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Linux/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 Zero 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 struct ak4642_drvdata {
138         const struct regmap_config *regmap_config;
139         int extended_frequencies;
140 };
141 
142 struct ak4642_priv {
143         const struct ak4642_drvdata *drvdata;
144 };
145 
146 /*
147  * Playback Volume (table 39)
148  *
149  * max : 0x00 : +12.0 dB
150  *       ( 0.5 dB step )
151  * min : 0xFE : -115.0 dB
152  * mute: 0xFF
153  */
154 static const DECLARE_TLV_DB_SCALE(out_tlv, -11550, 50, 1);
155 
156 static const struct snd_kcontrol_new ak4642_snd_controls[] = {
157 
158         SOC_DOUBLE_R_TLV("Digital Playback Volume", L_DVC, R_DVC,
159                          0, 0xFF, 1, out_tlv),
160         SOC_SINGLE("ALC Capture Switch", ALC_CTL1, 5, 1, 0),
161         SOC_SINGLE("ALC Capture ZC Switch", ALC_CTL1, 4, 1, 1),
162 };
163 
164 static const struct snd_kcontrol_new ak4642_headphone_control =
165         SOC_DAPM_SINGLE("Switch", PW_MGMT2, 6, 1, 0);
166 
167 static const struct snd_kcontrol_new ak4642_lout_mixer_controls[] = {
168         SOC_DAPM_SINGLE("DACL", SG_SL1, 4, 1, 0),
169 };
170 
171 static const struct snd_soc_dapm_widget ak4642_dapm_widgets[] = {
172 
173         /* Outputs */
174         SND_SOC_DAPM_OUTPUT("HPOUTL"),
175         SND_SOC_DAPM_OUTPUT("HPOUTR"),
176         SND_SOC_DAPM_OUTPUT("LINEOUT"),
177 
178         SND_SOC_DAPM_PGA("HPL Out", PW_MGMT2, 5, 0, NULL, 0),
179         SND_SOC_DAPM_PGA("HPR Out", PW_MGMT2, 4, 0, NULL, 0),
180         SND_SOC_DAPM_SWITCH("Headphone Enable", SND_SOC_NOPM, 0, 0,
181                             &ak4642_headphone_control),
182 
183         SND_SOC_DAPM_PGA("DACH", MD_CTL4, 0, 0, NULL, 0),
184 
185         SND_SOC_DAPM_MIXER("LINEOUT Mixer", PW_MGMT1, 3, 0,
186                            &ak4642_lout_mixer_controls[0],
187                            ARRAY_SIZE(ak4642_lout_mixer_controls)),
188 
189         /* DAC */
190         SND_SOC_DAPM_DAC("DAC", "HiFi Playback", PW_MGMT1, 2, 0),
191 };
192 
193 static const struct snd_soc_dapm_route ak4642_intercon[] = {
194 
195         /* Outputs */
196         {"HPOUTL", NULL, "HPL Out"},
197         {"HPOUTR", NULL, "HPR Out"},
198         {"LINEOUT", NULL, "LINEOUT Mixer"},
199 
200         {"HPL Out", NULL, "Headphone Enable"},
201         {"HPR Out", NULL, "Headphone Enable"},
202 
203         {"Headphone Enable", "Switch", "DACH"},
204 
205         {"DACH", NULL, "DAC"},
206 
207         {"LINEOUT Mixer", "DACL", "DAC"},
208 };
209 
210 /*
211  * ak4642 register cache
212  */
213 static const struct reg_default ak4642_reg[] = {
214         {  0, 0x00 }, {  1, 0x00 }, {  2, 0x01 }, {  3, 0x00 },
215         {  4, 0x02 }, {  5, 0x00 }, {  6, 0x00 }, {  7, 0x00 },
216         {  8, 0xe1 }, {  9, 0xe1 }, { 10, 0x18 }, { 11, 0x00 },
217         { 12, 0xe1 }, { 13, 0x18 }, { 14, 0x11 }, { 15, 0x08 },
218         { 16, 0x00 }, { 17, 0x00 }, { 18, 0x00 }, { 19, 0x00 },
219         { 20, 0x00 }, { 21, 0x00 }, { 22, 0x00 }, { 23, 0x00 },
220         { 24, 0x00 }, { 25, 0x00 }, { 26, 0x00 }, { 27, 0x00 },
221         { 28, 0x00 }, { 29, 0x00 }, { 30, 0x00 }, { 31, 0x00 },
222         { 32, 0x00 }, { 33, 0x00 }, { 34, 0x00 }, { 35, 0x00 },
223         { 36, 0x00 },
224 };
225 
226 static const struct reg_default ak4648_reg[] = {
227         {  0, 0x00 }, {  1, 0x00 }, {  2, 0x01 }, {  3, 0x00 },
228         {  4, 0x02 }, {  5, 0x00 }, {  6, 0x00 }, {  7, 0x00 },
229         {  8, 0xe1 }, {  9, 0xe1 }, { 10, 0x18 }, { 11, 0x00 },
230         { 12, 0xe1 }, { 13, 0x18 }, { 14, 0x11 }, { 15, 0xb8 },
231         { 16, 0x00 }, { 17, 0x00 }, { 18, 0x00 }, { 19, 0x00 },
232         { 20, 0x00 }, { 21, 0x00 }, { 22, 0x00 }, { 23, 0x00 },
233         { 24, 0x00 }, { 25, 0x00 }, { 26, 0x00 }, { 27, 0x00 },
234         { 28, 0x00 }, { 29, 0x00 }, { 30, 0x00 }, { 31, 0x00 },
235         { 32, 0x00 }, { 33, 0x00 }, { 34, 0x00 }, { 35, 0x00 },
236         { 36, 0x00 }, { 37, 0x88 }, { 38, 0x88 }, { 39, 0x08 },
237 };
238 
239 static int ak4642_dai_startup(struct snd_pcm_substream *substream,
240                               struct snd_soc_dai *dai)
241 {
242         int is_play = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
243         struct snd_soc_codec *codec = dai->codec;
244 
245         if (is_play) {
246                 /*
247                  * start headphone output
248                  *
249                  * PLL, Master Mode
250                  * Audio I/F Format :MSB justified (ADC & DAC)
251                  * Bass Boost Level : Middle
252                  *
253                  * This operation came from example code of
254                  * "ASAHI KASEI AK4642" (japanese) manual p97.
255                  */
256                 snd_soc_write(codec, L_IVC, 0x91); /* volume */
257                 snd_soc_write(codec, R_IVC, 0x91); /* volume */
258         } else {
259                 /*
260                  * start stereo input
261                  *
262                  * PLL Master Mode
263                  * Audio I/F Format:MSB justified (ADC & DAC)
264                  * Pre MIC AMP:+20dB
265                  * MIC Power On
266                  * ALC setting:Refer to Table 35
267                  * ALC bit=“1”
268                  *
269                  * This operation came from example code of
270                  * "ASAHI KASEI AK4642" (japanese) manual p94.
271                  */
272                 snd_soc_update_bits(codec, SG_SL1, PMMP | MGAIN0, PMMP | MGAIN0);
273                 snd_soc_write(codec, TIMER, ZTM(0x3) | WTM(0x3));
274                 snd_soc_write(codec, ALC_CTL1, ALC | LMTH0);
275                 snd_soc_update_bits(codec, PW_MGMT1, PMADL, PMADL);
276                 snd_soc_update_bits(codec, PW_MGMT3, PMADR, PMADR);
277         }
278 
279         return 0;
280 }
281 
282 static void ak4642_dai_shutdown(struct snd_pcm_substream *substream,
283                                struct snd_soc_dai *dai)
284 {
285         int is_play = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
286         struct snd_soc_codec *codec = dai->codec;
287 
288         if (is_play) {
289         } else {
290                 /* stop stereo input */
291                 snd_soc_update_bits(codec, PW_MGMT1, PMADL, 0);
292                 snd_soc_update_bits(codec, PW_MGMT3, PMADR, 0);
293                 snd_soc_update_bits(codec, ALC_CTL1, ALC, 0);
294         }
295 }
296 
297 static int ak4642_dai_set_sysclk(struct snd_soc_dai *codec_dai,
298         int clk_id, unsigned int freq, int dir)
299 {
300         struct snd_soc_codec *codec = codec_dai->codec;
301         struct ak4642_priv *priv = snd_soc_codec_get_drvdata(codec);
302         u8 pll;
303         int extended_freq = 0;
304 
305         switch (freq) {
306         case 11289600:
307                 pll = PLL2;
308                 break;
309         case 12288000:
310                 pll = PLL2 | PLL0;
311                 break;
312         case 12000000:
313                 pll = PLL2 | PLL1;
314                 break;
315         case 24000000:
316                 pll = PLL2 | PLL1 | PLL0;
317                 break;
318         case 13500000:
319                 pll = PLL3 | PLL2;
320                 break;
321         case 27000000:
322                 pll = PLL3 | PLL2 | PLL0;
323                 break;
324         case 19200000:
325                 pll = PLL3;
326                 extended_freq = 1;
327                 break;
328         case 13000000:
329                 pll = PLL3 | PLL2 | PLL1;
330                 extended_freq = 1;
331                 break;
332         case 26000000:
333                 pll = PLL3 | PLL2 | PLL1 | PLL0;
334                 extended_freq = 1;
335                 break;
336         default:
337                 return -EINVAL;
338         }
339 
340         if (extended_freq && !priv->drvdata->extended_frequencies)
341                 return -EINVAL;
342 
343         snd_soc_update_bits(codec, MD_CTL1, PLL_MASK, pll);
344 
345         return 0;
346 }
347 
348 static int ak4642_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
349 {
350         struct snd_soc_codec *codec = dai->codec;
351         u8 data;
352         u8 bcko;
353 
354         data = MCKO | PMPLL; /* use MCKO */
355         bcko = 0;
356 
357         /* set master/slave audio interface */
358         switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
359         case SND_SOC_DAIFMT_CBM_CFM:
360                 data |= MS;
361                 bcko = BCKO_64;
362                 break;
363         case SND_SOC_DAIFMT_CBS_CFS:
364                 break;
365         default:
366                 return -EINVAL;
367         }
368         snd_soc_update_bits(codec, PW_MGMT2, MS | MCKO | PMPLL, data);
369         snd_soc_update_bits(codec, MD_CTL1, BCKO_MASK, bcko);
370 
371         /* format type */
372         data = 0;
373         switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
374         case SND_SOC_DAIFMT_LEFT_J:
375                 data = LEFT_J;
376                 break;
377         case SND_SOC_DAIFMT_I2S:
378                 data = I2S;
379                 break;
380         /* FIXME
381          * Please add RIGHT_J / DSP support here
382          */
383         default:
384                 return -EINVAL;
385         }
386         snd_soc_update_bits(codec, MD_CTL1, DIF_MASK, data);
387 
388         return 0;
389 }
390 
391 static int ak4642_dai_hw_params(struct snd_pcm_substream *substream,
392                                 struct snd_pcm_hw_params *params,
393                                 struct snd_soc_dai *dai)
394 {
395         struct snd_soc_codec *codec = dai->codec;
396         u8 rate;
397 
398         switch (params_rate(params)) {
399         case 7350:
400                 rate = FS2;
401                 break;
402         case 8000:
403                 rate = 0;
404                 break;
405         case 11025:
406                 rate = FS2 | FS0;
407                 break;
408         case 12000:
409                 rate = FS0;
410                 break;
411         case 14700:
412                 rate = FS2 | FS1;
413                 break;
414         case 16000:
415                 rate = FS1;
416                 break;
417         case 22050:
418                 rate = FS2 | FS1 | FS0;
419                 break;
420         case 24000:
421                 rate = FS1 | FS0;
422                 break;
423         case 29400:
424                 rate = FS3 | FS2 | FS1;
425                 break;
426         case 32000:
427                 rate = FS3 | FS1;
428                 break;
429         case 44100:
430                 rate = FS3 | FS2 | FS1 | FS0;
431                 break;
432         case 48000:
433                 rate = FS3 | FS1 | FS0;
434                 break;
435         default:
436                 return -EINVAL;
437         }
438         snd_soc_update_bits(codec, MD_CTL2, FS_MASK, rate);
439 
440         return 0;
441 }
442 
443 static int ak4642_set_bias_level(struct snd_soc_codec *codec,
444                                  enum snd_soc_bias_level level)
445 {
446         switch (level) {
447         case SND_SOC_BIAS_OFF:
448                 snd_soc_write(codec, PW_MGMT1, 0x00);
449                 break;
450         default:
451                 snd_soc_update_bits(codec, PW_MGMT1, PMVCM, PMVCM);
452                 break;
453         }
454         codec->dapm.bias_level = level;
455 
456         return 0;
457 }
458 
459 static const struct snd_soc_dai_ops ak4642_dai_ops = {
460         .startup        = ak4642_dai_startup,
461         .shutdown       = ak4642_dai_shutdown,
462         .set_sysclk     = ak4642_dai_set_sysclk,
463         .set_fmt        = ak4642_dai_set_fmt,
464         .hw_params      = ak4642_dai_hw_params,
465 };
466 
467 static struct snd_soc_dai_driver ak4642_dai = {
468         .name = "ak4642-hifi",
469         .playback = {
470                 .stream_name = "Playback",
471                 .channels_min = 1,
472                 .channels_max = 2,
473                 .rates = SNDRV_PCM_RATE_8000_48000,
474                 .formats = SNDRV_PCM_FMTBIT_S16_LE },
475         .capture = {
476                 .stream_name = "Capture",
477                 .channels_min = 1,
478                 .channels_max = 2,
479                 .rates = SNDRV_PCM_RATE_8000_48000,
480                 .formats = SNDRV_PCM_FMTBIT_S16_LE },
481         .ops = &ak4642_dai_ops,
482         .symmetric_rates = 1,
483 };
484 
485 static int ak4642_resume(struct snd_soc_codec *codec)
486 {
487         struct regmap *regmap = dev_get_regmap(codec->dev, NULL);
488 
489         regcache_mark_dirty(regmap);
490         regcache_sync(regmap);
491         return 0;
492 }
493 
494 
495 static int ak4642_probe(struct snd_soc_codec *codec)
496 {
497         ak4642_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
498 
499         return 0;
500 }
501 
502 static int ak4642_remove(struct snd_soc_codec *codec)
503 {
504         ak4642_set_bias_level(codec, SND_SOC_BIAS_OFF);
505         return 0;
506 }
507 
508 static struct snd_soc_codec_driver soc_codec_dev_ak4642 = {
509         .probe                  = ak4642_probe,
510         .remove                 = ak4642_remove,
511         .resume                 = ak4642_resume,
512         .set_bias_level         = ak4642_set_bias_level,
513         .controls               = ak4642_snd_controls,
514         .num_controls           = ARRAY_SIZE(ak4642_snd_controls),
515         .dapm_widgets           = ak4642_dapm_widgets,
516         .num_dapm_widgets       = ARRAY_SIZE(ak4642_dapm_widgets),
517         .dapm_routes            = ak4642_intercon,
518         .num_dapm_routes        = ARRAY_SIZE(ak4642_intercon),
519 };
520 
521 static const struct regmap_config ak4642_regmap = {
522         .reg_bits               = 8,
523         .val_bits               = 8,
524         .max_register           = ARRAY_SIZE(ak4642_reg) + 1,
525         .reg_defaults           = ak4642_reg,
526         .num_reg_defaults       = ARRAY_SIZE(ak4642_reg),
527 };
528 
529 static const struct regmap_config ak4648_regmap = {
530         .reg_bits               = 8,
531         .val_bits               = 8,
532         .max_register           = ARRAY_SIZE(ak4648_reg) + 1,
533         .reg_defaults           = ak4648_reg,
534         .num_reg_defaults       = ARRAY_SIZE(ak4648_reg),
535 };
536 
537 static const struct ak4642_drvdata ak4642_drvdata = {
538         .regmap_config = &ak4642_regmap,
539 };
540 
541 static const struct ak4642_drvdata ak4643_drvdata = {
542         .regmap_config = &ak4642_regmap,
543 };
544 
545 static const struct ak4642_drvdata ak4648_drvdata = {
546         .regmap_config = &ak4648_regmap,
547         .extended_frequencies = 1,
548 };
549 
550 static const struct of_device_id ak4642_of_match[];
551 static int ak4642_i2c_probe(struct i2c_client *i2c,
552                             const struct i2c_device_id *id)
553 {
554         struct device_node *np = i2c->dev.of_node;
555         const struct ak4642_drvdata *drvdata = NULL;
556         struct regmap *regmap;
557         struct ak4642_priv *priv;
558 
559         if (np) {
560                 const struct of_device_id *of_id;
561 
562                 of_id = of_match_device(ak4642_of_match, &i2c->dev);
563                 if (of_id)
564                         drvdata = of_id->data;
565         } else {
566                 drvdata = (const struct ak4642_drvdata *)id->driver_data;
567         }
568 
569         if (!drvdata) {
570                 dev_err(&i2c->dev, "Unknown device type\n");
571                 return -EINVAL;
572         }
573 
574         priv = devm_kzalloc(&i2c->dev, sizeof(*priv), GFP_KERNEL);
575         if (!priv)
576                 return -ENOMEM;
577 
578         priv->drvdata = drvdata;
579 
580         i2c_set_clientdata(i2c, priv);
581 
582         regmap = devm_regmap_init_i2c(i2c, drvdata->regmap_config);
583         if (IS_ERR(regmap))
584                 return PTR_ERR(regmap);
585 
586         return snd_soc_register_codec(&i2c->dev,
587                                       &soc_codec_dev_ak4642, &ak4642_dai, 1);
588 }
589 
590 static int ak4642_i2c_remove(struct i2c_client *client)
591 {
592         snd_soc_unregister_codec(&client->dev);
593         return 0;
594 }
595 
596 static const struct of_device_id ak4642_of_match[] = {
597         { .compatible = "asahi-kasei,ak4642",   .data = &ak4642_drvdata},
598         { .compatible = "asahi-kasei,ak4643",   .data = &ak4643_drvdata},
599         { .compatible = "asahi-kasei,ak4648",   .data = &ak4648_drvdata},
600         {},
601 };
602 MODULE_DEVICE_TABLE(of, ak4642_of_match);
603 
604 static const struct i2c_device_id ak4642_i2c_id[] = {
605         { "ak4642", (kernel_ulong_t)&ak4642_drvdata },
606         { "ak4643", (kernel_ulong_t)&ak4643_drvdata },
607         { "ak4648", (kernel_ulong_t)&ak4648_drvdata },
608         { }
609 };
610 MODULE_DEVICE_TABLE(i2c, ak4642_i2c_id);
611 
612 static struct i2c_driver ak4642_i2c_driver = {
613         .driver = {
614                 .name = "ak4642-codec",
615                 .owner = THIS_MODULE,
616                 .of_match_table = ak4642_of_match,
617         },
618         .probe          = ak4642_i2c_probe,
619         .remove         = ak4642_i2c_remove,
620         .id_table       = ak4642_i2c_id,
621 };
622 
623 module_i2c_driver(ak4642_i2c_driver);
624 
625 MODULE_DESCRIPTION("Soc AK4642 driver");
626 MODULE_AUTHOR("Kuninori Morimoto <morimoto.kuninori@renesas.com>");
627 MODULE_LICENSE("GPL");
628 

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