE,
396 .open = cx81801_open,
397 .close = cx81801_close,
398 .hangup = cx81801_hangup,
399 .receive_buf = cx81801_receive,
400 .write_wakeup = cx81801_wakeup,
405 * Even if not very useful, the sound card can still work without any of the
406 * above functonality activated. You can still control its audio input/output
407 * constellation and speakerphone gain from userspace by issuing AT commands
408 * over the modem port.
411 static int ams_delta_hw_params(struct snd_pcm_substream *substream,
412 struct snd_pcm_hw_params *params)
414 struct snd_soc_pcm_runtime *rtd = substream->private_data;
416 /* Set cpu DAI configuration */
417 return snd_soc_dai_set_fmt(rtd->cpu_dai,
418 SND_SOC_DAIFMT_DSP_A |
419 SND_SOC_DAIFMT_NB_NF |
420 SND_SOC_DAIFMT_CBM_CFM);
423 static struct snd_soc_ops ams_delta_ops = {
424 .hw_params = ams_delta_hw_params,
428 /* Board specific codec bias level control */
429 static int ams_delta_set_bias_level(struct snd_soc_card *card,
430 struct snd_soc_dapm_context *dapm,
431 enum snd_soc_bias_level level)
433 struct snd_soc_codec *codec = card->rtd->codec;
435 switch (level) {
436 case SND_SOC_BIAS_ON:
437 case SND_SOC_BIAS_PREPARE:
438 case SND_SOC_BIAS_STANDBY:
439 if (codec->dapm.bias_level == SND_SOC_BIAS_OFF)
440 ams_delta_latch2_write(AMS_DELTA_LATCH2_MODEM_NRESET,
441 AMS_DELTA_LATCH2_MODEM_NRESET);
442 break;
443 case SND_SOC_BIAS_OFF:
444 if (codec->dapm.bias_level != SND_SOC_BIAS_OFF)
445 ams_delta_latch2_write(AMS_DELTA_LATCH2_MODEM_NRESET,
448 codec->dapm.bias_level = level;
450 return 0;
453 /* Digital mute implemented using modem/CPU multiplexer.
454 * Shares hardware with codec config pulse generation */
455 static bool ams_delta_muted = 1;
457 static int ams_delta_digital_mute(struct snd_soc_dai *dai, int mute)
459 int apply;
461 if (ams_delta_muted == mute)
462 return 0;
464 spin_lock_bh(&ams_delta_lock);
465 ams_delta_muted = mute;
466 apply = !cx81801_cmd_pending;
467 spin_unlock_bh(&ams_delta_lock);
469 if (apply)
470 ams_delta_latch2_write(AMS_DELTA_LATCH2_MODEM_CODEC,
471 mute ? AMS_DELTA_LATCH2_MODEM_CODEC : 0);
472 return 0;
475 /* Our codec DAI probably doesn't have its own .ops structure */
476 static struct snd_soc_dai_ops ams_delta_dai_ops = {
477 .digital_mute = ams_delta_digital_mute,
480 /* Will be used if the codec ever has its own digital_mute function */
481 static int ams_delta_startup(struct snd_pcm_substream *substream)
483 return ams_delta_digital_mute(NULL, 0);
486 static void ams_delta_shutdown(struct snd_pcm_substream *substream)
488 ams_delta_digital_mute(NULL, 1);
493 * Card initialization
496 static int ams_delta_cx20442_init(struct snd_soc_pcm_runtime *rtd)
498 struct snd_soc_codec *codec = rtd->codec;
499 struct snd_soc_dapm_context *dapm = &codec->dapm;
500 struct snd_soc_dai *codec_dai = rtd->codec_dai;
501 struct snd_soc_card *card = rtd->card;
502 int ret;
503 /* Codec is ready, now add/activate board specific controls */
505 /* Store a pointer to the codec structure for tty ldisc use */
506 cx20442_codec = codec;
508 /* Set up digital mute if not provided by the codec */
509 if (!codec_dai->driver->ops) {
510 codec_dai->driver->ops = &ams_delta_dai_ops;
511 } else {
512 ams_delta_ops.startup = ams_delta_startup;
513 ams_delta_ops.shutdown = ams_delta_shutdown;
516 /* Set codec bias level */
517 ams_delta_set_bias_level(card, dapm, SND_SOC_BIAS_STANDBY);
519 /* Add hook switch - can be used to control the codec from userspace
520 * even if line discipline fails */
521 ret = snd_soc_jack_new(rtd->codec, "hook_switch",
522 SND_JACK_HEADSET, &ams_delta_hook_switch);
523 if (ret)
524 dev_warn(card->dev,
525 "Failed to allocate resources for hook switch, "
526 "will continue without one.\n");
527 else {
528 ret = snd_soc_jack_add_gpios(&ams_delta_hook_switch,
529 ARRAY_SIZE(ams_delta_hook_switch_gpios),
530 ams_delta_hook_switch_gpios);
531 if (ret)
532 dev_warn(card->dev,
533 "Failed to set up hook switch GPIO line, "
534 "will continue with hook switch inactive.\n");
537 /* Register optional line discipline for over the modem control */
538 ret = tty_register_ldisc(N_V253, &cx81801_ops);
539 if (ret) {
540 dev_warn(card->dev,
541 "Failed to register line discipline, "
542 "will continue without any controls.\n");
543 return 0;
546 /* Add board specific DAPM widgets and routes */
547 ret = snd_soc_dapm_new_controls(dapm, ams_delta_dapm_widgets,
548 ARRAY_SIZE(ams_delta_dapm_widgets));
549 if (ret) {
550 dev_warn(card->dev,
551 "Failed to register DAPM controls, "
552 "will continue without any.\n");
553 return 0;
556 ret = snd_soc_dapm_add_routes(dapm, ams_delta_audio_map,
557 ARRAY_SIZE(ams_delta_audio_map));
558 if (ret) {
559 dev_warn(card->dev,
560 "Failed to set up DAPM routes, "
561 "will continue with codec default map.\n");
562 return 0;
565 /* Set up initial pin constellation */
566 snd_soc_dapm_disable_pin(dapm, "Mouthpiece");
567 snd_soc_dapm_enable_pin(dapm, "Earpiece");
568 snd_soc_dapm_enable_pin(dapm, "Microphone");
569 snd_soc_dapm_disable_pin(dapm, "Speaker");
570 snd_soc_dapm_disable_pin(dapm, "AGCIN");
571 snd_soc_dapm_disable_pin(dapm, "AGCOUT");
572 snd_soc_dapm_sync(dapm);
574 /* Add virtual switch */
575 ret = snd_soc_add_controls(codec, ams_delta_audio_controls,
576 ARRAY_SIZE(ams_delta_audio_controls));
577 if (ret)
578 dev_warn(card->dev,
579 "Failed to register audio mode control, "
580 "will continue without it.\n");
582 return 0;
585 /* DAI glue - connects codec <--> CPU */
586 static struct snd_soc_dai_link ams_delta_dai_link = {
587 .name = "CX20442",
588 .stream_name = "CX20442",
589 .cpu_dai_name ="omap-mcbsp-dai.0",
590 .codec_dai_name = "cx20442-voice",
591 .init = ams_delta_cx20442_init,
592 .platform_name = "omap-pcm-audio",
593 .codec_name = "cx20442-codec",
594 .ops = &ams_delta_ops,
597 /* Audio card driver */
598 static struct snd_soc_card ams_delta_audio_card = {
599 .name = "AMS_DELTA",
600 .dai_link = &ams_delta_dai_link,
601 .num_links = 1,
602 .set_bias_level = ams_delta_set_bias_level,
605 /* Module init/exit */
606 static struct platform_device *ams_delta_audio_platform_device;
607 static struct platform_device *cx20442_platform_device;
609 static int __init ams_delta_module_init(void)
611 int ret;
613 if (!(machine_is_ams_delta()))
614 return -ENODEV;
616 ams_delta_audio_platform_device =
617 platform_device_alloc("soc-audio", -1);
618 if (!ams_delta_audio_platform_device)
619 return -ENOMEM;
621 platform_set_drvdata(ams_delta_audio_platform_device,
622 &ams_delta_audio_card);
624 ret = platform_device_add(ams_delta_audio_platform_device);
625 if (ret)
626 goto err;
629 * Codec platform device could be registered from elsewhere (board?),
630 * but I do it here as it makes sense only if used with the card.
632 cx20442_platform_device =
633 platform_device_register_simple("cx20442-codec", -1, NULL, 0);
634 return 0;
635 err:
636 platform_device_put(ams_delta_audio_platform_device);
637 return ret;
639 module_init(ams_delta_module_init);
641 static void __exit ams_delta_module_exit(void)
643 if (tty_unregister_ldisc(N_V253) != 0)
644 dev_warn(&ams_delta_audio_platform_device->dev,
645 "failed to unregister V253 line discipline\n");
647 snd_soc_jack_free_gpios(&ams_delta_hook_switch,
648 ARRAY_SIZE(ams_delta_hook_switch_gpios),
649 ams_delta_hook_switch_gpios);
651 /* Keep modem power on */
652 ams_delta_set_bias_level(&ams_delta_audio_card,
653 &ams_delta_audio_card.rtd[0].codec->dapm,
654 SND_SOC_BIAS_STANDBY);
656 platform_device_unregister(cx20442_platform_device);
657 platform_device_unregister(ams_delta_audio_platform_device);
659 module_exit(ams_delta_module_exit);
661 MODULE_AUTHOR("Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>");
662 MODULE_DESCRIPTION("ALSA SoC driver for Amstrad E3 (Delta) videophone");
663 MODULE_LICENSE("GPL");