Merge tag 'cleanup-3.15' of git://git.kernel.org/pub/scm/linux/kernel/git/arm/arm-soc
[deliverable/linux.git] / sound / soc / codecs / ak4642.c
1 /*
2 * ak4642.c -- AK4642/AK4643 ALSA Soc Audio driver
3 *
4 * Copyright (C) 2009 Renesas Solutions Corp.
5 * Kuninori Morimoto <morimoto.kuninori@renesas.com>
6 *
7 * Based on wm8731.c by Richard Purdie
8 * Based on ak4535.c by Richard Purdie
9 * Based on wm8753.c by Liam Girdwood
10 *
11 * This program is free software; you can redistribute it and/or modify
12 * it under the terms of the GNU General Public License version 2 as
13 * published by the Free Software Foundation.
14 */
15
16 /* ** CAUTION **
17 *
18 * This is very simple driver.
19 * It can use headphone output / stereo input only
20 *
21 * AK4642 is tested.
22 * AK4643 is tested.
23 * AK4648 is tested.
24 */
25
26 #include <linux/delay.h>
27 #include <linux/i2c.h>
28 #include <linux/slab.h>
29 #include <linux/of_device.h>
30 #include <linux/module.h>
31 #include <linux/regmap.h>
32 #include <sound/soc.h>
33 #include <sound/initval.h>
34 #include <sound/tlv.h>
35
36 #define PW_MGMT1 0x00
37 #define PW_MGMT2 0x01
38 #define SG_SL1 0x02
39 #define SG_SL2 0x03
40 #define MD_CTL1 0x04
41 #define MD_CTL2 0x05
42 #define TIMER 0x06
43 #define ALC_CTL1 0x07
44 #define ALC_CTL2 0x08
45 #define L_IVC 0x09
46 #define L_DVC 0x0a
47 #define ALC_CTL3 0x0b
48 #define R_IVC 0x0c
49 #define R_DVC 0x0d
50 #define MD_CTL3 0x0e
51 #define MD_CTL4 0x0f
52 #define PW_MGMT3 0x10
53 #define DF_S 0x11
54 #define FIL3_0 0x12
55 #define FIL3_1 0x13
56 #define FIL3_2 0x14
57 #define FIL3_3 0x15
58 #define EQ_0 0x16
59 #define EQ_1 0x17
60 #define EQ_2 0x18
61 #define EQ_3 0x19
62 #define EQ_4 0x1a
63 #define EQ_5 0x1b
64 #define FIL1_0 0x1c
65 #define FIL1_1 0x1d
66 #define FIL1_2 0x1e
67 #define FIL1_3 0x1f
68 #define PW_MGMT4 0x20
69 #define MD_CTL5 0x21
70 #define LO_MS 0x22
71 #define HP_MS 0x23
72 #define SPK_MS 0x24
73
74 /* PW_MGMT1*/
75 #define PMVCM (1 << 6) /* VCOM Power Management */
76 #define PMMIN (1 << 5) /* MIN Input Power Management */
77 #define PMDAC (1 << 2) /* DAC Power Management */
78 #define PMADL (1 << 0) /* MIC Amp Lch and ADC Lch Power Management */
79
80 /* PW_MGMT2 */
81 #define HPMTN (1 << 6)
82 #define PMHPL (1 << 5)
83 #define PMHPR (1 << 4)
84 #define MS (1 << 3) /* master/slave select */
85 #define MCKO (1 << 1)
86 #define PMPLL (1 << 0)
87
88 #define PMHP_MASK (PMHPL | PMHPR)
89 #define PMHP PMHP_MASK
90
91 /* PW_MGMT3 */
92 #define PMADR (1 << 0) /* MIC L / ADC R Power Management */
93
94 /* SG_SL1 */
95 #define MINS (1 << 6) /* Switch from MIN to Speaker */
96 #define DACL (1 << 4) /* Switch from DAC to Stereo or Receiver */
97 #define PMMP (1 << 2) /* MPWR pin Power Management */
98 #define MGAIN0 (1 << 0) /* MIC amp gain*/
99
100 /* TIMER */
101 #define ZTM(param) ((param & 0x3) << 4) /* ALC Zoro Crossing TimeOut */
102 #define WTM(param) (((param & 0x4) << 4) | ((param & 0x3) << 2))
103
104 /* ALC_CTL1 */
105 #define ALC (1 << 5) /* ALC Enable */
106 #define LMTH0 (1 << 0) /* ALC Limiter / Recovery Level */
107
108 /* MD_CTL1 */
109 #define PLL3 (1 << 7)
110 #define PLL2 (1 << 6)
111 #define PLL1 (1 << 5)
112 #define PLL0 (1 << 4)
113 #define PLL_MASK (PLL3 | PLL2 | PLL1 | PLL0)
114
115 #define BCKO_MASK (1 << 3)
116 #define BCKO_64 BCKO_MASK
117
118 #define DIF_MASK (3 << 0)
119 #define DSP (0 << 0)
120 #define RIGHT_J (1 << 0)
121 #define LEFT_J (2 << 0)
122 #define I2S (3 << 0)
123
124 /* MD_CTL2 */
125 #define FS0 (1 << 0)
126 #define FS1 (1 << 1)
127 #define FS2 (1 << 2)
128 #define FS3 (1 << 5)
129 #define FS_MASK (FS0 | FS1 | FS2 | FS3)
130
131 /* MD_CTL3 */
132 #define BST1 (1 << 3)
133
134 /* MD_CTL4 */
135 #define DACH (1 << 0)
136
137 /*
138 * Playback Volume (table 39)
139 *
140 * max : 0x00 : +12.0 dB
141 * ( 0.5 dB step )
142 * min : 0xFE : -115.0 dB
143 * mute: 0xFF
144 */
145 static const DECLARE_TLV_DB_SCALE(out_tlv, -11550, 50, 1);
146
147 static const struct snd_kcontrol_new ak4642_snd_controls[] = {
148
149 SOC_DOUBLE_R_TLV("Digital Playback Volume", L_DVC, R_DVC,
150 0, 0xFF, 1, out_tlv),
151 };
152
153 static const struct snd_kcontrol_new ak4642_headphone_control =
154 SOC_DAPM_SINGLE("Switch", PW_MGMT2, 6, 1, 0);
155
156 static const struct snd_kcontrol_new ak4642_lout_mixer_controls[] = {
157 SOC_DAPM_SINGLE("DACL", SG_SL1, 4, 1, 0),
158 };
159
160 static const struct snd_soc_dapm_widget ak4642_dapm_widgets[] = {
161
162 /* Outputs */
163 SND_SOC_DAPM_OUTPUT("HPOUTL"),
164 SND_SOC_DAPM_OUTPUT("HPOUTR"),
165 SND_SOC_DAPM_OUTPUT("LINEOUT"),
166
167 SND_SOC_DAPM_PGA("HPL Out", PW_MGMT2, 5, 0, NULL, 0),
168 SND_SOC_DAPM_PGA("HPR Out", PW_MGMT2, 4, 0, NULL, 0),
169 SND_SOC_DAPM_SWITCH("Headphone Enable", SND_SOC_NOPM, 0, 0,
170 &ak4642_headphone_control),
171
172 SND_SOC_DAPM_PGA("DACH", MD_CTL4, 0, 0, NULL, 0),
173
174 SND_SOC_DAPM_MIXER("LINEOUT Mixer", PW_MGMT1, 3, 0,
175 &ak4642_lout_mixer_controls[0],
176 ARRAY_SIZE(ak4642_lout_mixer_controls)),
177
178 /* DAC */
179 SND_SOC_DAPM_DAC("DAC", "HiFi Playback", PW_MGMT1, 2, 0),
180 };
181
182 static const struct snd_soc_dapm_route ak4642_intercon[] = {
183
184 /* Outputs */
185 {"HPOUTL", NULL, "HPL Out"},
186 {"HPOUTR", NULL, "HPR Out"},
187 {"LINEOUT", NULL, "LINEOUT Mixer"},
188
189 {"HPL Out", NULL, "Headphone Enable"},
190 {"HPR Out", NULL, "Headphone Enable"},
191
192 {"Headphone Enable", "Switch", "DACH"},
193
194 {"DACH", NULL, "DAC"},
195
196 {"LINEOUT Mixer", "DACL", "DAC"},
197 };
198
199 /*
200 * ak4642 register cache
201 */
202 static const struct reg_default ak4642_reg[] = {
203 { 0, 0x00 }, { 1, 0x00 }, { 2, 0x01 }, { 3, 0x00 },
204 { 4, 0x02 }, { 5, 0x00 }, { 6, 0x00 }, { 7, 0x00 },
205 { 8, 0xe1 }, { 9, 0xe1 }, { 10, 0x18 }, { 11, 0x00 },
206 { 12, 0xe1 }, { 13, 0x18 }, { 14, 0x11 }, { 15, 0x08 },
207 { 16, 0x00 }, { 17, 0x00 }, { 18, 0x00 }, { 19, 0x00 },
208 { 20, 0x00 }, { 21, 0x00 }, { 22, 0x00 }, { 23, 0x00 },
209 { 24, 0x00 }, { 25, 0x00 }, { 26, 0x00 }, { 27, 0x00 },
210 { 28, 0x00 }, { 29, 0x00 }, { 30, 0x00 }, { 31, 0x00 },
211 { 32, 0x00 }, { 33, 0x00 }, { 34, 0x00 }, { 35, 0x00 },
212 { 36, 0x00 },
213 };
214
215 static const struct reg_default ak4648_reg[] = {
216 { 0, 0x00 }, { 1, 0x00 }, { 2, 0x01 }, { 3, 0x00 },
217 { 4, 0x02 }, { 5, 0x00 }, { 6, 0x00 }, { 7, 0x00 },
218 { 8, 0xe1 }, { 9, 0xe1 }, { 10, 0x18 }, { 11, 0x00 },
219 { 12, 0xe1 }, { 13, 0x18 }, { 14, 0x11 }, { 15, 0xb8 },
220 { 16, 0x00 }, { 17, 0x00 }, { 18, 0x00 }, { 19, 0x00 },
221 { 20, 0x00 }, { 21, 0x00 }, { 22, 0x00 }, { 23, 0x00 },
222 { 24, 0x00 }, { 25, 0x00 }, { 26, 0x00 }, { 27, 0x00 },
223 { 28, 0x00 }, { 29, 0x00 }, { 30, 0x00 }, { 31, 0x00 },
224 { 32, 0x00 }, { 33, 0x00 }, { 34, 0x00 }, { 35, 0x00 },
225 { 36, 0x00 }, { 37, 0x88 }, { 38, 0x88 }, { 39, 0x08 },
226 };
227
228 static int ak4642_dai_startup(struct snd_pcm_substream *substream,
229 struct snd_soc_dai *dai)
230 {
231 int is_play = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
232 struct snd_soc_codec *codec = dai->codec;
233
234 if (is_play) {
235 /*
236 * start headphone output
237 *
238 * PLL, Master Mode
239 * Audio I/F Format :MSB justified (ADC & DAC)
240 * Bass Boost Level : Middle
241 *
242 * This operation came from example code of
243 * "ASAHI KASEI AK4642" (japanese) manual p97.
244 */
245 snd_soc_write(codec, L_IVC, 0x91); /* volume */
246 snd_soc_write(codec, R_IVC, 0x91); /* volume */
247 } else {
248 /*
249 * start stereo input
250 *
251 * PLL Master Mode
252 * Audio I/F Format:MSB justified (ADC & DAC)
253 * Pre MIC AMP:+20dB
254 * MIC Power On
255 * ALC setting:Refer to Table 35
256 * ALC bit=“1”
257 *
258 * This operation came from example code of
259 * "ASAHI KASEI AK4642" (japanese) manual p94.
260 */
261 snd_soc_update_bits(codec, SG_SL1, PMMP | MGAIN0, PMMP | MGAIN0);
262 snd_soc_write(codec, TIMER, ZTM(0x3) | WTM(0x3));
263 snd_soc_write(codec, ALC_CTL1, ALC | LMTH0);
264 snd_soc_update_bits(codec, PW_MGMT1, PMADL, PMADL);
265 snd_soc_update_bits(codec, PW_MGMT3, PMADR, PMADR);
266 }
267
268 return 0;
269 }
270
271 static void ak4642_dai_shutdown(struct snd_pcm_substream *substream,
272 struct snd_soc_dai *dai)
273 {
274 int is_play = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
275 struct snd_soc_codec *codec = dai->codec;
276
277 if (is_play) {
278 } else {
279 /* stop stereo input */
280 snd_soc_update_bits(codec, PW_MGMT1, PMADL, 0);
281 snd_soc_update_bits(codec, PW_MGMT3, PMADR, 0);
282 snd_soc_update_bits(codec, ALC_CTL1, ALC, 0);
283 }
284 }
285
286 static int ak4642_dai_set_sysclk(struct snd_soc_dai *codec_dai,
287 int clk_id, unsigned int freq, int dir)
288 {
289 struct snd_soc_codec *codec = codec_dai->codec;
290 u8 pll;
291
292 switch (freq) {
293 case 11289600:
294 pll = PLL2;
295 break;
296 case 12288000:
297 pll = PLL2 | PLL0;
298 break;
299 case 12000000:
300 pll = PLL2 | PLL1;
301 break;
302 case 24000000:
303 pll = PLL2 | PLL1 | PLL0;
304 break;
305 case 13500000:
306 pll = PLL3 | PLL2;
307 break;
308 case 27000000:
309 pll = PLL3 | PLL2 | PLL0;
310 break;
311 default:
312 return -EINVAL;
313 }
314 snd_soc_update_bits(codec, MD_CTL1, PLL_MASK, pll);
315
316 return 0;
317 }
318
319 static int ak4642_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
320 {
321 struct snd_soc_codec *codec = dai->codec;
322 u8 data;
323 u8 bcko;
324
325 data = MCKO | PMPLL; /* use MCKO */
326 bcko = 0;
327
328 /* set master/slave audio interface */
329 switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
330 case SND_SOC_DAIFMT_CBM_CFM:
331 data |= MS;
332 bcko = BCKO_64;
333 break;
334 case SND_SOC_DAIFMT_CBS_CFS:
335 break;
336 default:
337 return -EINVAL;
338 }
339 snd_soc_update_bits(codec, PW_MGMT2, MS | MCKO | PMPLL, data);
340 snd_soc_update_bits(codec, MD_CTL1, BCKO_MASK, bcko);
341
342 /* format type */
343 data = 0;
344 switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
345 case SND_SOC_DAIFMT_LEFT_J:
346 data = LEFT_J;
347 break;
348 case SND_SOC_DAIFMT_I2S:
349 data = I2S;
350 break;
351 /* FIXME
352 * Please add RIGHT_J / DSP support here
353 */
354 default:
355 return -EINVAL;
356 }
357 snd_soc_update_bits(codec, MD_CTL1, DIF_MASK, data);
358
359 return 0;
360 }
361
362 static int ak4642_dai_hw_params(struct snd_pcm_substream *substream,
363 struct snd_pcm_hw_params *params,
364 struct snd_soc_dai *dai)
365 {
366 struct snd_soc_codec *codec = dai->codec;
367 u8 rate;
368
369 switch (params_rate(params)) {
370 case 7350:
371 rate = FS2;
372 break;
373 case 8000:
374 rate = 0;
375 break;
376 case 11025:
377 rate = FS2 | FS0;
378 break;
379 case 12000:
380 rate = FS0;
381 break;
382 case 14700:
383 rate = FS2 | FS1;
384 break;
385 case 16000:
386 rate = FS1;
387 break;
388 case 22050:
389 rate = FS2 | FS1 | FS0;
390 break;
391 case 24000:
392 rate = FS1 | FS0;
393 break;
394 case 29400:
395 rate = FS3 | FS2 | FS1;
396 break;
397 case 32000:
398 rate = FS3 | FS1;
399 break;
400 case 44100:
401 rate = FS3 | FS2 | FS1 | FS0;
402 break;
403 case 48000:
404 rate = FS3 | FS1 | FS0;
405 break;
406 default:
407 return -EINVAL;
408 }
409 snd_soc_update_bits(codec, MD_CTL2, FS_MASK, rate);
410
411 return 0;
412 }
413
414 static int ak4642_set_bias_level(struct snd_soc_codec *codec,
415 enum snd_soc_bias_level level)
416 {
417 switch (level) {
418 case SND_SOC_BIAS_OFF:
419 snd_soc_write(codec, PW_MGMT1, 0x00);
420 break;
421 default:
422 snd_soc_update_bits(codec, PW_MGMT1, PMVCM, PMVCM);
423 break;
424 }
425 codec->dapm.bias_level = level;
426
427 return 0;
428 }
429
430 static const struct snd_soc_dai_ops ak4642_dai_ops = {
431 .startup = ak4642_dai_startup,
432 .shutdown = ak4642_dai_shutdown,
433 .set_sysclk = ak4642_dai_set_sysclk,
434 .set_fmt = ak4642_dai_set_fmt,
435 .hw_params = ak4642_dai_hw_params,
436 };
437
438 static struct snd_soc_dai_driver ak4642_dai = {
439 .name = "ak4642-hifi",
440 .playback = {
441 .stream_name = "Playback",
442 .channels_min = 1,
443 .channels_max = 2,
444 .rates = SNDRV_PCM_RATE_8000_48000,
445 .formats = SNDRV_PCM_FMTBIT_S16_LE },
446 .capture = {
447 .stream_name = "Capture",
448 .channels_min = 1,
449 .channels_max = 2,
450 .rates = SNDRV_PCM_RATE_8000_48000,
451 .formats = SNDRV_PCM_FMTBIT_S16_LE },
452 .ops = &ak4642_dai_ops,
453 .symmetric_rates = 1,
454 };
455
456 static int ak4642_resume(struct snd_soc_codec *codec)
457 {
458 struct regmap *regmap = dev_get_regmap(codec->dev, NULL);
459
460 regcache_mark_dirty(regmap);
461 regcache_sync(regmap);
462 return 0;
463 }
464
465
466 static int ak4642_probe(struct snd_soc_codec *codec)
467 {
468 ak4642_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
469
470 return 0;
471 }
472
473 static int ak4642_remove(struct snd_soc_codec *codec)
474 {
475 ak4642_set_bias_level(codec, SND_SOC_BIAS_OFF);
476 return 0;
477 }
478
479 static struct snd_soc_codec_driver soc_codec_dev_ak4642 = {
480 .probe = ak4642_probe,
481 .remove = ak4642_remove,
482 .resume = ak4642_resume,
483 .set_bias_level = ak4642_set_bias_level,
484 .controls = ak4642_snd_controls,
485 .num_controls = ARRAY_SIZE(ak4642_snd_controls),
486 .dapm_widgets = ak4642_dapm_widgets,
487 .num_dapm_widgets = ARRAY_SIZE(ak4642_dapm_widgets),
488 .dapm_routes = ak4642_intercon,
489 .num_dapm_routes = ARRAY_SIZE(ak4642_intercon),
490 };
491
492 static const struct regmap_config ak4642_regmap = {
493 .reg_bits = 8,
494 .val_bits = 8,
495 .max_register = ARRAY_SIZE(ak4642_reg) + 1,
496 .reg_defaults = ak4642_reg,
497 .num_reg_defaults = ARRAY_SIZE(ak4642_reg),
498 };
499
500 static const struct regmap_config ak4648_regmap = {
501 .reg_bits = 8,
502 .val_bits = 8,
503 .max_register = ARRAY_SIZE(ak4648_reg) + 1,
504 .reg_defaults = ak4648_reg,
505 .num_reg_defaults = ARRAY_SIZE(ak4648_reg),
506 };
507
508 static struct of_device_id ak4642_of_match[];
509 static int ak4642_i2c_probe(struct i2c_client *i2c,
510 const struct i2c_device_id *id)
511 {
512 struct device_node *np = i2c->dev.of_node;
513 const struct regmap_config *regmap_config = NULL;
514 struct regmap *regmap;
515
516 if (np) {
517 const struct of_device_id *of_id;
518
519 of_id = of_match_device(ak4642_of_match, &i2c->dev);
520 if (of_id)
521 regmap_config = of_id->data;
522 } else {
523 regmap_config = (const struct regmap_config *)id->driver_data;
524 }
525
526 if (!regmap_config) {
527 dev_err(&i2c->dev, "Unknown device type\n");
528 return -EINVAL;
529 }
530
531 regmap = devm_regmap_init_i2c(i2c, regmap_config);
532 if (IS_ERR(regmap))
533 return PTR_ERR(regmap);
534
535 return snd_soc_register_codec(&i2c->dev,
536 &soc_codec_dev_ak4642, &ak4642_dai, 1);
537 }
538
539 static int ak4642_i2c_remove(struct i2c_client *client)
540 {
541 snd_soc_unregister_codec(&client->dev);
542 return 0;
543 }
544
545 static struct of_device_id ak4642_of_match[] = {
546 { .compatible = "asahi-kasei,ak4642", .data = &ak4642_regmap},
547 { .compatible = "asahi-kasei,ak4643", .data = &ak4642_regmap},
548 { .compatible = "asahi-kasei,ak4648", .data = &ak4648_regmap},
549 {},
550 };
551 MODULE_DEVICE_TABLE(of, ak4642_of_match);
552
553 static const struct i2c_device_id ak4642_i2c_id[] = {
554 { "ak4642", (kernel_ulong_t)&ak4642_regmap },
555 { "ak4643", (kernel_ulong_t)&ak4642_regmap },
556 { "ak4648", (kernel_ulong_t)&ak4648_regmap },
557 { }
558 };
559 MODULE_DEVICE_TABLE(i2c, ak4642_i2c_id);
560
561 static struct i2c_driver ak4642_i2c_driver = {
562 .driver = {
563 .name = "ak4642-codec",
564 .owner = THIS_MODULE,
565 .of_match_table = ak4642_of_match,
566 },
567 .probe = ak4642_i2c_probe,
568 .remove = ak4642_i2c_remove,
569 .id_table = ak4642_i2c_id,
570 };
571
572 module_i2c_driver(ak4642_i2c_driver);
573
574 MODULE_DESCRIPTION("Soc AK4642 driver");
575 MODULE_AUTHOR("Kuninori Morimoto <morimoto.kuninori@renesas.com>");
576 MODULE_LICENSE("GPL");
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