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)
470 ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP);
472 dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
476 ak4642_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
481 static int ak4642_remove(struct snd_soc_codec *codec)
483 ak4642_set_bias_level(codec, SND_SOC_BIAS_OFF);
487 static struct snd_soc_codec_driver soc_codec_dev_ak4642 = {
488 .probe = ak4642_probe,
489 .remove = ak4642_remove,
490 .resume = ak4642_resume,
491 .set_bias_level = ak4642_set_bias_level,
492 .controls = ak4642_snd_controls,
493 .num_controls = ARRAY_SIZE(ak4642_snd_controls),
494 .dapm_widgets = ak4642_dapm_widgets,
495 .num_dapm_widgets = ARRAY_SIZE(ak4642_dapm_widgets),
496 .dapm_routes = ak4642_intercon,
497 .num_dapm_routes = ARRAY_SIZE(ak4642_intercon),
500 static const struct regmap_config ak4642_regmap = {
503 .max_register = ARRAY_SIZE(ak4642_reg) + 1,
504 .reg_defaults = ak4642_reg,
505 .num_reg_defaults = ARRAY_SIZE(ak4642_reg),
508 static const struct regmap_config ak4648_regmap = {
511 .max_register = ARRAY_SIZE(ak4648_reg) + 1,
512 .reg_defaults = ak4648_reg,
513 .num_reg_defaults = ARRAY_SIZE(ak4648_reg),
516 static struct of_device_id ak4642_of_match[];
517 static int ak4642_i2c_probe(struct i2c_client *i2c,
518 const struct i2c_device_id *id)
520 struct device_node *np = i2c->dev.of_node;
521 const struct regmap_config *regmap_config = NULL;
522 struct regmap *regmap;
525 const struct of_device_id *of_id;
527 of_id = of_match_device(ak4642_of_match, &i2c->dev);
529 regmap_config = of_id->data;
531 regmap_config = (const struct regmap_config *)id->driver_data;
534 if (!regmap_config) {
535 dev_err(&i2c->dev, "Unknown device type\n");
539 regmap = devm_regmap_init_i2c(i2c, regmap_config);
541 return PTR_ERR(regmap);
543 return snd_soc_register_codec(&i2c->dev,
544 &soc_codec_dev_ak4642, &ak4642_dai, 1);
547 static int ak4642_i2c_remove(struct i2c_client *client)
549 snd_soc_unregister_codec(&client->dev);
553 static struct of_device_id ak4642_of_match[] = {
554 { .compatible = "asahi-kasei,ak4642", .data = &ak4642_regmap},
555 { .compatible = "asahi-kasei,ak4643", .data = &ak4642_regmap},
556 { .compatible = "asahi-kasei,ak4648", .data = &ak4648_regmap},
559 MODULE_DEVICE_TABLE(of, ak4642_of_match);
561 static const struct i2c_device_id ak4642_i2c_id[] = {
562 { "ak4642", (kernel_ulong_t)&ak4642_regmap },
563 { "ak4643", (kernel_ulong_t)&ak4642_regmap },
564 { "ak4648", (kernel_ulong_t)&ak4648_regmap },
567 MODULE_DEVICE_TABLE(i2c, ak4642_i2c_id);
569 static struct i2c_driver ak4642_i2c_driver = {
571 .name = "ak4642-codec",
572 .owner = THIS_MODULE,
573 .of_match_table = ak4642_of_match,
575 .probe = ak4642_i2c_probe,
576 .remove = ak4642_i2c_remove,
577 .id_table = ak4642_i2c_id,
580 module_i2c_driver(ak4642_i2c_driver);
582 MODULE_DESCRIPTION("Soc AK4642 driver");
583 MODULE_AUTHOR("Kuninori Morimoto <morimoto.kuninori@renesas.com>");
584 MODULE_LICENSE("GPL");