2 * ak4642.c -- AK4642/AK4643 ALSA Soc Audio driver
4 * Copyright (C) 2009 Renesas Solutions Corp.
5 * Kuninori Morimoto <morimoto.kuninori@renesas.com>
7 * Based on wm8731.c by Richard Purdie
8 * Based on ak4535.c by Richard Purdie
9 * Based on wm8753.c by Liam Girdwood
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.
18 * This is very simple driver.
19 * It can use headphone output / stereo input only
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>
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 */
81 #define HPMTN (1 << 6)
82 #define PMHPL (1 << 5)
83 #define PMHPR (1 << 4)
84 #define MS (1 << 3) /* master/slave select */
86 #define PMPLL (1 << 0)
88 #define PMHP_MASK (PMHPL | PMHPR)
89 #define PMHP PMHP_MASK
92 #define PMADR (1 << 0) /* MIC L / ADC R Power Management */
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*/
101 #define ZTM(param) ((param & 0x3) << 4) /* ALC Zoro Crossing TimeOut */
102 #define WTM(param) (((param & 0x4) << 4) | ((param & 0x3) << 2))
105 #define ALC (1 << 5) /* ALC Enable */
106 #define LMTH0 (1 << 0) /* ALC Limiter / Recovery Level */
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)
115 #define BCKO_MASK (1 << 3)
116 #define BCKO_64 BCKO_MASK
118 #define DIF_MASK (3 << 0)
120 #define RIGHT_J (1 << 0)
121 #define LEFT_J (2 << 0)
129 #define FS_MASK (FS0 | FS1 | FS2 | FS3)
132 #define BST1 (1 << 3)
135 #define DACH (1 << 0)
138 * Playback Volume (table 39)
140 * max : 0x00 : +12.0 dB
142 * min : 0xFE : -115.0 dB
145 static const DECLARE_TLV_DB_SCALE(out_tlv
, -11550, 50, 1);
147 static const struct snd_kcontrol_new ak4642_snd_controls
[] = {
149 SOC_DOUBLE_R_TLV("Digital Playback Volume", L_DVC
, R_DVC
,
150 0, 0xFF, 1, out_tlv
),
153 static const struct snd_kcontrol_new ak4642_headphone_control
=
154 SOC_DAPM_SINGLE("Switch", PW_MGMT2
, 6, 1, 0);
156 static const struct snd_kcontrol_new ak4642_lout_mixer_controls
[] = {
157 SOC_DAPM_SINGLE("DACL", SG_SL1
, 4, 1, 0),
160 static const struct snd_soc_dapm_widget ak4642_dapm_widgets
[] = {
163 SND_SOC_DAPM_OUTPUT("HPOUTL"),
164 SND_SOC_DAPM_OUTPUT("HPOUTR"),
165 SND_SOC_DAPM_OUTPUT("LINEOUT"),
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
),
172 SND_SOC_DAPM_PGA("DACH", MD_CTL4
, 0, 0, NULL
, 0),
174 SND_SOC_DAPM_MIXER("LINEOUT Mixer", PW_MGMT1
, 3, 0,
175 &ak4642_lout_mixer_controls
[0],
176 ARRAY_SIZE(ak4642_lout_mixer_controls
)),
179 SND_SOC_DAPM_DAC("DAC", "HiFi Playback", PW_MGMT1
, 2, 0),
182 static const struct snd_soc_dapm_route ak4642_intercon
[] = {
185 {"HPOUTL", NULL
, "HPL Out"},
186 {"HPOUTR", NULL
, "HPR Out"},
187 {"LINEOUT", NULL
, "LINEOUT Mixer"},
189 {"HPL Out", NULL
, "Headphone Enable"},
190 {"HPR Out", NULL
, "Headphone Enable"},
192 {"Headphone Enable", "Switch", "DACH"},
194 {"DACH", NULL
, "DAC"},
196 {"LINEOUT Mixer", "DACL", "DAC"},
200 * ak4642 register cache
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 },
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 },
228 static int ak4642_dai_startup(struct snd_pcm_substream
*substream
,
229 struct snd_soc_dai
*dai
)
231 int is_play
= substream
->stream
== SNDRV_PCM_STREAM_PLAYBACK
;
232 struct snd_soc_codec
*codec
= dai
->codec
;
236 * start headphone output
239 * Audio I/F Format :MSB justified (ADC & DAC)
240 * Bass Boost Level : Middle
242 * This operation came from example code of
243 * "ASAHI KASEI AK4642" (japanese) manual p97.
245 snd_soc_write(codec
, L_IVC
, 0x91); /* volume */
246 snd_soc_write(codec
, R_IVC
, 0x91); /* volume */
252 * Audio I/F Format:MSB justified (ADC & DAC)
255 * ALC setting:Refer to Table 35
258 * This operation came from example code of
259 * "ASAHI KASEI AK4642" (japanese) manual p94.
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
);
271 static void ak4642_dai_shutdown(struct snd_pcm_substream
*substream
,
272 struct snd_soc_dai
*dai
)
274 int is_play
= substream
->stream
== SNDRV_PCM_STREAM_PLAYBACK
;
275 struct snd_soc_codec
*codec
= dai
->codec
;
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);
286 static int ak4642_dai_set_sysclk(struct snd_soc_dai
*codec_dai
,
287 int clk_id
, unsigned int freq
, int dir
)
289 struct snd_soc_codec
*codec
= codec_dai
->codec
;
303 pll
= PLL2
| PLL1
| PLL0
;
309 pll
= PLL3
| PLL2
| PLL0
;
314 snd_soc_update_bits(codec
, MD_CTL1
, PLL_MASK
, pll
);
319 static int ak4642_dai_set_fmt(struct snd_soc_dai
*dai
, unsigned int fmt
)
321 struct snd_soc_codec
*codec
= dai
->codec
;
325 data
= MCKO
| PMPLL
; /* use MCKO */
328 /* set master/slave audio interface */
329 switch (fmt
& SND_SOC_DAIFMT_MASTER_MASK
) {
330 case SND_SOC_DAIFMT_CBM_CFM
:
334 case SND_SOC_DAIFMT_CBS_CFS
:
339 snd_soc_update_bits(codec
, PW_MGMT2
, MS
| MCKO
| PMPLL
, data
);
340 snd_soc_update_bits(codec
, MD_CTL1
, BCKO_MASK
, bcko
);
344 switch (fmt
& SND_SOC_DAIFMT_FORMAT_MASK
) {
345 case SND_SOC_DAIFMT_LEFT_J
:
348 case SND_SOC_DAIFMT_I2S
:
352 * Please add RIGHT_J / DSP support here
357 snd_soc_update_bits(codec
, MD_CTL1
, DIF_MASK
, data
);
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
)
366 struct snd_soc_codec
*codec
= dai
->codec
;
369 switch (params_rate(params
)) {
389 rate
= FS2
| FS1
| FS0
;
395 rate
= FS3
| FS2
| FS1
;
401 rate
= FS3
| FS2
| FS1
| FS0
;
404 rate
= FS3
| FS1
| FS0
;
409 snd_soc_update_bits(codec
, MD_CTL2
, FS_MASK
, rate
);
414 static int ak4642_set_bias_level(struct snd_soc_codec
*codec
,
415 enum snd_soc_bias_level level
)
418 case SND_SOC_BIAS_OFF
:
419 snd_soc_write(codec
, PW_MGMT1
, 0x00);
422 snd_soc_update_bits(codec
, PW_MGMT1
, PMVCM
, PMVCM
);
425 codec
->dapm
.bias_level
= level
;
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
,
438 static struct snd_soc_dai_driver ak4642_dai
= {
439 .name
= "ak4642-hifi",
441 .stream_name
= "Playback",
444 .rates
= SNDRV_PCM_RATE_8000_48000
,
445 .formats
= SNDRV_PCM_FMTBIT_S16_LE
},
447 .stream_name
= "Capture",
450 .rates
= SNDRV_PCM_RATE_8000_48000
,
451 .formats
= SNDRV_PCM_FMTBIT_S16_LE
},
452 .ops
= &ak4642_dai_ops
,
453 .symmetric_rates
= 1,
456 static int ak4642_resume(struct snd_soc_codec
*codec
)
458 struct regmap
*regmap
= dev_get_regmap(codec
->dev
, NULL
);
460 regcache_mark_dirty(regmap
);
461 regcache_sync(regmap
);
466 static int ak4642_probe(struct snd_soc_codec
*codec
)
468 ak4642_set_bias_level(codec
, SND_SOC_BIAS_STANDBY
);
473 static int ak4642_remove(struct snd_soc_codec
*codec
)
475 ak4642_set_bias_level(codec
, SND_SOC_BIAS_OFF
);
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
),
492 static const struct regmap_config ak4642_regmap
= {
495 .max_register
= ARRAY_SIZE(ak4642_reg
) + 1,
496 .reg_defaults
= ak4642_reg
,
497 .num_reg_defaults
= ARRAY_SIZE(ak4642_reg
),
500 static const struct regmap_config ak4648_regmap
= {
503 .max_register
= ARRAY_SIZE(ak4648_reg
) + 1,
504 .reg_defaults
= ak4648_reg
,
505 .num_reg_defaults
= ARRAY_SIZE(ak4648_reg
),
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
)
512 struct device_node
*np
= i2c
->dev
.of_node
;
513 const struct regmap_config
*regmap_config
= NULL
;
514 struct regmap
*regmap
;
517 const struct of_device_id
*of_id
;
519 of_id
= of_match_device(ak4642_of_match
, &i2c
->dev
);
521 regmap_config
= of_id
->data
;
523 regmap_config
= (const struct regmap_config
*)id
->driver_data
;
526 if (!regmap_config
) {
527 dev_err(&i2c
->dev
, "Unknown device type\n");
531 regmap
= devm_regmap_init_i2c(i2c
, regmap_config
);
533 return PTR_ERR(regmap
);
535 return snd_soc_register_codec(&i2c
->dev
,
536 &soc_codec_dev_ak4642
, &ak4642_dai
, 1);
539 static int ak4642_i2c_remove(struct i2c_client
*client
)
541 snd_soc_unregister_codec(&client
->dev
);
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
},
551 MODULE_DEVICE_TABLE(of
, ak4642_of_match
);
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
},
559 MODULE_DEVICE_TABLE(i2c
, ak4642_i2c_id
);
561 static struct i2c_driver ak4642_i2c_driver
= {
563 .name
= "ak4642-codec",
564 .owner
= THIS_MODULE
,
565 .of_match_table
= ak4642_of_match
,
567 .probe
= ak4642_i2c_probe
,
568 .remove
= ak4642_i2c_remove
,
569 .id_table
= ak4642_i2c_id
,
572 module_i2c_driver(ak4642_i2c_driver
);
574 MODULE_DESCRIPTION("Soc AK4642 driver");
575 MODULE_AUTHOR("Kuninori Morimoto <morimoto.kuninori@renesas.com>");
576 MODULE_LICENSE("GPL");