diff options
Diffstat (limited to 'kernel/sound/soc/codecs/ak4642.c')
| -rw-r--r-- | kernel/sound/soc/codecs/ak4642.c | 642 |
1 files changed, 642 insertions, 0 deletions
diff --git a/kernel/sound/soc/codecs/ak4642.c b/kernel/sound/soc/codecs/ak4642.c new file mode 100644 index 000000000..13585e88f --- /dev/null +++ b/kernel/sound/soc/codecs/ak4642.c @@ -0,0 +1,642 @@ +/* + * ak4642.c -- AK4642/AK4643 ALSA Soc Audio driver + * + * Copyright (C) 2009 Renesas Solutions Corp. + * Kuninori Morimoto <morimoto.kuninori@renesas.com> + * + * Based on wm8731.c by Richard Purdie + * Based on ak4535.c by Richard Purdie + * Based on wm8753.c by Liam Girdwood + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +/* ** CAUTION ** + * + * This is very simple driver. + * It can use headphone output / stereo input only + * + * AK4642 is tested. + * AK4643 is tested. + * AK4648 is tested. + */ + +#include <linux/delay.h> +#include <linux/i2c.h> +#include <linux/slab.h> +#include <linux/of_device.h> +#include <linux/module.h> +#include <linux/regmap.h> +#include <sound/soc.h> +#include <sound/initval.h> +#include <sound/tlv.h> + +#define PW_MGMT1 0x00 +#define PW_MGMT2 0x01 +#define SG_SL1 0x02 +#define SG_SL2 0x03 +#define MD_CTL1 0x04 +#define MD_CTL2 0x05 +#define TIMER 0x06 +#define ALC_CTL1 0x07 +#define ALC_CTL2 0x08 +#define L_IVC 0x09 +#define L_DVC 0x0a +#define ALC_CTL3 0x0b +#define R_IVC 0x0c +#define R_DVC 0x0d +#define MD_CTL3 0x0e +#define MD_CTL4 0x0f +#define PW_MGMT3 0x10 +#define DF_S 0x11 +#define FIL3_0 0x12 +#define FIL3_1 0x13 +#define FIL3_2 0x14 +#define FIL3_3 0x15 +#define EQ_0 0x16 +#define EQ_1 0x17 +#define EQ_2 0x18 +#define EQ_3 0x19 +#define EQ_4 0x1a +#define EQ_5 0x1b +#define FIL1_0 0x1c +#define FIL1_1 0x1d +#define FIL1_2 0x1e +#define FIL1_3 0x1f +#define PW_MGMT4 0x20 +#define MD_CTL5 0x21 +#define LO_MS 0x22 +#define HP_MS 0x23 +#define SPK_MS 0x24 + +/* PW_MGMT1*/ +#define PMVCM (1 << 6) /* VCOM Power Management */ +#define PMMIN (1 << 5) /* MIN Input Power Management */ +#define PMDAC (1 << 2) /* DAC Power Management */ +#define PMADL (1 << 0) /* MIC Amp Lch and ADC Lch Power Management */ + +/* PW_MGMT2 */ +#define HPMTN (1 << 6) +#define PMHPL (1 << 5) +#define PMHPR (1 << 4) +#define MS (1 << 3) /* master/slave select */ +#define MCKO (1 << 1) +#define PMPLL (1 << 0) + +#define PMHP_MASK (PMHPL | PMHPR) +#define PMHP PMHP_MASK + +/* PW_MGMT3 */ +#define PMADR (1 << 0) /* MIC L / ADC R Power Management */ + +/* SG_SL1 */ +#define MINS (1 << 6) /* Switch from MIN to Speaker */ +#define DACL (1 << 4) /* Switch from DAC to Stereo or Receiver */ +#define PMMP (1 << 2) /* MPWR pin Power Management */ +#define MGAIN0 (1 << 0) /* MIC amp gain*/ + +/* SG_SL2 */ +#define LOPS (1 << 6) /* Stero Line-out Power Save Mode */ + +/* TIMER */ +#define ZTM(param) ((param & 0x3) << 4) /* ALC Zero Crossing TimeOut */ +#define WTM(param) (((param & 0x4) << 4) | ((param & 0x3) << 2)) + +/* ALC_CTL1 */ +#define ALC (1 << 5) /* ALC Enable */ +#define LMTH0 (1 << 0) /* ALC Limiter / Recovery Level */ + +/* MD_CTL1 */ +#define PLL3 (1 << 7) +#define PLL2 (1 << 6) +#define PLL1 (1 << 5) +#define PLL0 (1 << 4) +#define PLL_MASK (PLL3 | PLL2 | PLL1 | PLL0) + +#define BCKO_MASK (1 << 3) +#define BCKO_64 BCKO_MASK + +#define DIF_MASK (3 << 0) +#define DSP (0 << 0) +#define RIGHT_J (1 << 0) +#define LEFT_J (2 << 0) +#define I2S (3 << 0) + +/* MD_CTL2 */ +#define FS0 (1 << 0) +#define FS1 (1 << 1) +#define FS2 (1 << 2) +#define FS3 (1 << 5) +#define FS_MASK (FS0 | FS1 | FS2 | FS3) + +/* MD_CTL3 */ +#define BST1 (1 << 3) + +/* MD_CTL4 */ +#define DACH (1 << 0) + +struct ak4642_drvdata { + const struct regmap_config *regmap_config; + int extended_frequencies; +}; + +struct ak4642_priv { + const struct ak4642_drvdata *drvdata; +}; + +/* + * Playback Volume (table 39) + * + * max : 0x00 : +12.0 dB + * ( 0.5 dB step ) + * min : 0xFE : -115.0 dB + * mute: 0xFF + */ +static const DECLARE_TLV_DB_SCALE(out_tlv, -11550, 50, 1); + +static const struct snd_kcontrol_new ak4642_snd_controls[] = { + + SOC_DOUBLE_R_TLV("Digital Playback Volume", L_DVC, R_DVC, + 0, 0xFF, 1, out_tlv), + SOC_SINGLE("ALC Capture Switch", ALC_CTL1, 5, 1, 0), + SOC_SINGLE("ALC Capture ZC Switch", ALC_CTL1, 4, 1, 1), +}; + +static const struct snd_kcontrol_new ak4642_headphone_control = + SOC_DAPM_SINGLE("Switch", PW_MGMT2, 6, 1, 0); + +static const struct snd_kcontrol_new ak4642_lout_mixer_controls[] = { + SOC_DAPM_SINGLE("DACL", SG_SL1, 4, 1, 0), +}; + +/* event handlers */ +static int ak4642_lout_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + + switch (event) { + case SND_SOC_DAPM_PRE_PMD: + case SND_SOC_DAPM_PRE_PMU: + /* Power save mode ON */ + snd_soc_update_bits(codec, SG_SL2, LOPS, LOPS); + break; + case SND_SOC_DAPM_POST_PMU: + case SND_SOC_DAPM_POST_PMD: + /* Power save mode OFF */ + mdelay(300); + snd_soc_update_bits(codec, SG_SL2, LOPS, 0); + break; + } + + return 0; +} + +static const struct snd_soc_dapm_widget ak4642_dapm_widgets[] = { + + /* Outputs */ + SND_SOC_DAPM_OUTPUT("HPOUTL"), + SND_SOC_DAPM_OUTPUT("HPOUTR"), + SND_SOC_DAPM_OUTPUT("LINEOUT"), + + SND_SOC_DAPM_PGA("HPL Out", PW_MGMT2, 5, 0, NULL, 0), + SND_SOC_DAPM_PGA("HPR Out", PW_MGMT2, 4, 0, NULL, 0), + SND_SOC_DAPM_SWITCH("Headphone Enable", SND_SOC_NOPM, 0, 0, + &ak4642_headphone_control), + + SND_SOC_DAPM_PGA("DACH", MD_CTL4, 0, 0, NULL, 0), + + SND_SOC_DAPM_MIXER_E("LINEOUT Mixer", PW_MGMT1, 3, 0, + &ak4642_lout_mixer_controls[0], + ARRAY_SIZE(ak4642_lout_mixer_controls), + ak4642_lout_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD), + + /* DAC */ + SND_SOC_DAPM_DAC("DAC", NULL, PW_MGMT1, 2, 0), +}; + +static const struct snd_soc_dapm_route ak4642_intercon[] = { + + /* Outputs */ + {"HPOUTL", NULL, "HPL Out"}, + {"HPOUTR", NULL, "HPR Out"}, + {"LINEOUT", NULL, "LINEOUT Mixer"}, + + {"HPL Out", NULL, "Headphone Enable"}, + {"HPR Out", NULL, "Headphone Enable"}, + + {"Headphone Enable", "Switch", "DACH"}, + + {"DACH", NULL, "DAC"}, + + {"LINEOUT Mixer", "DACL", "DAC"}, + + { "DAC", NULL, "Playback" }, +}; + +/* + * ak4642 register cache + */ +static const struct reg_default ak4642_reg[] = { + { 0, 0x00 }, { 1, 0x00 }, { 2, 0x01 }, { 3, 0x00 }, + { 4, 0x02 }, { 5, 0x00 }, { 6, 0x00 }, { 7, 0x00 }, + { 8, 0xe1 }, { 9, 0xe1 }, { 10, 0x18 }, { 11, 0x00 }, + { 12, 0xe1 }, { 13, 0x18 }, { 14, 0x11 }, { 15, 0x08 }, + { 16, 0x00 }, { 17, 0x00 }, { 18, 0x00 }, { 19, 0x00 }, + { 20, 0x00 }, { 21, 0x00 }, { 22, 0x00 }, { 23, 0x00 }, + { 24, 0x00 }, { 25, 0x00 }, { 26, 0x00 }, { 27, 0x00 }, + { 28, 0x00 }, { 29, 0x00 }, { 30, 0x00 }, { 31, 0x00 }, + { 32, 0x00 }, { 33, 0x00 }, { 34, 0x00 }, { 35, 0x00 }, + { 36, 0x00 }, +}; + +static const struct reg_default ak4648_reg[] = { + { 0, 0x00 }, { 1, 0x00 }, { 2, 0x01 }, { 3, 0x00 }, + { 4, 0x02 }, { 5, 0x00 }, { 6, 0x00 }, { 7, 0x00 }, + { 8, 0xe1 }, { 9, 0xe1 }, { 10, 0x18 }, { 11, 0x00 }, + { 12, 0xe1 }, { 13, 0x18 }, { 14, 0x11 }, { 15, 0xb8 }, + { 16, 0x00 }, { 17, 0x00 }, { 18, 0x00 }, { 19, 0x00 }, + { 20, 0x00 }, { 21, 0x00 }, { 22, 0x00 }, { 23, 0x00 }, + { 24, 0x00 }, { 25, 0x00 }, { 26, 0x00 }, { 27, 0x00 }, + { 28, 0x00 }, { 29, 0x00 }, { 30, 0x00 }, { 31, 0x00 }, + { 32, 0x00 }, { 33, 0x00 }, { 34, 0x00 }, { 35, 0x00 }, + { 36, 0x00 }, { 37, 0x88 }, { 38, 0x88 }, { 39, 0x08 }, +}; + +static int ak4642_dai_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + int is_play = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + struct snd_soc_codec *codec = dai->codec; + + if (is_play) { + /* + * start headphone output + * + * PLL, Master Mode + * Audio I/F Format :MSB justified (ADC & DAC) + * Bass Boost Level : Middle + * + * This operation came from example code of + * "ASAHI KASEI AK4642" (japanese) manual p97. + */ + snd_soc_write(codec, L_IVC, 0x91); /* volume */ + snd_soc_write(codec, R_IVC, 0x91); /* volume */ + } else { + /* + * start stereo input + * + * PLL Master Mode + * Audio I/F Format:MSB justified (ADC & DAC) + * Pre MIC AMP:+20dB + * MIC Power On + * ALC setting:Refer to Table 35 + * ALC bit=“1” + * + * This operation came from example code of + * "ASAHI KASEI AK4642" (japanese) manual p94. + */ + snd_soc_update_bits(codec, SG_SL1, PMMP | MGAIN0, PMMP | MGAIN0); + snd_soc_write(codec, TIMER, ZTM(0x3) | WTM(0x3)); + snd_soc_write(codec, ALC_CTL1, ALC | LMTH0); + snd_soc_update_bits(codec, PW_MGMT1, PMADL, PMADL); + snd_soc_update_bits(codec, PW_MGMT3, PMADR, PMADR); + } + + return 0; +} + +static void ak4642_dai_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + int is_play = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + struct snd_soc_codec *codec = dai->codec; + + if (is_play) { + } else { + /* stop stereo input */ + snd_soc_update_bits(codec, PW_MGMT1, PMADL, 0); + snd_soc_update_bits(codec, PW_MGMT3, PMADR, 0); + snd_soc_update_bits(codec, ALC_CTL1, ALC, 0); + } +} + +static int ak4642_dai_set_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct ak4642_priv *priv = snd_soc_codec_get_drvdata(codec); + u8 pll; + int extended_freq = 0; + + switch (freq) { + case 11289600: + pll = PLL2; + break; + case 12288000: + pll = PLL2 | PLL0; + break; + case 12000000: + pll = PLL2 | PLL1; + break; + case 24000000: + pll = PLL2 | PLL1 | PLL0; + break; + case 13500000: + pll = PLL3 | PLL2; + break; + case 27000000: + pll = PLL3 | PLL2 | PLL0; + break; + case 19200000: + pll = PLL3; + extended_freq = 1; + break; + case 13000000: + pll = PLL3 | PLL2 | PLL1; + extended_freq = 1; + break; + case 26000000: + pll = PLL3 | PLL2 | PLL1 | PLL0; + extended_freq = 1; + break; + default: + return -EINVAL; + } + + if (extended_freq && !priv->drvdata->extended_frequencies) + return -EINVAL; + + snd_soc_update_bits(codec, MD_CTL1, PLL_MASK, pll); + + return 0; +} + +static int ak4642_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = dai->codec; + u8 data; + u8 bcko; + + data = MCKO | PMPLL; /* use MCKO */ + bcko = 0; + + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + data |= MS; + bcko = BCKO_64; + break; + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + return -EINVAL; + } + snd_soc_update_bits(codec, PW_MGMT2, MS | MCKO | PMPLL, data); + snd_soc_update_bits(codec, MD_CTL1, BCKO_MASK, bcko); + + /* format type */ + data = 0; + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_LEFT_J: + data = LEFT_J; + break; + case SND_SOC_DAIFMT_I2S: + data = I2S; + break; + /* FIXME + * Please add RIGHT_J / DSP support here + */ + default: + return -EINVAL; + } + snd_soc_update_bits(codec, MD_CTL1, DIF_MASK, data); + + return 0; +} + +static int ak4642_dai_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + u8 rate; + + switch (params_rate(params)) { + case 7350: + rate = FS2; + break; + case 8000: + rate = 0; + break; + case 11025: + rate = FS2 | FS0; + break; + case 12000: + rate = FS0; + break; + case 14700: + rate = FS2 | FS1; + break; + case 16000: + rate = FS1; + break; + case 22050: + rate = FS2 | FS1 | FS0; + break; + case 24000: + rate = FS1 | FS0; + break; + case 29400: + rate = FS3 | FS2 | FS1; + break; + case 32000: + rate = FS3 | FS1; + break; + case 44100: + rate = FS3 | FS2 | FS1 | FS0; + break; + case 48000: + rate = FS3 | FS1 | FS0; + break; + default: + return -EINVAL; + } + snd_soc_update_bits(codec, MD_CTL2, FS_MASK, rate); + + return 0; +} + +static int ak4642_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + switch (level) { + case SND_SOC_BIAS_OFF: + snd_soc_write(codec, PW_MGMT1, 0x00); + break; + default: + snd_soc_update_bits(codec, PW_MGMT1, PMVCM, PMVCM); + break; + } + codec->dapm.bias_level = level; + + return 0; +} + +static const struct snd_soc_dai_ops ak4642_dai_ops = { + .startup = ak4642_dai_startup, + .shutdown = ak4642_dai_shutdown, + .set_sysclk = ak4642_dai_set_sysclk, + .set_fmt = ak4642_dai_set_fmt, + .hw_params = ak4642_dai_hw_params, +}; + +static struct snd_soc_dai_driver ak4642_dai = { + .name = "ak4642-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE }, + .capture = { + .stream_name = "Capture", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE }, + .ops = &ak4642_dai_ops, + .symmetric_rates = 1, +}; + +static int ak4642_resume(struct snd_soc_codec *codec) +{ + struct regmap *regmap = dev_get_regmap(codec->dev, NULL); + + regcache_mark_dirty(regmap); + regcache_sync(regmap); + return 0; +} + +static struct snd_soc_codec_driver soc_codec_dev_ak4642 = { + .resume = ak4642_resume, + .set_bias_level = ak4642_set_bias_level, + .controls = ak4642_snd_controls, + .num_controls = ARRAY_SIZE(ak4642_snd_controls), + .dapm_widgets = ak4642_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(ak4642_dapm_widgets), + .dapm_routes = ak4642_intercon, + .num_dapm_routes = ARRAY_SIZE(ak4642_intercon), +}; + +static const struct regmap_config ak4642_regmap = { + .reg_bits = 8, + .val_bits = 8, + .max_register = ARRAY_SIZE(ak4642_reg) + 1, + .reg_defaults = ak4642_reg, + .num_reg_defaults = ARRAY_SIZE(ak4642_reg), +}; + +static const struct regmap_config ak4648_regmap = { + .reg_bits = 8, + .val_bits = 8, + .max_register = ARRAY_SIZE(ak4648_reg) + 1, + .reg_defaults = ak4648_reg, + .num_reg_defaults = ARRAY_SIZE(ak4648_reg), +}; + +static const struct ak4642_drvdata ak4642_drvdata = { + .regmap_config = &ak4642_regmap, +}; + +static const struct ak4642_drvdata ak4643_drvdata = { + .regmap_config = &ak4642_regmap, +}; + +static const struct ak4642_drvdata ak4648_drvdata = { + .regmap_config = &ak4648_regmap, + .extended_frequencies = 1, +}; + +static const struct of_device_id ak4642_of_match[]; +static int ak4642_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct device_node *np = i2c->dev.of_node; + const struct ak4642_drvdata *drvdata = NULL; + struct regmap *regmap; + struct ak4642_priv *priv; + + if (np) { + const struct of_device_id *of_id; + + of_id = of_match_device(ak4642_of_match, &i2c->dev); + if (of_id) + drvdata = of_id->data; + } else { + drvdata = (const struct ak4642_drvdata *)id->driver_data; + } + + if (!drvdata) { + dev_err(&i2c->dev, "Unknown device type\n"); + return -EINVAL; + } + + priv = devm_kzalloc(&i2c->dev, sizeof(*priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + + priv->drvdata = drvdata; + + i2c_set_clientdata(i2c, priv); + + regmap = devm_regmap_init_i2c(i2c, drvdata->regmap_config); + if (IS_ERR(regmap)) + return PTR_ERR(regmap); + + return snd_soc_register_codec(&i2c->dev, + &soc_codec_dev_ak4642, &ak4642_dai, 1); +} + +static int ak4642_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + return 0; +} + +static const struct of_device_id ak4642_of_match[] = { + { .compatible = "asahi-kasei,ak4642", .data = &ak4642_drvdata}, + { .compatible = "asahi-kasei,ak4643", .data = &ak4643_drvdata}, + { .compatible = "asahi-kasei,ak4648", .data = &ak4648_drvdata}, + {}, +}; +MODULE_DEVICE_TABLE(of, ak4642_of_match); + +static const struct i2c_device_id ak4642_i2c_id[] = { + { "ak4642", (kernel_ulong_t)&ak4642_drvdata }, + { "ak4643", (kernel_ulong_t)&ak4643_drvdata }, + { "ak4648", (kernel_ulong_t)&ak4648_drvdata }, + { } +}; +MODULE_DEVICE_TABLE(i2c, ak4642_i2c_id); + +static struct i2c_driver ak4642_i2c_driver = { + .driver = { + .name = "ak4642-codec", + .owner = THIS_MODULE, + .of_match_table = ak4642_of_match, + }, + .probe = ak4642_i2c_probe, + .remove = ak4642_i2c_remove, + .id_table = ak4642_i2c_id, +}; + +module_i2c_driver(ak4642_i2c_driver); + +MODULE_DESCRIPTION("Soc AK4642 driver"); +MODULE_AUTHOR("Kuninori Morimoto <morimoto.kuninori@renesas.com>"); +MODULE_LICENSE("GPL"); |