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-rw-r--r--sound/soc/omap/Kconfig15
-rw-r--r--sound/soc/omap/Makefile4
-rw-r--r--sound/soc/omap/ams-delta.c646
-rw-r--r--sound/soc/omap/n810.c12
-rw-r--r--sound/soc/omap/omap-mcbsp.c123
-rw-r--r--sound/soc/omap/omap-mcbsp.h4
-rw-r--r--sound/soc/omap/omap-pcm.c53
-rw-r--r--sound/soc/omap/omap-pcm.h2
-rw-r--r--sound/soc/omap/sdp3430.c18
-rw-r--r--sound/soc/omap/zoom2.c314
10 files changed, 1159 insertions, 32 deletions
diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig
index b771238662b6..2dee9839be86 100644
--- a/sound/soc/omap/Kconfig
+++ b/sound/soc/omap/Kconfig
@@ -15,6 +15,14 @@ config SND_OMAP_SOC_N810
15 help 15 help
16 Say Y if you want to add support for SoC audio on Nokia N810. 16 Say Y if you want to add support for SoC audio on Nokia N810.
17 17
18config SND_OMAP_SOC_AMS_DELTA
19 tristate "SoC Audio support for Amstrad E3 (Delta) videophone"
20 depends on SND_OMAP_SOC && MACH_AMS_DELTA
21 select SND_OMAP_SOC_MCBSP
22 select SND_SOC_CX20442
23 help
24 Say Y if you want to add support for SoC audio on Amstrad Delta.
25
18config SND_OMAP_SOC_OSK5912 26config SND_OMAP_SOC_OSK5912
19 tristate "SoC Audio support for omap osk5912" 27 tristate "SoC Audio support for omap osk5912"
20 depends on SND_OMAP_SOC && MACH_OMAP_OSK && I2C 28 depends on SND_OMAP_SOC && MACH_OMAP_OSK && I2C
@@ -72,4 +80,11 @@ config SND_OMAP_SOC_OMAP3_BEAGLE
72 help 80 help
73 Say Y if you want to add support for SoC audio on the Beagleboard. 81 Say Y if you want to add support for SoC audio on the Beagleboard.
74 82
83config SND_OMAP_SOC_ZOOM2
84 tristate "SoC Audio support for Zoom2"
85 depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP_ZOOM2
86 select SND_OMAP_SOC_MCBSP
87 select SND_SOC_TWL4030
88 help
89 Say Y if you want to add support for Soc audio on Zoom2 board.
75 90
diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile
index a37f49862389..02d69471dcb5 100644
--- a/sound/soc/omap/Makefile
+++ b/sound/soc/omap/Makefile
@@ -7,6 +7,7 @@ obj-$(CONFIG_SND_OMAP_SOC_MCBSP) += snd-soc-omap-mcbsp.o
7 7
8# OMAP Machine Support 8# OMAP Machine Support
9snd-soc-n810-objs := n810.o 9snd-soc-n810-objs := n810.o
10snd-soc-ams-delta-objs := ams-delta.o
10snd-soc-osk5912-objs := osk5912.o 11snd-soc-osk5912-objs := osk5912.o
11snd-soc-overo-objs := overo.o 12snd-soc-overo-objs := overo.o
12snd-soc-omap2evm-objs := omap2evm.o 13snd-soc-omap2evm-objs := omap2evm.o
@@ -14,8 +15,10 @@ snd-soc-omap3evm-objs := omap3evm.o
14snd-soc-sdp3430-objs := sdp3430.o 15snd-soc-sdp3430-objs := sdp3430.o
15snd-soc-omap3pandora-objs := omap3pandora.o 16snd-soc-omap3pandora-objs := omap3pandora.o
16snd-soc-omap3beagle-objs := omap3beagle.o 17snd-soc-omap3beagle-objs := omap3beagle.o
18snd-soc-zoom2-objs := zoom2.o
17 19
18obj-$(CONFIG_SND_OMAP_SOC_N810) += snd-soc-n810.o 20obj-$(CONFIG_SND_OMAP_SOC_N810) += snd-soc-n810.o
21obj-$(CONFIG_SND_OMAP_SOC_AMS_DELTA) += snd-soc-ams-delta.o
19obj-$(CONFIG_SND_OMAP_SOC_OSK5912) += snd-soc-osk5912.o 22obj-$(CONFIG_SND_OMAP_SOC_OSK5912) += snd-soc-osk5912.o
20obj-$(CONFIG_SND_OMAP_SOC_OVERO) += snd-soc-overo.o 23obj-$(CONFIG_SND_OMAP_SOC_OVERO) += snd-soc-overo.o
21obj-$(CONFIG_MACH_OMAP2EVM) += snd-soc-omap2evm.o 24obj-$(CONFIG_MACH_OMAP2EVM) += snd-soc-omap2evm.o
@@ -23,3 +26,4 @@ obj-$(CONFIG_MACH_OMAP3EVM) += snd-soc-omap3evm.o
23obj-$(CONFIG_SND_OMAP_SOC_SDP3430) += snd-soc-sdp3430.o 26obj-$(CONFIG_SND_OMAP_SOC_SDP3430) += snd-soc-sdp3430.o
24obj-$(CONFIG_SND_OMAP_SOC_OMAP3_PANDORA) += snd-soc-omap3pandora.o 27obj-$(CONFIG_SND_OMAP_SOC_OMAP3_PANDORA) += snd-soc-omap3pandora.o
25obj-$(CONFIG_SND_OMAP_SOC_OMAP3_BEAGLE) += snd-soc-omap3beagle.o 28obj-$(CONFIG_SND_OMAP_SOC_OMAP3_BEAGLE) += snd-soc-omap3beagle.o
29obj-$(CONFIG_SND_OMAP_SOC_ZOOM2) += snd-soc-zoom2.o
diff --git a/sound/soc/omap/ams-delta.c b/sound/soc/omap/ams-delta.c
new file mode 100644
index 000000000000..5a5166ac7279
--- /dev/null
+++ b/sound/soc/omap/ams-delta.c
@@ -0,0 +1,646 @@
1/*
2 * ams-delta.c -- SoC audio for Amstrad E3 (Delta) videophone
3 *
4 * Copyright (C) 2009 Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
5 *
6 * Initially based on sound/soc/omap/osk5912.x
7 * Copyright (C) 2008 Mistral Solutions
8 *
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.
12 *
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.
17 *
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
22 *
23 */
24
25#include <linux/gpio.h>
26#include <linux/spinlock.h>
27#include <linux/tty.h>
28
29#include <sound/soc-dapm.h>
30#include <sound/jack.h>
31
32#include <asm/mach-types.h>
33
34#include <mach/board-ams-delta.h>
35#include <mach/mcbsp.h>
36
37#include "omap-mcbsp.h"
38#include "omap-pcm.h"
39#include "../codecs/cx20442.h"
40
41
42/* Board specific DAPM widgets */
43 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),
50};
51
52/* How they are connected to codec pins */
53static const struct snd_soc_dapm_route ams_delta_audio_map[] = {
54 {"TELIN", NULL, "Mouthpiece"},
55 {"Earpiece", NULL, "TELOUT"},
56
57 {"MIC", NULL, "Microphone"},
58 {"Speaker", NULL, "SPKOUT"},
59};
60
61/*
62 * Controls, functional after the modem line discipline is activated.
63 */
64
65/* Virtual switch: audio input/output constellations */
66static const char *ams_delta_audio_mode[] =
67 {"Mixed", "Handset", "Handsfree", "Speakerphone"};
68
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
75
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))
83
84unsigned short ams_delta_audio_mode_pins[] = {
85 AMS_DELTA_MIXED,
86 AMS_DELTA_HANDSET,
87 AMS_DELTA_HANDSFREE,
88 AMS_DELTA_SPEAKERPHONE,
89};
90
91static unsigned short ams_delta_audio_agc;
92
93static int ams_delta_set_audio_mode(struct snd_kcontrol *kcontrol,
94 struct snd_ctl_elem_value *ucontrol)
95{
96 struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
97 struct soc_enum *control = (struct soc_enum *)kcontrol->private_value;
98 unsigned short pins;
99 int pin, changed = 0;
100
101 /* Refuse any mode changes if we are not able to control the codec. */
102 if (!codec->control_data)
103 return -EUNATCH;
104
105 if (ucontrol->value.enumerated.item[0] >= control->max)
106 return -EINVAL;
107
108 mutex_lock(&codec->mutex);
109
110 /* Translate selection to bitmap */
111 pins = ams_delta_audio_mode_pins[ucontrol->value.enumerated.item[0]];
112
113 /* Setup pins after corresponding bits if changed */
114 pin = !!(pins & (1 << AMS_DELTA_MOUTHPIECE));
115 if (pin != snd_soc_dapm_get_pin_status(codec, "Mouthpiece")) {
116 changed = 1;
117 if (pin)
118 snd_soc_dapm_enable_pin(codec, "Mouthpiece");
119 else
120 snd_soc_dapm_disable_pin(codec, "Mouthpiece");
121 }
122 pin = !!(pins & (1 << AMS_DELTA_EARPIECE));
123 if (pin != snd_soc_dapm_get_pin_status(codec, "Earpiece")) {
124 changed = 1;
125 if (pin)
126 snd_soc_dapm_enable_pin(codec, "Earpiece");
127 else
128 snd_soc_dapm_disable_pin(codec, "Earpiece");
129 }
130 pin = !!(pins & (1 << AMS_DELTA_MICROPHONE));
131 if (pin != snd_soc_dapm_get_pin_status(codec, "Microphone")) {
132 changed = 1;
133 if (pin)
134 snd_soc_dapm_enable_pin(codec, "Microphone");
135 else
136 snd_soc_dapm_disable_pin(codec, "Microphone");
137 }
138 pin = !!(pins & (1 << AMS_DELTA_SPEAKER));
139 if (pin != snd_soc_dapm_get_pin_status(codec, "Speaker")) {
140 changed = 1;
141 if (pin)
142 snd_soc_dapm_enable_pin(codec, "Speaker");
143 else
144 snd_soc_dapm_disable_pin(codec, "Speaker");
145 }
146 pin = !!(pins & (1 << AMS_DELTA_AGC));
147 if (pin != ams_delta_audio_agc) {
148 ams_delta_audio_agc = pin;
149 changed = 1;
150 if (pin)
151 snd_soc_dapm_enable_pin(codec, "AGCIN");
152 else
153 snd_soc_dapm_disable_pin(codec, "AGCIN");
154 }
155 if (changed)
156 snd_soc_dapm_sync(codec);
157
158 mutex_unlock(&codec->mutex);
159
160 return changed;
161}
162
163static int ams_delta_get_audio_mode(struct snd_kcontrol *kcontrol,
164 struct snd_ctl_elem_value *ucontrol)
165{
166 struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
167 unsigned short pins, mode;
168
169 pins = ((snd_soc_dapm_get_pin_status(codec, "Mouthpiece") <<
170 AMS_DELTA_MOUTHPIECE) |
171 (snd_soc_dapm_get_pin_status(codec, "Earpiece") <<
172 AMS_DELTA_EARPIECE));
173 if (pins)
174 pins |= (snd_soc_dapm_get_pin_status(codec, "Microphone") <<
175 AMS_DELTA_MICROPHONE);
176 else
177 pins = ((snd_soc_dapm_get_pin_status(codec, "Microphone") <<
178 AMS_DELTA_MICROPHONE) |
179 (snd_soc_dapm_get_pin_status(codec, "Speaker") <<
180 AMS_DELTA_SPEAKER) |
181 (ams_delta_audio_agc << AMS_DELTA_AGC));
182
183 for (mode = 0; mode < ARRAY_SIZE(ams_delta_audio_mode); mode++)
184 if (pins == ams_delta_audio_mode_pins[mode])
185 break;
186
187 if (mode >= ARRAY_SIZE(ams_delta_audio_mode))
188 return -EINVAL;
189
190 ucontrol->value.enumerated.item[0] = mode;
191
192 return 0;
193}
194
195static const struct soc_enum ams_delta_audio_enum[] = {
196 SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(ams_delta_audio_mode),
197 ams_delta_audio_mode),
198};
199
200static const struct snd_kcontrol_new ams_delta_audio_controls[] = {
201 SOC_ENUM_EXT("Audio Mode", ams_delta_audio_enum[0],
202 ams_delta_get_audio_mode, ams_delta_set_audio_mode),
203};
204
205/* Hook switch */
206static struct snd_soc_jack ams_delta_hook_switch;
207static struct snd_soc_jack_gpio ams_delta_hook_switch_gpios[] = {
208 {
209 .gpio = 4,
210 .name = "hook_switch",
211 .report = SND_JACK_HEADSET,
212 .invert = 1,
213 .debounce_time = 150,
214 }
215};
216
217/* After we are able to control the codec over the modem,
218 * the hook switch can be used for dynamic DAPM reconfiguration. */
219static struct snd_soc_jack_pin ams_delta_hook_switch_pins[] = {
220 /* Handset */
221 {
222 .pin = "Mouthpiece",
223 .mask = SND_JACK_MICROPHONE,
224 },
225 {
226 .pin = "Earpiece",
227 .mask = SND_JACK_HEADPHONE,
228 },
229 /* Handsfree */
230 {
231 .pin = "Microphone",
232 .mask = SND_JACK_MICROPHONE,
233 .invert = 1,
234 },
235 {
236 .pin = "Speaker",
237 .mask = SND_JACK_HEADPHONE,
238 .invert = 1,
239 },
240};
241
242
243/*
244 * Modem line discipline, required for making above controls functional.
245 * Activated from userspace with ldattach, possibly invoked from udev rule.
246 */
247
248/* To actually apply any modem controlled configuration changes to the codec,
249 * we must connect codec DAI pins to the modem for a moment. Be carefull not
250 * to interfere with our digital mute function that shares the same hardware. */
251static struct timer_list cx81801_timer;
252static bool cx81801_cmd_pending;
253static bool ams_delta_muted;
254static DEFINE_SPINLOCK(ams_delta_lock);
255
256static void cx81801_timeout(unsigned long data)
257{
258 int muted;
259
260 spin_lock(&ams_delta_lock);
261 cx81801_cmd_pending = 0;
262 muted = ams_delta_muted;
263 spin_unlock(&ams_delta_lock);
264
265 /* Reconnect the codec DAI back from the modem to the CPU DAI
266 * only if digital mute still off */
267 if (!muted)
268 ams_delta_latch2_write(AMS_DELTA_LATCH2_MODEM_CODEC, 0);
269}
270
271/* Line discipline .open() */
272static int cx81801_open(struct tty_struct *tty)
273{
274 return v253_ops.open(tty);
275}
276
277/* Line discipline .close() */
278static void cx81801_close(struct tty_struct *tty)
279{
280 struct snd_soc_codec *codec = tty->disc_data;
281
282 del_timer_sync(&cx81801_timer);
283
284 v253_ops.close(tty);
285
286 /* Prevent the hook switch from further changing the DAPM pins */
287 INIT_LIST_HEAD(&ams_delta_hook_switch.pins);
288
289 /* Revert back to default audio input/output constellation */
290 snd_soc_dapm_disable_pin(codec, "Mouthpiece");
291 snd_soc_dapm_enable_pin(codec, "Earpiece");
292 snd_soc_dapm_enable_pin(codec, "Microphone");
293 snd_soc_dapm_disable_pin(codec, "Speaker");
294 snd_soc_dapm_disable_pin(codec, "AGCIN");
295 snd_soc_dapm_sync(codec);
296}
297
298/* Line discipline .hangup() */
299static int cx81801_hangup(struct tty_struct *tty)
300{
301 cx81801_close(tty);
302 return 0;
303}
304
305/* Line discipline .recieve_buf() */
306static void cx81801_receive(struct tty_struct *tty,
307 const unsigned char *cp, char *fp, int count)
308{
309 struct snd_soc_codec *codec = tty->disc_data;
310 const unsigned char *c;
311 int apply, ret;
312
313 if (!codec->control_data) {
314 /* First modem response, complete setup procedure */
315
316 /* Initialize timer used for config pulse generation */
317 setup_timer(&cx81801_timer, cx81801_timeout, 0);
318
319 v253_ops.receive_buf(tty, cp, fp, count);
320
321 /* Link hook switch to DAPM pins */
322 ret = snd_soc_jack_add_pins(&ams_delta_hook_switch,
323 ARRAY_SIZE(ams_delta_hook_switch_pins),
324 ams_delta_hook_switch_pins);
325 if (ret)
326 dev_warn(codec->socdev->card->dev,
327 "Failed to link hook switch to DAPM pins, "
328 "will continue with hook switch unlinked.\n");
329
330 return;
331 }
332
333 v253_ops.receive_buf(tty, cp, fp, count);
334
335 for (c = &cp[count - 1]; c >= cp; c--) {
336 if (*c != '\r')
337 continue;
338 /* Complete modem response received, apply config to codec */
339
340 spin_lock_bh(&ams_delta_lock);
341 mod_timer(&cx81801_timer, jiffies + msecs_to_jiffies(150));
342 apply = !ams_delta_muted && !cx81801_cmd_pending;
343 cx81801_cmd_pending = 1;
344 spin_unlock_bh(&ams_delta_lock);
345
346 /* Apply config pulse by connecting the codec to the modem
347 * if not already done */
348 if (apply)
349 ams_delta_latch2_write(AMS_DELTA_LATCH2_MODEM_CODEC,
350 AMS_DELTA_LATCH2_MODEM_CODEC);
351 break;
352 }
353}
354
355/* Line discipline .write_wakeup() */
356static void cx81801_wakeup(struct tty_struct *tty)
357{
358 v253_ops.write_wakeup(tty);
359}
360
361static struct tty_ldisc_ops cx81801_ops = {
362 .magic = TTY_LDISC_MAGIC,
363 .name = "cx81801",
364 .owner = THIS_MODULE,
365 .open = cx81801_open,
366 .close = cx81801_close,
367 .hangup = cx81801_hangup,
368 .receive_buf = cx81801_receive,
369 .write_wakeup = cx81801_wakeup,
370};
371
372
373/*
374 * Even if not very usefull, the sound card can still work without any of the
375 * above functonality activated. You can still control its audio input/output
376 * constellation and speakerphone gain from userspace by issueing AT commands
377 * over the modem port.
378 */
379
380static int ams_delta_hw_params(struct snd_pcm_substream *substream,
381 struct snd_pcm_hw_params *params)
382{
383 struct snd_soc_pcm_runtime *rtd = substream->private_data;
384
385 /* Set cpu DAI configuration */
386 return snd_soc_dai_set_fmt(rtd->dai->cpu_dai,
387 SND_SOC_DAIFMT_DSP_A |
388 SND_SOC_DAIFMT_NB_NF |
389 SND_SOC_DAIFMT_CBM_CFM);
390}
391
392static struct snd_soc_ops ams_delta_ops = {
393 .hw_params = ams_delta_hw_params,
394};
395
396
397/* Board specific codec bias level control */
398static int ams_delta_set_bias_level(struct snd_soc_card *card,
399 enum snd_soc_bias_level level)
400{
401 struct snd_soc_codec *codec = card->codec;
402
403 switch (level) {
404 case SND_SOC_BIAS_ON:
405 case SND_SOC_BIAS_PREPARE:
406 case SND_SOC_BIAS_STANDBY:
407 if (codec->bias_level == SND_SOC_BIAS_OFF)
408 ams_delta_latch2_write(AMS_DELTA_LATCH2_MODEM_NRESET,
409 AMS_DELTA_LATCH2_MODEM_NRESET);
410 break;
411 case SND_SOC_BIAS_OFF:
412 if (codec->bias_level != SND_SOC_BIAS_OFF)
413 ams_delta_latch2_write(AMS_DELTA_LATCH2_MODEM_NRESET,
414 0);
415 }
416 codec->bias_level = level;
417
418 return 0;
419}
420
421/* Digital mute implemented using modem/CPU multiplexer.
422 * Shares hardware with codec config pulse generation */
423static bool ams_delta_muted = 1;
424
425static int ams_delta_digital_mute(struct snd_soc_dai *dai, int mute)
426{
427 int apply;
428
429 if (ams_delta_muted == mute)
430 return 0;
431
432 spin_lock_bh(&ams_delta_lock);
433 ams_delta_muted = mute;
434 apply = !cx81801_cmd_pending;
435 spin_unlock_bh(&ams_delta_lock);
436
437 if (apply)
438 ams_delta_latch2_write(AMS_DELTA_LATCH2_MODEM_CODEC,
439 mute ? AMS_DELTA_LATCH2_MODEM_CODEC : 0);
440 return 0;
441}
442
443/* Our codec DAI probably doesn't have its own .ops structure */
444static struct snd_soc_dai_ops ams_delta_dai_ops = {
445 .digital_mute = ams_delta_digital_mute,
446};
447
448/* Will be used if the codec ever has its own digital_mute function */
449static int ams_delta_startup(struct snd_pcm_substream *substream)
450{
451 return ams_delta_digital_mute(NULL, 0);
452}
453
454static void ams_delta_shutdown(struct snd_pcm_substream *substream)
455{
456 ams_delta_digital_mute(NULL, 1);
457}
458
459
460/*
461 * Card initialization
462 */
463
464static int ams_delta_cx20442_init(struct snd_soc_codec *codec)
465{
466 struct snd_soc_dai *codec_dai = codec->dai;
467 struct snd_soc_card *card = codec->socdev->card;
468 int ret;
469 /* Codec is ready, now add/activate board specific controls */
470
471 /* Set up digital mute if not provided by the codec */
472 if (!codec_dai->ops) {
473 codec_dai->ops = &ams_delta_dai_ops;
474 } else if (!codec_dai->ops->digital_mute) {
475 codec_dai->ops->digital_mute = ams_delta_digital_mute;
476 } else {
477 ams_delta_ops.startup = ams_delta_startup;
478 ams_delta_ops.shutdown = ams_delta_shutdown;
479 }
480
481 /* Set codec bias level */
482 ams_delta_set_bias_level(card, SND_SOC_BIAS_STANDBY);
483
484 /* Add hook switch - can be used to control the codec from userspace
485 * even if line discipline fails */
486 ret = snd_soc_jack_new(card, "hook_switch",
487 SND_JACK_HEADSET, &ams_delta_hook_switch);
488 if (ret)
489 dev_warn(card->dev,
490 "Failed to allocate resources for hook switch, "
491 "will continue without one.\n");
492 else {
493 ret = snd_soc_jack_add_gpios(&ams_delta_hook_switch,
494 ARRAY_SIZE(ams_delta_hook_switch_gpios),
495 ams_delta_hook_switch_gpios);
496 if (ret)
497 dev_warn(card->dev,
498 "Failed to set up hook switch GPIO line, "
499 "will continue with hook switch inactive.\n");
500 }
501
502 /* Register optional line discipline for over the modem control */
503 ret = tty_register_ldisc(N_V253, &cx81801_ops);
504 if (ret) {
505 dev_warn(card->dev,
506 "Failed to register line discipline, "
507 "will continue without any controls.\n");
508 return 0;
509 }
510
511 /* Add board specific DAPM widgets and routes */
512 ret = snd_soc_dapm_new_controls(codec, ams_delta_dapm_widgets,
513 ARRAY_SIZE(ams_delta_dapm_widgets));
514 if (ret) {
515 dev_warn(card->dev,
516 "Failed to register DAPM controls, "
517 "will continue without any.\n");
518 return 0;
519 }
520
521 ret = snd_soc_dapm_add_routes(codec, ams_delta_audio_map,
522 ARRAY_SIZE(ams_delta_audio_map));
523 if (ret) {
524 dev_warn(card->dev,
525 "Failed to set up DAPM routes, "
526 "will continue with codec default map.\n");
527 return 0;
528 }
529
530 /* Set up initial pin constellation */
531 snd_soc_dapm_disable_pin(codec, "Mouthpiece");
532 snd_soc_dapm_enable_pin(codec, "Earpiece");
533 snd_soc_dapm_enable_pin(codec, "Microphone");
534 snd_soc_dapm_disable_pin(codec, "Speaker");
535 snd_soc_dapm_disable_pin(codec, "AGCIN");
536 snd_soc_dapm_disable_pin(codec, "AGCOUT");
537 snd_soc_dapm_sync(codec);
538
539 /* Add virtual switch */
540 ret = snd_soc_add_controls(codec, ams_delta_audio_controls,
541 ARRAY_SIZE(ams_delta_audio_controls));
542 if (ret)
543 dev_warn(card->dev,
544 "Failed to register audio mode control, "
545 "will continue without it.\n");
546
547 return 0;
548}
549
550/* DAI glue - connects codec <--> CPU */
551static struct snd_soc_dai_link ams_delta_dai_link = {
552 .name = "CX20442",
553 .stream_name = "CX20442",
554 .cpu_dai = &omap_mcbsp_dai[0],
555 .codec_dai = &cx20442_dai,
556 .init = ams_delta_cx20442_init,
557 .ops = &ams_delta_ops,
558};
559
560/* Audio card driver */
561static struct snd_soc_card ams_delta_audio_card = {
562 .name = "AMS_DELTA",
563 .platform = &omap_soc_platform,
564 .dai_link = &ams_delta_dai_link,
565 .num_links = 1,
566 .set_bias_level = ams_delta_set_bias_level,
567};
568
569/* Audio subsystem */
570static struct snd_soc_device ams_delta_snd_soc_device = {
571 .card = &ams_delta_audio_card,
572 .codec_dev = &cx20442_codec_dev,
573};
574
575/* Module init/exit */
576static struct platform_device *ams_delta_audio_platform_device;
577static struct platform_device *cx20442_platform_device;
578
579static int __init ams_delta_module_init(void)
580{
581 int ret;
582
583 if (!(machine_is_ams_delta()))
584 return -ENODEV;
585
586 ams_delta_audio_platform_device =
587 platform_device_alloc("soc-audio", -1);
588 if (!ams_delta_audio_platform_device)
589 return -ENOMEM;
590
591 platform_set_drvdata(ams_delta_audio_platform_device,
592 &ams_delta_snd_soc_device);
593 ams_delta_snd_soc_device.dev = &ams_delta_audio_platform_device->dev;
594 *(unsigned int *)ams_delta_dai_link.cpu_dai->private_data = OMAP_MCBSP1;
595
596 ret = platform_device_add(ams_delta_audio_platform_device);
597 if (ret)
598 goto err;
599
600 /*
601 * Codec platform device could be registered from elsewhere (board?),
602 * but I do it here as it makes sense only if used with the card.
603 */
604 cx20442_platform_device = platform_device_register_simple("cx20442",
605 -1, NULL, 0);
606 return 0;
607err:
608 platform_device_put(ams_delta_audio_platform_device);
609 return ret;
610}
611module_init(ams_delta_module_init);
612
613static void __exit ams_delta_module_exit(void)
614{
615 struct snd_soc_codec *codec;
616 struct tty_struct *tty;
617
618 if (ams_delta_audio_card.codec) {
619 codec = ams_delta_audio_card.codec;
620
621 if (codec->control_data) {
622 tty = codec->control_data;
623
624 tty_hangup(tty);
625 }
626 }
627
628 if (tty_unregister_ldisc(N_V253) != 0)
629 dev_warn(&ams_delta_audio_platform_device->dev,
630 "failed to unregister V253 line discipline\n");
631
632 snd_soc_jack_free_gpios(&ams_delta_hook_switch,
633 ARRAY_SIZE(ams_delta_hook_switch_gpios),
634 ams_delta_hook_switch_gpios);
635
636 /* Keep modem power on */
637 ams_delta_set_bias_level(&ams_delta_audio_card, SND_SOC_BIAS_STANDBY);
638
639 platform_device_unregister(cx20442_platform_device);
640 platform_device_unregister(ams_delta_audio_platform_device);
641}
642module_exit(ams_delta_module_exit);
643
644MODULE_AUTHOR("Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>");
645MODULE_DESCRIPTION("ALSA SoC driver for Amstrad E3 (Delta) videophone");
646MODULE_LICENSE("GPL");
diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c
index b60b1dfbc435..0a505938e42b 100644
--- a/sound/soc/omap/n810.c
+++ b/sound/soc/omap/n810.c
@@ -22,6 +22,7 @@
22 */ 22 */
23 23
24#include <linux/clk.h> 24#include <linux/clk.h>
25#include <linux/i2c.h>
25#include <linux/platform_device.h> 26#include <linux/platform_device.h>
26#include <sound/core.h> 27#include <sound/core.h>
27#include <sound/pcm.h> 28#include <sound/pcm.h>
@@ -322,8 +323,6 @@ static struct snd_soc_card snd_soc_n810 = {
322 323
323/* Audio private data */ 324/* Audio private data */
324static struct aic3x_setup_data n810_aic33_setup = { 325static struct aic3x_setup_data n810_aic33_setup = {
325 .i2c_bus = 2,
326 .i2c_address = 0x18,
327 .gpio_func[0] = AIC3X_GPIO1_FUNC_DISABLED, 326 .gpio_func[0] = AIC3X_GPIO1_FUNC_DISABLED,
328 .gpio_func[1] = AIC3X_GPIO2_FUNC_DIGITAL_MIC_INPUT, 327 .gpio_func[1] = AIC3X_GPIO2_FUNC_DIGITAL_MIC_INPUT,
329}; 328};
@@ -337,6 +336,13 @@ static struct snd_soc_device n810_snd_devdata = {
337 336
338static struct platform_device *n810_snd_device; 337static struct platform_device *n810_snd_device;
339 338
339/* temporary i2c device creation until this can be moved into the machine
340 * support file.
341*/
342static struct i2c_board_info i2c_device[] = {
343 { I2C_BOARD_INFO("tlv320aic3x", 0x1b), }
344};
345
340static int __init n810_soc_init(void) 346static int __init n810_soc_init(void)
341{ 347{
342 int err; 348 int err;
@@ -345,6 +351,8 @@ static int __init n810_soc_init(void)
345 if (!(machine_is_nokia_n810() || machine_is_nokia_n810_wimax())) 351 if (!(machine_is_nokia_n810() || machine_is_nokia_n810_wimax()))
346 return -ENODEV; 352 return -ENODEV;
347 353
354 i2c_register_board_info(1, i2c_device, ARRAY_SIZE(i2c_device));
355
348 n810_snd_device = platform_device_alloc("soc-audio", -1); 356 n810_snd_device = platform_device_alloc("soc-audio", -1);
349 if (!n810_snd_device) 357 if (!n810_snd_device)
350 return -ENOMEM; 358 return -ENOMEM;
diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c
index a5d46a7b196a..3341f49402ca 100644
--- a/sound/soc/omap/omap-mcbsp.c
+++ b/sound/soc/omap/omap-mcbsp.c
@@ -139,27 +139,67 @@ static const unsigned long omap34xx_mcbsp_port[][2] = {
139static const unsigned long omap34xx_mcbsp_port[][2] = {}; 139static const unsigned long omap34xx_mcbsp_port[][2] = {};
140#endif 140#endif
141 141
142static void omap_mcbsp_set_threshold(struct snd_pcm_substream *substream)
143{
144 struct snd_soc_pcm_runtime *rtd = substream->private_data;
145 struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
146 struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data);
147 int dma_op_mode = omap_mcbsp_get_dma_op_mode(mcbsp_data->bus_id);
148 int samples;
149
150 /* TODO: Currently, MODE_ELEMENT == MODE_FRAME */
151 if (dma_op_mode == MCBSP_DMA_MODE_THRESHOLD)
152 samples = snd_pcm_lib_period_bytes(substream) >> 1;
153 else
154 samples = 1;
155
156 /* Configure McBSP internal buffer usage */
157 if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
158 omap_mcbsp_set_tx_threshold(mcbsp_data->bus_id, samples - 1);
159 else
160 omap_mcbsp_set_rx_threshold(mcbsp_data->bus_id, samples - 1);
161}
162
142static int omap_mcbsp_dai_startup(struct snd_pcm_substream *substream, 163static int omap_mcbsp_dai_startup(struct snd_pcm_substream *substream,
143 struct snd_soc_dai *dai) 164 struct snd_soc_dai *dai)
144{ 165{
145 struct snd_soc_pcm_runtime *rtd = substream->private_data; 166 struct snd_soc_pcm_runtime *rtd = substream->private_data;
146 struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; 167 struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
147 struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data); 168 struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data);
169 int bus_id = mcbsp_data->bus_id;
148 int err = 0; 170 int err = 0;
149 171
150 if (cpu_is_omap343x() && mcbsp_data->bus_id == 1) { 172 if (!cpu_dai->active)
173 err = omap_mcbsp_request(bus_id);
174
175 if (cpu_is_omap343x()) {
176 int dma_op_mode = omap_mcbsp_get_dma_op_mode(bus_id);
177 int max_period;
178
151 /* 179 /*
152 * McBSP2 in OMAP3 has 1024 * 32-bit internal audio buffer. 180 * McBSP2 in OMAP3 has 1024 * 32-bit internal audio buffer.
153 * Set constraint for minimum buffer size to the same than FIFO 181 * Set constraint for minimum buffer size to the same than FIFO
154 * size in order to avoid underruns in playback startup because 182 * size in order to avoid underruns in playback startup because
155 * HW is keeping the DMA request active until FIFO is filled. 183 * HW is keeping the DMA request active until FIFO is filled.
156 */ 184 */
157 snd_pcm_hw_constraint_minmax(substream->runtime, 185 if (bus_id == 1)
158 SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 4096, UINT_MAX); 186 snd_pcm_hw_constraint_minmax(substream->runtime,
159 } 187 SNDRV_PCM_HW_PARAM_BUFFER_BYTES,
188 4096, UINT_MAX);
160 189
161 if (!cpu_dai->active) 190 if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
162 err = omap_mcbsp_request(mcbsp_data->bus_id); 191 max_period = omap_mcbsp_get_max_tx_threshold(bus_id);
192 else
193 max_period = omap_mcbsp_get_max_rx_threshold(bus_id);
194
195 max_period++;
196 max_period <<= 1;
197
198 if (dma_op_mode == MCBSP_DMA_MODE_THRESHOLD)
199 snd_pcm_hw_constraint_minmax(substream->runtime,
200 SNDRV_PCM_HW_PARAM_PERIOD_BYTES,
201 32, max_period);
202 }
163 203
164 return err; 204 return err;
165} 205}
@@ -183,21 +223,21 @@ static int omap_mcbsp_dai_trigger(struct snd_pcm_substream *substream, int cmd,
183 struct snd_soc_pcm_runtime *rtd = substream->private_data; 223 struct snd_soc_pcm_runtime *rtd = substream->private_data;
184 struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; 224 struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
185 struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data); 225 struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data);
186 int err = 0; 226 int err = 0, play = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK);
187 227
188 switch (cmd) { 228 switch (cmd) {
189 case SNDRV_PCM_TRIGGER_START: 229 case SNDRV_PCM_TRIGGER_START:
190 case SNDRV_PCM_TRIGGER_RESUME: 230 case SNDRV_PCM_TRIGGER_RESUME:
191 case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: 231 case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
192 if (!mcbsp_data->active++) 232 mcbsp_data->active++;
193 omap_mcbsp_start(mcbsp_data->bus_id); 233 omap_mcbsp_start(mcbsp_data->bus_id, play, !play);
194 break; 234 break;
195 235
196 case SNDRV_PCM_TRIGGER_STOP: 236 case SNDRV_PCM_TRIGGER_STOP:
197 case SNDRV_PCM_TRIGGER_SUSPEND: 237 case SNDRV_PCM_TRIGGER_SUSPEND:
198 case SNDRV_PCM_TRIGGER_PAUSE_PUSH: 238 case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
199 if (!--mcbsp_data->active) 239 omap_mcbsp_stop(mcbsp_data->bus_id, play, !play);
200 omap_mcbsp_stop(mcbsp_data->bus_id); 240 mcbsp_data->active--;
201 break; 241 break;
202 default: 242 default:
203 err = -EINVAL; 243 err = -EINVAL;
@@ -215,7 +255,7 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream,
215 struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data); 255 struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data);
216 struct omap_mcbsp_reg_cfg *regs = &mcbsp_data->regs; 256 struct omap_mcbsp_reg_cfg *regs = &mcbsp_data->regs;
217 int dma, bus_id = mcbsp_data->bus_id, id = cpu_dai->id; 257 int dma, bus_id = mcbsp_data->bus_id, id = cpu_dai->id;
218 int wlen, channels, wpf; 258 int wlen, channels, wpf, sync_mode = OMAP_DMA_SYNC_ELEMENT;
219 unsigned long port; 259 unsigned long port;
220 unsigned int format; 260 unsigned int format;
221 261
@@ -231,6 +271,12 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream,
231 } else if (cpu_is_omap343x()) { 271 } else if (cpu_is_omap343x()) {
232 dma = omap24xx_dma_reqs[bus_id][substream->stream]; 272 dma = omap24xx_dma_reqs[bus_id][substream->stream];
233 port = omap34xx_mcbsp_port[bus_id][substream->stream]; 273 port = omap34xx_mcbsp_port[bus_id][substream->stream];
274 omap_mcbsp_dai_dma_params[id][substream->stream].set_threshold =
275 omap_mcbsp_set_threshold;
276 /* TODO: Currently, MODE_ELEMENT == MODE_FRAME */
277 if (omap_mcbsp_get_dma_op_mode(bus_id) ==
278 MCBSP_DMA_MODE_THRESHOLD)
279 sync_mode = OMAP_DMA_SYNC_FRAME;
234 } else { 280 } else {
235 return -ENODEV; 281 return -ENODEV;
236 } 282 }
@@ -238,6 +284,7 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream,
238 substream->stream ? "Audio Capture" : "Audio Playback"; 284 substream->stream ? "Audio Capture" : "Audio Playback";
239 omap_mcbsp_dai_dma_params[id][substream->stream].dma_req = dma; 285 omap_mcbsp_dai_dma_params[id][substream->stream].dma_req = dma;
240 omap_mcbsp_dai_dma_params[id][substream->stream].port_addr = port; 286 omap_mcbsp_dai_dma_params[id][substream->stream].port_addr = port;
287 omap_mcbsp_dai_dma_params[id][substream->stream].sync_mode = sync_mode;
241 cpu_dai->dma_data = &omap_mcbsp_dai_dma_params[id][substream->stream]; 288 cpu_dai->dma_data = &omap_mcbsp_dai_dma_params[id][substream->stream];
242 289
243 if (mcbsp_data->configured) { 290 if (mcbsp_data->configured) {
@@ -321,11 +368,14 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai,
321 /* Generic McBSP register settings */ 368 /* Generic McBSP register settings */
322 regs->spcr2 |= XINTM(3) | FREE; 369 regs->spcr2 |= XINTM(3) | FREE;
323 regs->spcr1 |= RINTM(3); 370 regs->spcr1 |= RINTM(3);
324 regs->rcr2 |= RFIG; 371 /* RFIG and XFIG are not defined in 34xx */
325 regs->xcr2 |= XFIG; 372 if (!cpu_is_omap34xx()) {
373 regs->rcr2 |= RFIG;
374 regs->xcr2 |= XFIG;
375 }
326 if (cpu_is_omap2430() || cpu_is_omap34xx()) { 376 if (cpu_is_omap2430() || cpu_is_omap34xx()) {
327 regs->xccr = DXENDLY(1) | XDMAEN; 377 regs->xccr = DXENDLY(1) | XDMAEN | XDISABLE;
328 regs->rccr = RFULL_CYCLE | RDMAEN; 378 regs->rccr = RFULL_CYCLE | RDMAEN | RDISABLE;
329 } 379 }
330 380
331 switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { 381 switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
@@ -462,6 +512,40 @@ static int omap_mcbsp_dai_set_clks_src(struct omap_mcbsp_data *mcbsp_data,
462 return 0; 512 return 0;
463} 513}
464 514
515static int omap_mcbsp_dai_set_rcvr_src(struct omap_mcbsp_data *mcbsp_data,
516 int clk_id)
517{
518 int sel_bit, set = 0;
519 u16 reg = OMAP2_CONTROL_DEVCONF0;
520
521 if (cpu_class_is_omap1())
522 return -EINVAL; /* TODO: Can this be implemented for OMAP1? */
523 if (mcbsp_data->bus_id != 0)
524 return -EINVAL;
525
526 switch (clk_id) {
527 case OMAP_MCBSP_CLKR_SRC_CLKX:
528 set = 1;
529 case OMAP_MCBSP_CLKR_SRC_CLKR:
530 sel_bit = 3;
531 break;
532 case OMAP_MCBSP_FSR_SRC_FSX:
533 set = 1;
534 case OMAP_MCBSP_FSR_SRC_FSR:
535 sel_bit = 4;
536 break;
537 default:
538 return -EINVAL;
539 }
540
541 if (set)
542 omap_ctrl_writel(omap_ctrl_readl(reg) | (1 << sel_bit), reg);
543 else
544 omap_ctrl_writel(omap_ctrl_readl(reg) & ~(1 << sel_bit), reg);
545
546 return 0;
547}
548
465static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_dai *cpu_dai, 549static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
466 int clk_id, unsigned int freq, 550 int clk_id, unsigned int freq,
467 int dir) 551 int dir)
@@ -484,6 +568,13 @@ static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
484 case OMAP_MCBSP_SYSCLK_CLKR_EXT: 568 case OMAP_MCBSP_SYSCLK_CLKR_EXT:
485 regs->pcr0 |= SCLKME; 569 regs->pcr0 |= SCLKME;
486 break; 570 break;
571
572 case OMAP_MCBSP_CLKR_SRC_CLKR:
573 case OMAP_MCBSP_CLKR_SRC_CLKX:
574 case OMAP_MCBSP_FSR_SRC_FSR:
575 case OMAP_MCBSP_FSR_SRC_FSX:
576 err = omap_mcbsp_dai_set_rcvr_src(mcbsp_data, clk_id);
577 break;
487 default: 578 default:
488 err = -ENODEV; 579 err = -ENODEV;
489 } 580 }
diff --git a/sound/soc/omap/omap-mcbsp.h b/sound/soc/omap/omap-mcbsp.h
index c8147aace813..647d2f981ab0 100644
--- a/sound/soc/omap/omap-mcbsp.h
+++ b/sound/soc/omap/omap-mcbsp.h
@@ -32,6 +32,10 @@ enum omap_mcbsp_clksrg_clk {
32 OMAP_MCBSP_SYSCLK_CLK, /* Internal ICLK */ 32 OMAP_MCBSP_SYSCLK_CLK, /* Internal ICLK */
33 OMAP_MCBSP_SYSCLK_CLKX_EXT, /* External CLKX pin */ 33 OMAP_MCBSP_SYSCLK_CLKX_EXT, /* External CLKX pin */
34 OMAP_MCBSP_SYSCLK_CLKR_EXT, /* External CLKR pin */ 34 OMAP_MCBSP_SYSCLK_CLKR_EXT, /* External CLKR pin */
35 OMAP_MCBSP_CLKR_SRC_CLKR, /* CLKR from CLKR pin */
36 OMAP_MCBSP_CLKR_SRC_CLKX, /* CLKR from CLKX pin */
37 OMAP_MCBSP_FSR_SRC_FSR, /* FSR from FSR pin */
38 OMAP_MCBSP_FSR_SRC_FSX, /* FSR from FSX pin */
35}; 39};
36 40
37/* McBSP dividers */ 41/* McBSP dividers */
diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c
index 84a1950880eb..5735945788bf 100644
--- a/sound/soc/omap/omap-pcm.c
+++ b/sound/soc/omap/omap-pcm.c
@@ -59,16 +59,31 @@ static void omap_pcm_dma_irq(int ch, u16 stat, void *data)
59 struct omap_runtime_data *prtd = runtime->private_data; 59 struct omap_runtime_data *prtd = runtime->private_data;
60 unsigned long flags; 60 unsigned long flags;
61 61
62 if (cpu_is_omap1510()) { 62 if ((cpu_is_omap1510()) &&
63 (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)) {
63 /* 64 /*
64 * OMAP1510 doesn't support DMA chaining so have to restart 65 * OMAP1510 doesn't fully support DMA progress counter
65 * the transfer after all periods are transferred 66 * and there is no software emulation implemented yet,
67 * so have to maintain our own playback progress counter
68 * that can be used by omap_pcm_pointer() instead.
66 */ 69 */
67 spin_lock_irqsave(&prtd->lock, flags); 70 spin_lock_irqsave(&prtd->lock, flags);
71 if ((stat == OMAP_DMA_LAST_IRQ) &&
72 (prtd->period_index == runtime->periods - 1)) {
73 /* we are in sync, do nothing */
74 spin_unlock_irqrestore(&prtd->lock, flags);
75 return;
76 }
68 if (prtd->period_index >= 0) { 77 if (prtd->period_index >= 0) {
69 if (++prtd->period_index == runtime->periods) { 78 if (stat & OMAP_DMA_BLOCK_IRQ) {
79 /* end of buffer reached, loop back */
80 prtd->period_index = 0;
81 } else if (stat & OMAP_DMA_LAST_IRQ) {
82 /* update the counter for the last period */
83 prtd->period_index = runtime->periods - 1;
84 } else if (++prtd->period_index >= runtime->periods) {
85 /* end of buffer missed? loop back */
70 prtd->period_index = 0; 86 prtd->period_index = 0;
71 omap_start_dma(prtd->dma_ch);
72 } 87 }
73 } 88 }
74 spin_unlock_irqrestore(&prtd->lock, flags); 89 spin_unlock_irqrestore(&prtd->lock, flags);
@@ -100,7 +115,7 @@ static int omap_pcm_hw_params(struct snd_pcm_substream *substream,
100 prtd->dma_data = dma_data; 115 prtd->dma_data = dma_data;
101 err = omap_request_dma(dma_data->dma_req, dma_data->name, 116 err = omap_request_dma(dma_data->dma_req, dma_data->name,
102 omap_pcm_dma_irq, substream, &prtd->dma_ch); 117 omap_pcm_dma_irq, substream, &prtd->dma_ch);
103 if (!err && !cpu_is_omap1510()) { 118 if (!err) {
104 /* 119 /*
105 * Link channel with itself so DMA doesn't need any 120 * Link channel with itself so DMA doesn't need any
106 * reprogramming while looping the buffer 121 * reprogramming while looping the buffer
@@ -119,8 +134,7 @@ static int omap_pcm_hw_free(struct snd_pcm_substream *substream)
119 if (prtd->dma_data == NULL) 134 if (prtd->dma_data == NULL)
120 return 0; 135 return 0;
121 136
122 if (!cpu_is_omap1510()) 137 omap_dma_unlink_lch(prtd->dma_ch, prtd->dma_ch);
123 omap_dma_unlink_lch(prtd->dma_ch, prtd->dma_ch);
124 omap_free_dma(prtd->dma_ch); 138 omap_free_dma(prtd->dma_ch);
125 prtd->dma_data = NULL; 139 prtd->dma_data = NULL;
126 140
@@ -148,7 +162,7 @@ static int omap_pcm_prepare(struct snd_pcm_substream *substream)
148 */ 162 */
149 dma_params.data_type = OMAP_DMA_DATA_TYPE_S16; 163 dma_params.data_type = OMAP_DMA_DATA_TYPE_S16;
150 dma_params.trigger = dma_data->dma_req; 164 dma_params.trigger = dma_data->dma_req;
151 dma_params.sync_mode = OMAP_DMA_SYNC_ELEMENT; 165 dma_params.sync_mode = dma_data->sync_mode;
152 if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { 166 if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
153 dma_params.src_amode = OMAP_DMA_AMODE_POST_INC; 167 dma_params.src_amode = OMAP_DMA_AMODE_POST_INC;
154 dma_params.dst_amode = OMAP_DMA_AMODE_CONSTANT; 168 dma_params.dst_amode = OMAP_DMA_AMODE_CONSTANT;
@@ -174,7 +188,15 @@ static int omap_pcm_prepare(struct snd_pcm_substream *substream)
174 dma_params.frame_count = runtime->periods; 188 dma_params.frame_count = runtime->periods;
175 omap_set_dma_params(prtd->dma_ch, &dma_params); 189 omap_set_dma_params(prtd->dma_ch, &dma_params);
176 190
177 omap_enable_dma_irq(prtd->dma_ch, OMAP_DMA_FRAME_IRQ); 191 if ((cpu_is_omap1510()) &&
192 (substream->stream == SNDRV_PCM_STREAM_PLAYBACK))
193 omap_enable_dma_irq(prtd->dma_ch, OMAP_DMA_FRAME_IRQ |
194 OMAP_DMA_LAST_IRQ | OMAP_DMA_BLOCK_IRQ);
195 else
196 omap_enable_dma_irq(prtd->dma_ch, OMAP_DMA_FRAME_IRQ);
197
198 omap_set_dma_src_burst_mode(prtd->dma_ch, OMAP_DMA_DATA_BURST_16);
199 omap_set_dma_dest_burst_mode(prtd->dma_ch, OMAP_DMA_DATA_BURST_16);
178 200
179 return 0; 201 return 0;
180} 202}
@@ -183,6 +205,7 @@ static int omap_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
183{ 205{
184 struct snd_pcm_runtime *runtime = substream->runtime; 206 struct snd_pcm_runtime *runtime = substream->runtime;
185 struct omap_runtime_data *prtd = runtime->private_data; 207 struct omap_runtime_data *prtd = runtime->private_data;
208 struct omap_pcm_dma_data *dma_data = prtd->dma_data;
186 unsigned long flags; 209 unsigned long flags;
187 int ret = 0; 210 int ret = 0;
188 211
@@ -192,6 +215,10 @@ static int omap_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
192 case SNDRV_PCM_TRIGGER_RESUME: 215 case SNDRV_PCM_TRIGGER_RESUME:
193 case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: 216 case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
194 prtd->period_index = 0; 217 prtd->period_index = 0;
218 /* Configure McBSP internal buffer usage */
219 if (dma_data->set_threshold)
220 dma_data->set_threshold(substream);
221
195 omap_start_dma(prtd->dma_ch); 222 omap_start_dma(prtd->dma_ch);
196 break; 223 break;
197 224
@@ -288,7 +315,7 @@ static struct snd_pcm_ops omap_pcm_ops = {
288 .mmap = omap_pcm_mmap, 315 .mmap = omap_pcm_mmap,
289}; 316};
290 317
291static u64 omap_pcm_dmamask = DMA_BIT_MASK(32); 318static u64 omap_pcm_dmamask = DMA_BIT_MASK(64);
292 319
293static int omap_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, 320static int omap_pcm_preallocate_dma_buffer(struct snd_pcm *pcm,
294 int stream) 321 int stream)
@@ -330,7 +357,7 @@ static void omap_pcm_free_dma_buffers(struct snd_pcm *pcm)
330 } 357 }
331} 358}
332 359
333int omap_pcm_new(struct snd_card *card, struct snd_soc_dai *dai, 360static int omap_pcm_new(struct snd_card *card, struct snd_soc_dai *dai,
334 struct snd_pcm *pcm) 361 struct snd_pcm *pcm)
335{ 362{
336 int ret = 0; 363 int ret = 0;
@@ -338,7 +365,7 @@ int omap_pcm_new(struct snd_card *card, struct snd_soc_dai *dai,
338 if (!card->dev->dma_mask) 365 if (!card->dev->dma_mask)
339 card->dev->dma_mask = &omap_pcm_dmamask; 366 card->dev->dma_mask = &omap_pcm_dmamask;
340 if (!card->dev->coherent_dma_mask) 367 if (!card->dev->coherent_dma_mask)
341 card->dev->coherent_dma_mask = DMA_BIT_MASK(32); 368 card->dev->coherent_dma_mask = DMA_BIT_MASK(64);
342 369
343 if (dai->playback.channels_min) { 370 if (dai->playback.channels_min) {
344 ret = omap_pcm_preallocate_dma_buffer(pcm, 371 ret = omap_pcm_preallocate_dma_buffer(pcm,
diff --git a/sound/soc/omap/omap-pcm.h b/sound/soc/omap/omap-pcm.h
index 8d9d26916b05..38a821dd4118 100644
--- a/sound/soc/omap/omap-pcm.h
+++ b/sound/soc/omap/omap-pcm.h
@@ -29,6 +29,8 @@ struct omap_pcm_dma_data {
29 char *name; /* stream identifier */ 29 char *name; /* stream identifier */
30 int dma_req; /* DMA request line */ 30 int dma_req; /* DMA request line */
31 unsigned long port_addr; /* transmit/receive register */ 31 unsigned long port_addr; /* transmit/receive register */
32 int sync_mode; /* DMA sync mode */
33 void (*set_threshold)(struct snd_pcm_substream *substream);
32}; 34};
33 35
34extern struct snd_soc_platform omap_soc_platform; 36extern struct snd_soc_platform omap_soc_platform;
diff --git a/sound/soc/omap/sdp3430.c b/sound/soc/omap/sdp3430.c
index b719e5db4f57..4a3f62d1f295 100644
--- a/sound/soc/omap/sdp3430.c
+++ b/sound/soc/omap/sdp3430.c
@@ -24,6 +24,7 @@
24 24
25#include <linux/clk.h> 25#include <linux/clk.h>
26#include <linux/platform_device.h> 26#include <linux/platform_device.h>
27#include <linux/i2c/twl4030.h>
27#include <sound/core.h> 28#include <sound/core.h>
28#include <sound/pcm.h> 29#include <sound/pcm.h>
29#include <sound/soc.h> 30#include <sound/soc.h>
@@ -39,6 +40,11 @@
39#include "omap-pcm.h" 40#include "omap-pcm.h"
40#include "../codecs/twl4030.h" 41#include "../codecs/twl4030.h"
41 42
43/* TWL4030 PMBR1 Register */
44#define TWL4030_INTBR_PMBR1 0x0D
45/* TWL4030 PMBR1 Register GPIO6 mux bit */
46#define TWL4030_GPIO6_PWM0_MUTE(value) (value << 2)
47
42static struct snd_soc_card snd_soc_sdp3430; 48static struct snd_soc_card snd_soc_sdp3430;
43 49
44static int sdp3430_hw_params(struct snd_pcm_substream *substream, 50static int sdp3430_hw_params(struct snd_pcm_substream *substream,
@@ -96,7 +102,7 @@ static int sdp3430_hw_voice_params(struct snd_pcm_substream *substream,
96 ret = snd_soc_dai_set_fmt(codec_dai, 102 ret = snd_soc_dai_set_fmt(codec_dai,
97 SND_SOC_DAIFMT_DSP_A | 103 SND_SOC_DAIFMT_DSP_A |
98 SND_SOC_DAIFMT_IB_NF | 104 SND_SOC_DAIFMT_IB_NF |
99 SND_SOC_DAIFMT_CBS_CFM); 105 SND_SOC_DAIFMT_CBM_CFM);
100 if (ret) { 106 if (ret) {
101 printk(KERN_ERR "can't set codec DAI configuration\n"); 107 printk(KERN_ERR "can't set codec DAI configuration\n");
102 return ret; 108 return ret;
@@ -280,6 +286,7 @@ static struct snd_soc_card snd_soc_sdp3430 = {
280static struct twl4030_setup_data twl4030_setup = { 286static struct twl4030_setup_data twl4030_setup = {
281 .ramp_delay_value = 3, 287 .ramp_delay_value = 3,
282 .sysclk = 26000, 288 .sysclk = 26000,
289 .hs_extmute = 1,
283}; 290};
284 291
285/* Audio subsystem */ 292/* Audio subsystem */
@@ -294,6 +301,7 @@ static struct platform_device *sdp3430_snd_device;
294static int __init sdp3430_soc_init(void) 301static int __init sdp3430_soc_init(void)
295{ 302{
296 int ret; 303 int ret;
304 u8 pin_mux;
297 305
298 if (!machine_is_omap_3430sdp()) { 306 if (!machine_is_omap_3430sdp()) {
299 pr_debug("Not SDP3430!\n"); 307 pr_debug("Not SDP3430!\n");
@@ -312,6 +320,14 @@ static int __init sdp3430_soc_init(void)
312 *(unsigned int *)sdp3430_dai[0].cpu_dai->private_data = 1; /* McBSP2 */ 320 *(unsigned int *)sdp3430_dai[0].cpu_dai->private_data = 1; /* McBSP2 */
313 *(unsigned int *)sdp3430_dai[1].cpu_dai->private_data = 2; /* McBSP3 */ 321 *(unsigned int *)sdp3430_dai[1].cpu_dai->private_data = 2; /* McBSP3 */
314 322
323 /* Set TWL4030 GPIO6 as EXTMUTE signal */
324 twl4030_i2c_read_u8(TWL4030_MODULE_INTBR, &pin_mux,
325 TWL4030_INTBR_PMBR1);
326 pin_mux &= ~TWL4030_GPIO6_PWM0_MUTE(0x03);
327 pin_mux |= TWL4030_GPIO6_PWM0_MUTE(0x02);
328 twl4030_i2c_write_u8(TWL4030_MODULE_INTBR, pin_mux,
329 TWL4030_INTBR_PMBR1);
330
315 ret = platform_device_add(sdp3430_snd_device); 331 ret = platform_device_add(sdp3430_snd_device);
316 if (ret) 332 if (ret)
317 goto err1; 333 goto err1;
diff --git a/sound/soc/omap/zoom2.c b/sound/soc/omap/zoom2.c
new file mode 100644
index 000000000000..f90b45f56220
--- /dev/null
+++ b/sound/soc/omap/zoom2.c
@@ -0,0 +1,314 @@
1/*
2 * zoom2.c -- SoC audio for Zoom2
3 *
4 * Author: Misael Lopez Cruz <x0052729@ti.com>
5 *
6 * This program is free software; you can redistribute it and/or
7 * modify it under the terms of the GNU General Public License
8 * version 2 as published by the Free Software Foundation.
9 *
10 * This program is distributed in the hope that it will be useful, but
11 * WITHOUT ANY WARRANTY; without even the implied warranty of
12 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
13 * General Public License for more details.
14 *
15 * You should have received a copy of the GNU General Public License
16 * along with this program; if not, write to the Free Software
17 * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
18 * 02110-1301 USA
19 *
20 */
21
22#include <linux/clk.h>
23#include <linux/platform_device.h>
24#include <sound/core.h>
25#include <sound/pcm.h>
26#include <sound/soc.h>
27#include <sound/soc-dapm.h>
28
29#include <asm/mach-types.h>
30#include <mach/hardware.h>
31#include <mach/gpio.h>
32#include <mach/mcbsp.h>
33
34#include "omap-mcbsp.h"
35#include "omap-pcm.h"
36#include "../codecs/twl4030.h"
37
38#define ZOOM2_HEADSET_MUX_GPIO (OMAP_MAX_GPIO_LINES + 15)
39#define ZOOM2_HEADSET_EXTMUTE_GPIO 153
40
41static int zoom2_hw_params(struct snd_pcm_substream *substream,
42 struct snd_pcm_hw_params *params)
43{
44 struct snd_soc_pcm_runtime *rtd = substream->private_data;
45 struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
46 struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
47 int ret;
48
49 /* Set codec DAI configuration */
50 ret = snd_soc_dai_set_fmt(codec_dai,
51 SND_SOC_DAIFMT_I2S |
52 SND_SOC_DAIFMT_NB_NF |
53 SND_SOC_DAIFMT_CBM_CFM);
54 if (ret < 0) {
55 printk(KERN_ERR "can't set codec DAI configuration\n");
56 return ret;
57 }
58
59 /* Set cpu DAI configuration */
60 ret = snd_soc_dai_set_fmt(cpu_dai,
61 SND_SOC_DAIFMT_I2S |
62 SND_SOC_DAIFMT_NB_NF |
63 SND_SOC_DAIFMT_CBM_CFM);
64 if (ret < 0) {
65 printk(KERN_ERR "can't set cpu DAI configuration\n");
66 return ret;
67 }
68
69 /* Set the codec system clock for DAC and ADC */
70 ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000,
71 SND_SOC_CLOCK_IN);
72 if (ret < 0) {
73 printk(KERN_ERR "can't set codec system clock\n");
74 return ret;
75 }
76
77 return 0;
78}
79
80static struct snd_soc_ops zoom2_ops = {
81 .hw_params = zoom2_hw_params,
82};
83
84static int zoom2_hw_voice_params(struct snd_pcm_substream *substream,
85 struct snd_pcm_hw_params *params)
86{
87 struct snd_soc_pcm_runtime *rtd = substream->private_data;
88 struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
89 struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
90 int ret;
91
92 /* Set codec DAI configuration */
93 ret = snd_soc_dai_set_fmt(codec_dai,
94 SND_SOC_DAIFMT_DSP_A |
95 SND_SOC_DAIFMT_IB_NF |
96 SND_SOC_DAIFMT_CBM_CFM);
97 if (ret) {
98 printk(KERN_ERR "can't set codec DAI configuration\n");
99 return ret;
100 }
101
102 /* Set cpu DAI configuration */
103 ret = snd_soc_dai_set_fmt(cpu_dai,
104 SND_SOC_DAIFMT_DSP_A |
105 SND_SOC_DAIFMT_IB_NF |
106 SND_SOC_DAIFMT_CBM_CFM);
107 if (ret < 0) {
108 printk(KERN_ERR "can't set cpu DAI configuration\n");
109 return ret;
110 }
111
112 /* Set the codec system clock for DAC and ADC */
113 ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000,
114 SND_SOC_CLOCK_IN);
115 if (ret < 0) {
116 printk(KERN_ERR "can't set codec system clock\n");
117 return ret;
118 }
119
120 return 0;
121}
122
123static struct snd_soc_ops zoom2_voice_ops = {
124 .hw_params = zoom2_hw_voice_params,
125};
126
127/* Zoom2 machine DAPM */
128static const struct snd_soc_dapm_widget zoom2_twl4030_dapm_widgets[] = {
129 SND_SOC_DAPM_MIC("Ext Mic", NULL),
130 SND_SOC_DAPM_SPK("Ext Spk", NULL),
131 SND_SOC_DAPM_MIC("Headset Mic", NULL),
132 SND_SOC_DAPM_HP("Headset Stereophone", NULL),
133 SND_SOC_DAPM_LINE("Aux In", NULL),
134};
135
136static const struct snd_soc_dapm_route audio_map[] = {
137 /* External Mics: MAINMIC, SUBMIC with bias*/
138 {"MAINMIC", NULL, "Mic Bias 1"},
139 {"SUBMIC", NULL, "Mic Bias 2"},
140 {"Mic Bias 1", NULL, "Ext Mic"},
141 {"Mic Bias 2", NULL, "Ext Mic"},
142
143 /* External Speakers: HFL, HFR */
144 {"Ext Spk", NULL, "HFL"},
145 {"Ext Spk", NULL, "HFR"},
146
147 /* Headset Stereophone: HSOL, HSOR */
148 {"Headset Stereophone", NULL, "HSOL"},
149 {"Headset Stereophone", NULL, "HSOR"},
150
151 /* Headset Mic: HSMIC with bias */
152 {"HSMIC", NULL, "Headset Mic Bias"},
153 {"Headset Mic Bias", NULL, "Headset Mic"},
154
155 /* Aux In: AUXL, AUXR */
156 {"Aux In", NULL, "AUXL"},
157 {"Aux In", NULL, "AUXR"},
158};
159
160static int zoom2_twl4030_init(struct snd_soc_codec *codec)
161{
162 int ret;
163
164 /* Add Zoom2 specific widgets */
165 ret = snd_soc_dapm_new_controls(codec, zoom2_twl4030_dapm_widgets,
166 ARRAY_SIZE(zoom2_twl4030_dapm_widgets));
167 if (ret)
168 return ret;
169
170 /* Set up Zoom2 specific audio path audio_map */
171 snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
172
173 /* Zoom2 connected pins */
174 snd_soc_dapm_enable_pin(codec, "Ext Mic");
175 snd_soc_dapm_enable_pin(codec, "Ext Spk");
176 snd_soc_dapm_enable_pin(codec, "Headset Mic");
177 snd_soc_dapm_enable_pin(codec, "Headset Stereophone");
178 snd_soc_dapm_enable_pin(codec, "Aux In");
179
180 /* TWL4030 not connected pins */
181 snd_soc_dapm_nc_pin(codec, "CARKITMIC");
182 snd_soc_dapm_nc_pin(codec, "DIGIMIC0");
183 snd_soc_dapm_nc_pin(codec, "DIGIMIC1");
184
185 snd_soc_dapm_nc_pin(codec, "OUTL");
186 snd_soc_dapm_nc_pin(codec, "OUTR");
187 snd_soc_dapm_nc_pin(codec, "EARPIECE");
188 snd_soc_dapm_nc_pin(codec, "PREDRIVEL");
189 snd_soc_dapm_nc_pin(codec, "PREDRIVER");
190 snd_soc_dapm_nc_pin(codec, "CARKITL");
191 snd_soc_dapm_nc_pin(codec, "CARKITR");
192
193 ret = snd_soc_dapm_sync(codec);
194
195 return ret;
196}
197
198static int zoom2_twl4030_voice_init(struct snd_soc_codec *codec)
199{
200 unsigned short reg;
201
202 /* Enable voice interface */
203 reg = codec->read(codec, TWL4030_REG_VOICE_IF);
204 reg |= TWL4030_VIF_DIN_EN | TWL4030_VIF_DOUT_EN | TWL4030_VIF_EN;
205 codec->write(codec, TWL4030_REG_VOICE_IF, reg);
206
207 return 0;
208}
209
210/* Digital audio interface glue - connects codec <--> CPU */
211static struct snd_soc_dai_link zoom2_dai[] = {
212 {
213 .name = "TWL4030 I2S",
214 .stream_name = "TWL4030 Audio",
215 .cpu_dai = &omap_mcbsp_dai[0],
216 .codec_dai = &twl4030_dai[TWL4030_DAI_HIFI],
217 .init = zoom2_twl4030_init,
218 .ops = &zoom2_ops,
219 },
220 {
221 .name = "TWL4030 PCM",
222 .stream_name = "TWL4030 Voice",
223 .cpu_dai = &omap_mcbsp_dai[1],
224 .codec_dai = &twl4030_dai[TWL4030_DAI_VOICE],
225 .init = zoom2_twl4030_voice_init,
226 .ops = &zoom2_voice_ops,
227 },
228};
229
230/* Audio machine driver */
231static struct snd_soc_card snd_soc_zoom2 = {
232 .name = "Zoom2",
233 .platform = &omap_soc_platform,
234 .dai_link = zoom2_dai,
235 .num_links = ARRAY_SIZE(zoom2_dai),
236};
237
238/* EXTMUTE callback function */
239void zoom2_set_hs_extmute(int mute)
240{
241 gpio_set_value(ZOOM2_HEADSET_EXTMUTE_GPIO, mute);
242}
243
244/* twl4030 setup */
245static struct twl4030_setup_data twl4030_setup = {
246 .ramp_delay_value = 3, /* 161 ms */
247 .sysclk = 26000,
248 .hs_extmute = 1,
249 .set_hs_extmute = zoom2_set_hs_extmute,
250};
251
252/* Audio subsystem */
253static struct snd_soc_device zoom2_snd_devdata = {
254 .card = &snd_soc_zoom2,
255 .codec_dev = &soc_codec_dev_twl4030,
256 .codec_data = &twl4030_setup,
257};
258
259static struct platform_device *zoom2_snd_device;
260
261static int __init zoom2_soc_init(void)
262{
263 int ret;
264
265 if (!machine_is_omap_zoom2()) {
266 pr_debug("Not Zoom2!\n");
267 return -ENODEV;
268 }
269 printk(KERN_INFO "Zoom2 SoC init\n");
270
271 zoom2_snd_device = platform_device_alloc("soc-audio", -1);
272 if (!zoom2_snd_device) {
273 printk(KERN_ERR "Platform device allocation failed\n");
274 return -ENOMEM;
275 }
276
277 platform_set_drvdata(zoom2_snd_device, &zoom2_snd_devdata);
278 zoom2_snd_devdata.dev = &zoom2_snd_device->dev;
279 *(unsigned int *)zoom2_dai[0].cpu_dai->private_data = 1; /* McBSP2 */
280 *(unsigned int *)zoom2_dai[1].cpu_dai->private_data = 2; /* McBSP3 */
281
282 ret = platform_device_add(zoom2_snd_device);
283 if (ret)
284 goto err1;
285
286 BUG_ON(gpio_request(ZOOM2_HEADSET_MUX_GPIO, "hs_mux") < 0);
287 gpio_direction_output(ZOOM2_HEADSET_MUX_GPIO, 0);
288
289 BUG_ON(gpio_request(ZOOM2_HEADSET_EXTMUTE_GPIO, "ext_mute") < 0);
290 gpio_direction_output(ZOOM2_HEADSET_EXTMUTE_GPIO, 0);
291
292 return 0;
293
294err1:
295 printk(KERN_ERR "Unable to add platform device\n");
296 platform_device_put(zoom2_snd_device);
297
298 return ret;
299}
300module_init(zoom2_soc_init);
301
302static void __exit zoom2_soc_exit(void)
303{
304 gpio_free(ZOOM2_HEADSET_MUX_GPIO);
305 gpio_free(ZOOM2_HEADSET_EXTMUTE_GPIO);
306
307 platform_device_unregister(zoom2_snd_device);
308}
309module_exit(zoom2_soc_exit);
310
311MODULE_AUTHOR("Misael Lopez Cruz <x0052729@ti.com>");
312MODULE_DESCRIPTION("ALSA SoC Zoom2");
313MODULE_LICENSE("GPL");
314
generated by cgit v1.2.2 (git 2.25.0) at 2024-11-13 00:56:55 -0500