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 Zero 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)
137 struct ak4642_drvdata {
138 const struct regmap_config *regmap_config;
139 int extended_frequencies;
143 const struct ak4642_drvdata *drvdata;
147 * Playback Volume (table 39)
149 * max : 0x00 : +12.0 dB
151 * min : 0xFE : -115.0 dB
154 static const DECLARE_TLV_DB_SCALE(out_tlv, -11550, 50, 1);
156 static const struct snd_kcontrol_new ak4642_snd_controls[] = {
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),
164 static const struct snd_kcontrol_new ak4642_headphone_control =
165 SOC_DAPM_SINGLE("Switch", PW_MGMT2, 6, 1, 0);
167 static const struct snd_kcontrol_new ak4642_lout_mixer_controls[] = {
168 SOC_DAPM_SINGLE("DACL", SG_SL1, 4, 1, 0),
171 static const struct snd_soc_dapm_widget ak4642_dapm_widgets[] = {
174 SND_SOC_DAPM_OUTPUT("HPOUTL"),
175 SND_SOC_DAPM_OUTPUT("HPOUTR"),
176 SND_SOC_DAPM_OUTPUT("LINEOUT"),
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),
183 SND_SOC_DAPM_PGA("DACH", MD_CTL4, 0, 0, NULL, 0),
185 SND_SOC_DAPM_MIXER("LINEOUT Mixer", PW_MGMT1, 3, 0,
186 &ak4642_lout_mixer_controls[0],
187 ARRAY_SIZE(ak4642_lout_mixer_controls)),
190 SND_SOC_DAPM_DAC("DAC", "HiFi Playback", PW_MGMT1, 2, 0),
193 static const struct snd_soc_dapm_route ak4642_intercon[] = {
196 {"HPOUTL", NULL, "HPL Out"},
197 {"HPOUTR", NULL, "HPR Out"},
198 {"LINEOUT", NULL, "LINEOUT Mixer"},
200 {"HPL Out", NULL, "Headphone Enable"},
201 {"HPR Out", NULL, "Headphone Enable"},
203 {"Headphone Enable", "Switch", "DACH"},
205 {"DACH", NULL, "DAC"},
207 {"LINEOUT Mixer", "DACL", "DAC"},
211 * ak4642 register cache
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 },
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 },
239 static int ak4642_dai_startup(struct snd_pcm_substream *substream,
240 struct snd_soc_dai *dai)
242 int is_play = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
243 struct snd_soc_codec *codec = dai->codec;
247 * start headphone output
250 * Audio I/F Format :MSB justified (ADC & DAC)
251 * Bass Boost Level : Middle
253 * This operation came from example code of
254 * "ASAHI KASEI AK4642" (japanese) manual p97.
256 snd_soc_write(codec, L_IVC, 0x91); /* volume */
257 snd_soc_write(codec, R_IVC, 0x91); /* volume */
263 * Audio I/F Format:MSB justified (ADC & DAC)
266 * ALC setting:Refer to Table 35
269 * This operation came from example code of
270 * "ASAHI KASEI AK4642" (japanese) manual p94.
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);
282 static void ak4642_dai_shutdown(struct snd_pcm_substream *substream,
283 struct snd_soc_dai *dai)
285 int is_play = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
286 struct snd_soc_codec *codec = dai->codec;
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);
297 static int ak4642_dai_set_sysclk(struct snd_soc_dai *codec_dai,
298 int clk_id, unsigned int freq, int dir)
300 struct snd_soc_codec *codec = codec_dai->codec;
301 struct ak4642_priv *priv = snd_soc_codec_get_drvdata(codec);
303 int extended_freq = 0;
316 pll = PLL2 | PLL1 | PLL0;
322 pll = PLL3 | PLL2 | PLL0;
329 pll = PLL3 | PLL2 | PLL1;
333 pll = PLL3 | PLL2 | PLL1 | PLL0;
340 if (extended_freq && !priv->drvdata->extended_frequencies)
343 snd_soc_update_bits(codec, MD_CTL1, PLL_MASK, pll);
348 static int ak4642_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
350 struct snd_soc_codec *codec = dai->codec;
354 data = MCKO | PMPLL; /* use MCKO */
357 /* set master/slave audio interface */
358 switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
359 case SND_SOC_DAIFMT_CBM_CFM:
363 case SND_SOC_DAIFMT_CBS_CFS:
368 snd_soc_update_bits(codec, PW_MGMT2, MS | MCKO | PMPLL, data);
369 snd_soc_update_bits(codec, MD_CTL1, BCKO_MASK, bcko);
373 switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
374 case SND_SOC_DAIFMT_LEFT_J:
377 case SND_SOC_DAIFMT_I2S:
381 * Please add RIGHT_J / DSP support here
386 snd_soc_update_bits(codec, MD_CTL1, DIF_MASK, data);
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)
395 struct snd_soc_codec *codec = dai->codec;
398 switch (params_rate(params)) {
418 rate = FS2 | FS1 | FS0;
424 rate = FS3 | FS2 | FS1;
430 rate = FS3 | FS2 | FS1 | FS0;
433 rate = FS3 | FS1 | FS0;
438 snd_soc_update_bits(codec, MD_CTL2, FS_MASK, rate);
443 static int ak4642_set_bias_level(struct snd_soc_codec *codec,
444 enum snd_soc_bias_level level)
447 case SND_SOC_BIAS_OFF:
448 snd_soc_write(codec, PW_MGMT1, 0x00);
451 snd_soc_update_bits(codec, PW_MGMT1, PMVCM, PMVCM);
454 codec->dapm.bias_level = level;
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,
467 static struct snd_soc_dai_driver ak4642_dai = {
468 .name = "ak4642-hifi",
470 .stream_name = "Playback",
473 .rates = SNDRV_PCM_RATE_8000_48000,
474 .formats = SNDRV_PCM_FMTBIT_S16_LE },
476 .stream_name = "Capture",
479 .rates = SNDRV_PCM_RATE_8000_48000,
480 .formats = SNDRV_PCM_FMTBIT_S16_LE },
481 .ops = &ak4642_dai_ops,
482 .symmetric_rates = 1,
485 static int ak4642_resume(struct snd_soc_codec *codec)
487 struct regmap *regmap = dev_get_regmap(codec->dev, NULL);
489 regcache_mark_dirty(regmap);
490 regcache_sync(regmap);
494 static struct snd_soc_codec_driver soc_codec_dev_ak4642 = {
495 .resume = ak4642_resume,
496 .set_bias_level = ak4642_set_bias_level,
497 .controls = ak4642_snd_controls,
498 .num_controls = ARRAY_SIZE(ak4642_snd_controls),
499 .dapm_widgets = ak4642_dapm_widgets,
500 .num_dapm_widgets = ARRAY_SIZE(ak4642_dapm_widgets),
501 .dapm_routes = ak4642_intercon,
502 .num_dapm_routes = ARRAY_SIZE(ak4642_intercon),
505 static const struct regmap_config ak4642_regmap = {
508 .max_register = ARRAY_SIZE(ak4642_reg) + 1,
509 .reg_defaults = ak4642_reg,
510 .num_reg_defaults = ARRAY_SIZE(ak4642_reg),
513 static const struct regmap_config ak4648_regmap = {
516 .max_register = ARRAY_SIZE(ak4648_reg) + 1,
517 .reg_defaults = ak4648_reg,
518 .num_reg_defaults = ARRAY_SIZE(ak4648_reg),
521 static const struct ak4642_drvdata ak4642_drvdata = {
522 .regmap_config = &ak4642_regmap,
525 static const struct ak4642_drvdata ak4643_drvdata = {
526 .regmap_config = &ak4642_regmap,
529 static const struct ak4642_drvdata ak4648_drvdata = {
530 .regmap_config = &ak4648_regmap,
531 .extended_frequencies = 1,
534 static const struct of_device_id ak4642_of_match[];
535 static int ak4642_i2c_probe(struct i2c_client *i2c,
536 const struct i2c_device_id *id)
538 struct device_node *np = i2c->dev.of_node;
539 const struct ak4642_drvdata *drvdata = NULL;
540 struct regmap *regmap;
541 struct ak4642_priv *priv;
544 const struct of_device_id *of_id;
546 of_id = of_match_device(ak4642_of_match, &i2c->dev);
548 drvdata = of_id->data;
550 drvdata = (const struct ak4642_drvdata *)id->driver_data;
554 dev_err(&i2c->dev, "Unknown device type\n");
558 priv = devm_kzalloc(&i2c->dev, sizeof(*priv), GFP_KERNEL);
562 priv->drvdata = drvdata;
564 i2c_set_clientdata(i2c, priv);
566 regmap = devm_regmap_init_i2c(i2c, drvdata->regmap_config);
568 return PTR_ERR(regmap);
570 return snd_soc_register_codec(&i2c->dev,
571 &soc_codec_dev_ak4642, &ak4642_dai, 1);
574 static int ak4642_i2c_remove(struct i2c_client *client)
576 snd_soc_unregister_codec(&client->dev);
580 static const struct of_device_id ak4642_of_match[] = {
581 { .compatible = "asahi-kasei,ak4642", .data = &ak4642_drvdata},
582 { .compatible = "asahi-kasei,ak4643", .data = &ak4643_drvdata},
583 { .compatible = "asahi-kasei,ak4648", .data = &ak4648_drvdata},
586 MODULE_DEVICE_TABLE(of, ak4642_of_match);
588 static const struct i2c_device_id ak4642_i2c_id[] = {
589 { "ak4642", (kernel_ulong_t)&ak4642_drvdata },
590 { "ak4643", (kernel_ulong_t)&ak4643_drvdata },
591 { "ak4648", (kernel_ulong_t)&ak4648_drvdata },
594 MODULE_DEVICE_TABLE(i2c, ak4642_i2c_id);
596 static struct i2c_driver ak4642_i2c_driver = {
598 .name = "ak4642-codec",
599 .owner = THIS_MODULE,
600 .of_match_table = ak4642_of_match,
602 .probe = ak4642_i2c_probe,
603 .remove = ak4642_i2c_remove,
604 .id_table = ak4642_i2c_id,
607 module_i2c_driver(ak4642_i2c_driver);
609 MODULE_DESCRIPTION("Soc AK4642 driver");
610 MODULE_AUTHOR("Kuninori Morimoto <morimoto.kuninori@renesas.com>");
611 MODULE_LICENSE("GPL");