Loading...
1// SPDX-License-Identifier: GPL-2.0
2//
3// ak4642.c -- AK4642/AK4643 ALSA Soc Audio driver
4//
5// Copyright (C) 2009 Renesas Solutions Corp.
6// Kuninori Morimoto <morimoto.kuninori@renesas.com>
7//
8// Based on wm8731.c by Richard Purdie
9// Based on ak4535.c by Richard Purdie
10// Based on wm8753.c by Liam Girdwood
11
12/* ** CAUTION **
13 *
14 * This is very simple driver.
15 * It can use headphone output / stereo input only
16 *
17 * AK4642 is tested.
18 * AK4643 is tested.
19 * AK4648 is tested.
20 */
21
22#include <linux/clk.h>
23#include <linux/clk-provider.h>
24#include <linux/delay.h>
25#include <linux/i2c.h>
26#include <linux/slab.h>
27#include <linux/of.h>
28#include <linux/module.h>
29#include <linux/regmap.h>
30#include <sound/soc.h>
31#include <sound/initval.h>
32#include <sound/tlv.h>
33
34#define PW_MGMT1 0x00
35#define PW_MGMT2 0x01
36#define SG_SL1 0x02
37#define SG_SL2 0x03
38#define MD_CTL1 0x04
39#define MD_CTL2 0x05
40#define TIMER 0x06
41#define ALC_CTL1 0x07
42#define ALC_CTL2 0x08
43#define L_IVC 0x09
44#define L_DVC 0x0a
45#define ALC_CTL3 0x0b
46#define R_IVC 0x0c
47#define R_DVC 0x0d
48#define MD_CTL3 0x0e
49#define MD_CTL4 0x0f
50#define PW_MGMT3 0x10
51#define DF_S 0x11
52#define FIL3_0 0x12
53#define FIL3_1 0x13
54#define FIL3_2 0x14
55#define FIL3_3 0x15
56#define EQ_0 0x16
57#define EQ_1 0x17
58#define EQ_2 0x18
59#define EQ_3 0x19
60#define EQ_4 0x1a
61#define EQ_5 0x1b
62#define FIL1_0 0x1c
63#define FIL1_1 0x1d
64#define FIL1_2 0x1e
65#define FIL1_3 0x1f /* The maximum valid register for ak4642 */
66#define PW_MGMT4 0x20
67#define MD_CTL5 0x21
68#define LO_MS 0x22
69#define HP_MS 0x23
70#define SPK_MS 0x24 /* The maximum valid register for ak4643 */
71#define EQ_FBEQAB 0x25
72#define EQ_FBEQCD 0x26
73#define EQ_FBEQE 0x27 /* The maximum valid register for ak4648 */
74
75/* PW_MGMT1*/
76#define PMVCM (1 << 6) /* VCOM Power Management */
77#define PMMIN (1 << 5) /* MIN Input Power Management */
78#define PMDAC (1 << 2) /* DAC Power Management */
79#define PMADL (1 << 0) /* MIC Amp Lch and ADC Lch Power Management */
80
81/* PW_MGMT2 */
82#define HPMTN (1 << 6)
83#define PMHPL (1 << 5)
84#define PMHPR (1 << 4)
85#define MS (1 << 3) /* master/slave select */
86#define MCKO (1 << 1)
87#define PMPLL (1 << 0)
88
89#define PMHP_MASK (PMHPL | PMHPR)
90#define PMHP PMHP_MASK
91
92/* PW_MGMT3 */
93#define PMADR (1 << 0) /* MIC L / ADC R Power Management */
94
95/* SG_SL1 */
96#define MINS (1 << 6) /* Switch from MIN to Speaker */
97#define DACL (1 << 4) /* Switch from DAC to Stereo or Receiver */
98#define PMMP (1 << 2) /* MPWR pin Power Management */
99#define MGAIN0 (1 << 0) /* MIC amp gain*/
100
101/* SG_SL2 */
102#define LOPS (1 << 6) /* Stero Line-out Power Save Mode */
103
104/* TIMER */
105#define ZTM(param) ((param & 0x3) << 4) /* ALC Zero Crossing TimeOut */
106#define WTM(param) (((param & 0x4) << 4) | ((param & 0x3) << 2))
107
108/* ALC_CTL1 */
109#define ALC (1 << 5) /* ALC Enable */
110#define LMTH0 (1 << 0) /* ALC Limiter / Recovery Level */
111
112/* MD_CTL1 */
113#define PLL3 (1 << 7)
114#define PLL2 (1 << 6)
115#define PLL1 (1 << 5)
116#define PLL0 (1 << 4)
117#define PLL_MASK (PLL3 | PLL2 | PLL1 | PLL0)
118
119#define BCKO_MASK (1 << 3)
120#define BCKO_64 BCKO_MASK
121
122#define DIF_MASK (3 << 0)
123#define DSP (0 << 0)
124#define RIGHT_J (1 << 0)
125#define LEFT_J (2 << 0)
126#define I2S (3 << 0)
127
128/* MD_CTL2 */
129#define FSs(val) (((val & 0x7) << 0) | ((val & 0x8) << 2))
130#define PSs(val) ((val & 0x3) << 6)
131
132/* MD_CTL3 */
133#define BST1 (1 << 3)
134
135/* MD_CTL4 */
136#define DACH (1 << 0)
137
138struct ak4642_drvdata {
139 const struct regmap_config *regmap_config;
140 int extended_frequencies;
141};
142
143struct ak4642_priv {
144 const struct ak4642_drvdata *drvdata;
145 struct clk *mcko;
146};
147
148/*
149 * Playback Volume (table 39)
150 *
151 * max : 0x00 : +12.0 dB
152 * ( 0.5 dB step )
153 * min : 0xFE : -115.0 dB
154 * mute: 0xFF
155 */
156static const DECLARE_TLV_DB_SCALE(out_tlv, -11550, 50, 1);
157
158static const struct snd_kcontrol_new ak4642_snd_controls[] = {
159
160 SOC_DOUBLE_R_TLV("Digital Playback Volume", L_DVC, R_DVC,
161 0, 0xFF, 1, out_tlv),
162 SOC_SINGLE("ALC Capture Switch", ALC_CTL1, 5, 1, 0),
163 SOC_SINGLE("ALC Capture ZC Switch", ALC_CTL1, 4, 1, 1),
164};
165
166static const struct snd_kcontrol_new ak4642_headphone_control =
167 SOC_DAPM_SINGLE("Switch", PW_MGMT2, 6, 1, 0);
168
169static const struct snd_kcontrol_new ak4642_lout_mixer_controls[] = {
170 SOC_DAPM_SINGLE("DACL", SG_SL1, 4, 1, 0),
171};
172
173/* event handlers */
174static int ak4642_lout_event(struct snd_soc_dapm_widget *w,
175 struct snd_kcontrol *kcontrol, int event)
176{
177 struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm);
178
179 switch (event) {
180 case SND_SOC_DAPM_PRE_PMD:
181 case SND_SOC_DAPM_PRE_PMU:
182 /* Power save mode ON */
183 snd_soc_component_update_bits(component, SG_SL2, LOPS, LOPS);
184 break;
185 case SND_SOC_DAPM_POST_PMU:
186 case SND_SOC_DAPM_POST_PMD:
187 /* Power save mode OFF */
188 msleep(300);
189 snd_soc_component_update_bits(component, SG_SL2, LOPS, 0);
190 break;
191 }
192
193 return 0;
194}
195
196static const struct snd_soc_dapm_widget ak4642_dapm_widgets[] = {
197
198 /* Outputs */
199 SND_SOC_DAPM_OUTPUT("HPOUTL"),
200 SND_SOC_DAPM_OUTPUT("HPOUTR"),
201 SND_SOC_DAPM_OUTPUT("LINEOUT"),
202
203 SND_SOC_DAPM_PGA("HPL Out", PW_MGMT2, 5, 0, NULL, 0),
204 SND_SOC_DAPM_PGA("HPR Out", PW_MGMT2, 4, 0, NULL, 0),
205 SND_SOC_DAPM_SWITCH("Headphone Enable", SND_SOC_NOPM, 0, 0,
206 &ak4642_headphone_control),
207
208 SND_SOC_DAPM_PGA("DACH", MD_CTL4, 0, 0, NULL, 0),
209
210 SND_SOC_DAPM_MIXER_E("LINEOUT Mixer", PW_MGMT1, 3, 0,
211 &ak4642_lout_mixer_controls[0],
212 ARRAY_SIZE(ak4642_lout_mixer_controls),
213 ak4642_lout_event,
214 SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU |
215 SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD),
216
217 /* DAC */
218 SND_SOC_DAPM_DAC("DAC", NULL, PW_MGMT1, 2, 0),
219};
220
221static const struct snd_soc_dapm_route ak4642_intercon[] = {
222
223 /* Outputs */
224 {"HPOUTL", NULL, "HPL Out"},
225 {"HPOUTR", NULL, "HPR Out"},
226 {"LINEOUT", NULL, "LINEOUT Mixer"},
227
228 {"HPL Out", NULL, "Headphone Enable"},
229 {"HPR Out", NULL, "Headphone Enable"},
230
231 {"Headphone Enable", "Switch", "DACH"},
232
233 {"DACH", NULL, "DAC"},
234
235 {"LINEOUT Mixer", "DACL", "DAC"},
236
237 { "DAC", NULL, "Playback" },
238};
239
240/*
241 * ak4642 register cache
242 */
243static const struct reg_default ak4643_reg[] = {
244 { 0, 0x00 }, { 1, 0x00 }, { 2, 0x01 }, { 3, 0x00 },
245 { 4, 0x02 }, { 5, 0x00 }, { 6, 0x00 }, { 7, 0x00 },
246 { 8, 0xe1 }, { 9, 0xe1 }, { 10, 0x18 }, { 11, 0x00 },
247 { 12, 0xe1 }, { 13, 0x18 }, { 14, 0x11 }, { 15, 0x08 },
248 { 16, 0x00 }, { 17, 0x00 }, { 18, 0x00 }, { 19, 0x00 },
249 { 20, 0x00 }, { 21, 0x00 }, { 22, 0x00 }, { 23, 0x00 },
250 { 24, 0x00 }, { 25, 0x00 }, { 26, 0x00 }, { 27, 0x00 },
251 { 28, 0x00 }, { 29, 0x00 }, { 30, 0x00 }, { 31, 0x00 },
252 { 32, 0x00 }, { 33, 0x00 }, { 34, 0x00 }, { 35, 0x00 },
253 { 36, 0x00 },
254};
255
256/* The default settings for 0x0 ~ 0x1f registers are the same for ak4642
257 and ak4643. So we reuse the ak4643 reg_default for ak4642.
258 The valid registers for ak4642 are 0x0 ~ 0x1f which is a subset of ak4643,
259 so define NUM_AK4642_REG_DEFAULTS for ak4642.
260*/
261#define ak4642_reg ak4643_reg
262#define NUM_AK4642_REG_DEFAULTS (FIL1_3 + 1)
263
264static const struct reg_default ak4648_reg[] = {
265 { 0, 0x00 }, { 1, 0x00 }, { 2, 0x01 }, { 3, 0x00 },
266 { 4, 0x02 }, { 5, 0x00 }, { 6, 0x00 }, { 7, 0x00 },
267 { 8, 0xe1 }, { 9, 0xe1 }, { 10, 0x18 }, { 11, 0x00 },
268 { 12, 0xe1 }, { 13, 0x18 }, { 14, 0x11 }, { 15, 0xb8 },
269 { 16, 0x00 }, { 17, 0x00 }, { 18, 0x00 }, { 19, 0x00 },
270 { 20, 0x00 }, { 21, 0x00 }, { 22, 0x00 }, { 23, 0x00 },
271 { 24, 0x00 }, { 25, 0x00 }, { 26, 0x00 }, { 27, 0x00 },
272 { 28, 0x00 }, { 29, 0x00 }, { 30, 0x00 }, { 31, 0x00 },
273 { 32, 0x00 }, { 33, 0x00 }, { 34, 0x00 }, { 35, 0x00 },
274 { 36, 0x00 }, { 37, 0x88 }, { 38, 0x88 }, { 39, 0x08 },
275};
276
277static int ak4642_dai_startup(struct snd_pcm_substream *substream,
278 struct snd_soc_dai *dai)
279{
280 int is_play = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
281 struct snd_soc_component *component = dai->component;
282
283 if (is_play) {
284 /*
285 * start headphone output
286 *
287 * PLL, Master Mode
288 * Audio I/F Format :MSB justified (ADC & DAC)
289 * Bass Boost Level : Middle
290 *
291 * This operation came from example code of
292 * "ASAHI KASEI AK4642" (japanese) manual p97.
293 */
294 snd_soc_component_write(component, L_IVC, 0x91); /* volume */
295 snd_soc_component_write(component, R_IVC, 0x91); /* volume */
296 } else {
297 /*
298 * start stereo input
299 *
300 * PLL Master Mode
301 * Audio I/F Format:MSB justified (ADC & DAC)
302 * Pre MIC AMP:+20dB
303 * MIC Power On
304 * ALC setting:Refer to Table 35
305 * ALC bit=“1”
306 *
307 * This operation came from example code of
308 * "ASAHI KASEI AK4642" (japanese) manual p94.
309 */
310 snd_soc_component_update_bits(component, SG_SL1, PMMP | MGAIN0, PMMP | MGAIN0);
311 snd_soc_component_write(component, TIMER, ZTM(0x3) | WTM(0x3));
312 snd_soc_component_write(component, ALC_CTL1, ALC | LMTH0);
313 snd_soc_component_update_bits(component, PW_MGMT1, PMADL, PMADL);
314 snd_soc_component_update_bits(component, PW_MGMT3, PMADR, PMADR);
315 }
316
317 return 0;
318}
319
320static void ak4642_dai_shutdown(struct snd_pcm_substream *substream,
321 struct snd_soc_dai *dai)
322{
323 int is_play = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
324 struct snd_soc_component *component = dai->component;
325
326 if (is_play) {
327 } else {
328 /* stop stereo input */
329 snd_soc_component_update_bits(component, PW_MGMT1, PMADL, 0);
330 snd_soc_component_update_bits(component, PW_MGMT3, PMADR, 0);
331 snd_soc_component_update_bits(component, ALC_CTL1, ALC, 0);
332 }
333}
334
335static int ak4642_dai_set_sysclk(struct snd_soc_dai *codec_dai,
336 int clk_id, unsigned int freq, int dir)
337{
338 struct snd_soc_component *component = codec_dai->component;
339 struct ak4642_priv *priv = snd_soc_component_get_drvdata(component);
340 u8 pll;
341 int extended_freq = 0;
342
343 switch (freq) {
344 case 11289600:
345 pll = PLL2;
346 break;
347 case 12288000:
348 pll = PLL2 | PLL0;
349 break;
350 case 12000000:
351 pll = PLL2 | PLL1;
352 break;
353 case 24000000:
354 pll = PLL2 | PLL1 | PLL0;
355 break;
356 case 13500000:
357 pll = PLL3 | PLL2;
358 break;
359 case 27000000:
360 pll = PLL3 | PLL2 | PLL0;
361 break;
362 case 19200000:
363 pll = PLL3;
364 extended_freq = 1;
365 break;
366 case 13000000:
367 pll = PLL3 | PLL2 | PLL1;
368 extended_freq = 1;
369 break;
370 case 26000000:
371 pll = PLL3 | PLL2 | PLL1 | PLL0;
372 extended_freq = 1;
373 break;
374 default:
375 return -EINVAL;
376 }
377
378 if (extended_freq && !priv->drvdata->extended_frequencies)
379 return -EINVAL;
380
381 snd_soc_component_update_bits(component, MD_CTL1, PLL_MASK, pll);
382
383 return 0;
384}
385
386static int ak4642_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
387{
388 struct snd_soc_component *component = dai->component;
389 u8 data;
390 u8 bcko;
391
392 data = MCKO | PMPLL; /* use MCKO */
393 bcko = 0;
394
395 /* set clocking for audio interface */
396 switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) {
397 case SND_SOC_DAIFMT_CBP_CFP:
398 data |= MS;
399 bcko = BCKO_64;
400 break;
401 case SND_SOC_DAIFMT_CBC_CFC:
402 break;
403 default:
404 return -EINVAL;
405 }
406 snd_soc_component_update_bits(component, PW_MGMT2, MS | MCKO | PMPLL, data);
407 snd_soc_component_update_bits(component, MD_CTL1, BCKO_MASK, bcko);
408
409 /* format type */
410 data = 0;
411 switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
412 case SND_SOC_DAIFMT_LEFT_J:
413 data = LEFT_J;
414 break;
415 case SND_SOC_DAIFMT_I2S:
416 data = I2S;
417 break;
418 /* FIXME
419 * Please add RIGHT_J / DSP support here
420 */
421 default:
422 return -EINVAL;
423 }
424 snd_soc_component_update_bits(component, MD_CTL1, DIF_MASK, data);
425
426 return 0;
427}
428
429static int ak4642_set_mcko(struct snd_soc_component *component,
430 u32 frequency)
431{
432 static const u32 fs_list[] = {
433 [0] = 8000,
434 [1] = 12000,
435 [2] = 16000,
436 [3] = 24000,
437 [4] = 7350,
438 [5] = 11025,
439 [6] = 14700,
440 [7] = 22050,
441 [10] = 32000,
442 [11] = 48000,
443 [14] = 29400,
444 [15] = 44100,
445 };
446 static const u32 ps_list[] = {
447 [0] = 256,
448 [1] = 128,
449 [2] = 64,
450 [3] = 32
451 };
452 int ps, fs;
453
454 for (ps = 0; ps < ARRAY_SIZE(ps_list); ps++) {
455 for (fs = 0; fs < ARRAY_SIZE(fs_list); fs++) {
456 if (frequency == ps_list[ps] * fs_list[fs]) {
457 snd_soc_component_write(component, MD_CTL2,
458 PSs(ps) | FSs(fs));
459 return 0;
460 }
461 }
462 }
463
464 return 0;
465}
466
467static int ak4642_dai_hw_params(struct snd_pcm_substream *substream,
468 struct snd_pcm_hw_params *params,
469 struct snd_soc_dai *dai)
470{
471 struct snd_soc_component *component = dai->component;
472 struct ak4642_priv *priv = snd_soc_component_get_drvdata(component);
473 u32 rate = clk_get_rate(priv->mcko);
474
475 if (!rate)
476 rate = params_rate(params) * 256;
477
478 return ak4642_set_mcko(component, rate);
479}
480
481static int ak4642_set_bias_level(struct snd_soc_component *component,
482 enum snd_soc_bias_level level)
483{
484 switch (level) {
485 case SND_SOC_BIAS_OFF:
486 snd_soc_component_write(component, PW_MGMT1, 0x00);
487 break;
488 default:
489 snd_soc_component_update_bits(component, PW_MGMT1, PMVCM, PMVCM);
490 break;
491 }
492
493 return 0;
494}
495
496static const struct snd_soc_dai_ops ak4642_dai_ops = {
497 .startup = ak4642_dai_startup,
498 .shutdown = ak4642_dai_shutdown,
499 .set_sysclk = ak4642_dai_set_sysclk,
500 .set_fmt = ak4642_dai_set_fmt,
501 .hw_params = ak4642_dai_hw_params,
502};
503
504static struct snd_soc_dai_driver ak4642_dai = {
505 .name = "ak4642-hifi",
506 .playback = {
507 .stream_name = "Playback",
508 .channels_min = 2,
509 .channels_max = 2,
510 .rates = SNDRV_PCM_RATE_8000_48000,
511 .formats = SNDRV_PCM_FMTBIT_S16_LE },
512 .capture = {
513 .stream_name = "Capture",
514 .channels_min = 2,
515 .channels_max = 2,
516 .rates = SNDRV_PCM_RATE_8000_48000,
517 .formats = SNDRV_PCM_FMTBIT_S16_LE },
518 .ops = &ak4642_dai_ops,
519 .symmetric_rate = 1,
520};
521
522static int ak4642_suspend(struct snd_soc_component *component)
523{
524 struct regmap *regmap = dev_get_regmap(component->dev, NULL);
525
526 regcache_cache_only(regmap, true);
527 regcache_mark_dirty(regmap);
528 return 0;
529}
530
531static int ak4642_resume(struct snd_soc_component *component)
532{
533 struct regmap *regmap = dev_get_regmap(component->dev, NULL);
534
535 regcache_cache_only(regmap, false);
536 regcache_sync(regmap);
537 return 0;
538}
539static int ak4642_probe(struct snd_soc_component *component)
540{
541 struct ak4642_priv *priv = snd_soc_component_get_drvdata(component);
542
543 if (priv->mcko)
544 ak4642_set_mcko(component, clk_get_rate(priv->mcko));
545
546 return 0;
547}
548
549static const struct snd_soc_component_driver soc_component_dev_ak4642 = {
550 .probe = ak4642_probe,
551 .suspend = ak4642_suspend,
552 .resume = ak4642_resume,
553 .set_bias_level = ak4642_set_bias_level,
554 .controls = ak4642_snd_controls,
555 .num_controls = ARRAY_SIZE(ak4642_snd_controls),
556 .dapm_widgets = ak4642_dapm_widgets,
557 .num_dapm_widgets = ARRAY_SIZE(ak4642_dapm_widgets),
558 .dapm_routes = ak4642_intercon,
559 .num_dapm_routes = ARRAY_SIZE(ak4642_intercon),
560 .idle_bias_on = 1,
561 .endianness = 1,
562};
563
564static const struct regmap_config ak4642_regmap = {
565 .reg_bits = 8,
566 .val_bits = 8,
567 .max_register = FIL1_3,
568 .reg_defaults = ak4642_reg,
569 .num_reg_defaults = NUM_AK4642_REG_DEFAULTS,
570 .cache_type = REGCACHE_RBTREE,
571};
572
573static const struct regmap_config ak4643_regmap = {
574 .reg_bits = 8,
575 .val_bits = 8,
576 .max_register = SPK_MS,
577 .reg_defaults = ak4643_reg,
578 .num_reg_defaults = ARRAY_SIZE(ak4643_reg),
579 .cache_type = REGCACHE_RBTREE,
580};
581
582static const struct regmap_config ak4648_regmap = {
583 .reg_bits = 8,
584 .val_bits = 8,
585 .max_register = EQ_FBEQE,
586 .reg_defaults = ak4648_reg,
587 .num_reg_defaults = ARRAY_SIZE(ak4648_reg),
588 .cache_type = REGCACHE_RBTREE,
589};
590
591static const struct ak4642_drvdata ak4642_drvdata = {
592 .regmap_config = &ak4642_regmap,
593};
594
595static const struct ak4642_drvdata ak4643_drvdata = {
596 .regmap_config = &ak4643_regmap,
597};
598
599static const struct ak4642_drvdata ak4648_drvdata = {
600 .regmap_config = &ak4648_regmap,
601 .extended_frequencies = 1,
602};
603
604#ifdef CONFIG_COMMON_CLK
605static struct clk *ak4642_of_parse_mcko(struct device *dev)
606{
607 struct device_node *np = dev->of_node;
608 struct clk *clk;
609 const char *clk_name = np->name;
610 const char *parent_clk_name = NULL;
611 u32 rate;
612
613 if (of_property_read_u32(np, "clock-frequency", &rate))
614 return NULL;
615
616 if (of_property_read_bool(np, "clocks"))
617 parent_clk_name = of_clk_get_parent_name(np, 0);
618
619 of_property_read_string(np, "clock-output-names", &clk_name);
620
621 clk = clk_register_fixed_rate(dev, clk_name, parent_clk_name, 0, rate);
622 if (!IS_ERR(clk))
623 of_clk_add_provider(np, of_clk_src_simple_get, clk);
624
625 return clk;
626}
627#else
628#define ak4642_of_parse_mcko(d) 0
629#endif
630
631static int ak4642_i2c_probe(struct i2c_client *i2c)
632{
633 struct device *dev = &i2c->dev;
634 const struct ak4642_drvdata *drvdata;
635 struct regmap *regmap;
636 struct ak4642_priv *priv;
637 struct clk *mcko = NULL;
638
639 if (dev_fwnode(dev)) {
640 mcko = ak4642_of_parse_mcko(dev);
641 if (IS_ERR(mcko))
642 mcko = NULL;
643 }
644
645 drvdata = i2c_get_match_data(i2c);
646 if (!drvdata)
647 return dev_err_probe(dev, -EINVAL, "Unknown device type\n");
648
649 priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL);
650 if (!priv)
651 return -ENOMEM;
652
653 priv->drvdata = drvdata;
654 priv->mcko = mcko;
655
656 i2c_set_clientdata(i2c, priv);
657
658 regmap = devm_regmap_init_i2c(i2c, drvdata->regmap_config);
659 if (IS_ERR(regmap))
660 return PTR_ERR(regmap);
661
662 return devm_snd_soc_register_component(dev,
663 &soc_component_dev_ak4642, &ak4642_dai, 1);
664}
665
666static const struct of_device_id ak4642_of_match[] = {
667 { .compatible = "asahi-kasei,ak4642", .data = &ak4642_drvdata},
668 { .compatible = "asahi-kasei,ak4643", .data = &ak4643_drvdata},
669 { .compatible = "asahi-kasei,ak4648", .data = &ak4648_drvdata},
670 {}
671};
672MODULE_DEVICE_TABLE(of, ak4642_of_match);
673
674static const struct i2c_device_id ak4642_i2c_id[] = {
675 { "ak4642", (kernel_ulong_t)&ak4642_drvdata },
676 { "ak4643", (kernel_ulong_t)&ak4643_drvdata },
677 { "ak4648", (kernel_ulong_t)&ak4648_drvdata },
678 {}
679};
680MODULE_DEVICE_TABLE(i2c, ak4642_i2c_id);
681
682static struct i2c_driver ak4642_i2c_driver = {
683 .driver = {
684 .name = "ak4642-codec",
685 .of_match_table = ak4642_of_match,
686 },
687 .probe = ak4642_i2c_probe,
688 .id_table = ak4642_i2c_id,
689};
690
691module_i2c_driver(ak4642_i2c_driver);
692
693MODULE_DESCRIPTION("Soc AK4642 driver");
694MODULE_AUTHOR("Kuninori Morimoto <morimoto.kuninori@renesas.com>");
695MODULE_LICENSE("GPL v2");
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 */
145static const DECLARE_TLV_DB_SCALE(out_tlv, -11550, 50, 1);
146
147static 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
153static const struct snd_kcontrol_new ak4642_headphone_control =
154 SOC_DAPM_SINGLE("Switch", PW_MGMT2, 6, 1, 0);
155
156static const struct snd_kcontrol_new ak4642_lout_mixer_controls[] = {
157 SOC_DAPM_SINGLE("DACL", SG_SL1, 4, 1, 0),
158};
159
160static 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
182static 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 */
202static 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
215static 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
228static 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
271static 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
286static 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
319static 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
362static 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
414static 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
430static 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
438static 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
456static 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
466static 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
473static int ak4642_remove(struct snd_soc_codec *codec)
474{
475 ak4642_set_bias_level(codec, SND_SOC_BIAS_OFF);
476 return 0;
477}
478
479static 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
492static 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
500static 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
508static struct of_device_id ak4642_of_match[];
509static 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
539static int ak4642_i2c_remove(struct i2c_client *client)
540{
541 snd_soc_unregister_codec(&client->dev);
542 return 0;
543}
544
545static 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};
551MODULE_DEVICE_TABLE(of, ak4642_of_match);
552
553static 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};
559MODULE_DEVICE_TABLE(i2c, ak4642_i2c_id);
560
561static 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
572module_i2c_driver(ak4642_i2c_driver);
573
574MODULE_DESCRIPTION("Soc AK4642 driver");
575MODULE_AUTHOR("Kuninori Morimoto <morimoto.kuninori@renesas.com>");
576MODULE_LICENSE("GPL");