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-rw-r--r--sound/soc/codecs/ak4642.c590
1 files changed, 590 insertions, 0 deletions
diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c
new file mode 100644
index 00000000..b3e24f28
--- /dev/null
+++ b/sound/soc/codecs/ak4642.c
@@ -0,0 +1,590 @@
+/*
+ * 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/module.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*/
+
+/* TIMER */
+#define ZTM(param) ((param & 0x3) << 4) /* ALC Zoro 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)
+
+/*
+ * 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),
+};
+
+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),
+};
+
+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("LINEOUT Mixer", PW_MGMT1, 3, 0,
+ &ak4642_lout_mixer_controls[0],
+ ARRAY_SIZE(ak4642_lout_mixer_controls)),
+
+ /* DAC */
+ SND_SOC_DAPM_DAC("DAC", "HiFi Playback", 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"},
+};
+
+/* codec private data */
+struct ak4642_priv {
+ unsigned int sysclk;
+ enum snd_soc_control_type control_type;
+};
+
+/*
+ * ak4642 register cache
+ */
+static const u8 ak4642_reg[] = {
+ 0x00, 0x00, 0x01, 0x00,
+ 0x02, 0x00, 0x00, 0x00,
+ 0xe1, 0xe1, 0x18, 0x00,
+ 0xe1, 0x18, 0x11, 0x08,
+ 0x00, 0x00, 0x00, 0x00,
+ 0x00, 0x00, 0x00, 0x00,
+ 0x00, 0x00, 0x00, 0x00,
+ 0x00, 0x00, 0x00, 0x00,
+ 0x00, 0x00, 0x00, 0x00,
+ 0x00,
+};
+
+static const u8 ak4648_reg[] = {
+ 0x00, 0x00, 0x01, 0x00,
+ 0x02, 0x00, 0x00, 0x00,
+ 0xe1, 0xe1, 0x18, 0x00,
+ 0xe1, 0x18, 0x11, 0xb8,
+ 0x00, 0x00, 0x00, 0x00,
+ 0x00, 0x00, 0x00, 0x00,
+ 0x00, 0x00, 0x00, 0x00,
+ 0x00, 0x00, 0x00, 0x00,
+ 0x00, 0x00, 0x00, 0x00,
+ 0x00, 0x88, 0x88, 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_write(codec, SG_SL1, 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;
+ u8 pll;
+
+ 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;
+ default:
+ 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;
+ break;
+ }
+ 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;
+ break;
+ }
+ 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 = 1,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_8000_48000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE },
+ .capture = {
+ .stream_name = "Capture",
+ .channels_min = 1,
+ .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)
+{
+ snd_soc_cache_sync(codec);
+ return 0;
+}
+
+
+static int ak4642_probe(struct snd_soc_codec *codec)
+{
+ struct ak4642_priv *ak4642 = snd_soc_codec_get_drvdata(codec);
+ int ret;
+
+ ret = snd_soc_codec_set_cache_io(codec, 8, 8, ak4642->control_type);
+ if (ret < 0) {
+ dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
+ return ret;
+ }
+
+ snd_soc_add_codec_controls(codec, ak4642_snd_controls,
+ ARRAY_SIZE(ak4642_snd_controls));
+
+ ak4642_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+ return 0;
+}
+
+static int ak4642_remove(struct snd_soc_codec *codec)
+{
+ ak4642_set_bias_level(codec, SND_SOC_BIAS_OFF);
+ return 0;
+}
+
+static struct snd_soc_codec_driver soc_codec_dev_ak4642 = {
+ .probe = ak4642_probe,
+ .remove = ak4642_remove,
+ .resume = ak4642_resume,
+ .set_bias_level = ak4642_set_bias_level,
+ .reg_cache_default = ak4642_reg, /* ak4642 reg */
+ .reg_cache_size = ARRAY_SIZE(ak4642_reg), /* ak4642 reg */
+ .reg_word_size = sizeof(u8),
+ .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 struct snd_soc_codec_driver soc_codec_dev_ak4648 = {
+ .probe = ak4642_probe,
+ .remove = ak4642_remove,
+ .resume = ak4642_resume,
+ .set_bias_level = ak4642_set_bias_level,
+ .reg_cache_default = ak4648_reg, /* ak4648 reg */
+ .reg_cache_size = ARRAY_SIZE(ak4648_reg), /* ak4648 reg */
+ .reg_word_size = sizeof(u8),
+ .dapm_widgets = ak4642_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(ak4642_dapm_widgets),
+ .dapm_routes = ak4642_intercon,
+ .num_dapm_routes = ARRAY_SIZE(ak4642_intercon),
+};
+
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+static __devinit int ak4642_i2c_probe(struct i2c_client *i2c,
+ const struct i2c_device_id *id)
+{
+ struct ak4642_priv *ak4642;
+ int ret;
+
+ ak4642 = devm_kzalloc(&i2c->dev, sizeof(struct ak4642_priv),
+ GFP_KERNEL);
+ if (!ak4642)
+ return -ENOMEM;
+
+ i2c_set_clientdata(i2c, ak4642);
+ ak4642->control_type = SND_SOC_I2C;
+
+ ret = snd_soc_register_codec(&i2c->dev,
+ (struct snd_soc_codec_driver *)id->driver_data,
+ &ak4642_dai, 1);
+ return ret;
+}
+
+static __devexit int ak4642_i2c_remove(struct i2c_client *client)
+{
+ snd_soc_unregister_codec(&client->dev);
+ return 0;
+}
+
+static const struct i2c_device_id ak4642_i2c_id[] = {
+ { "ak4642", (kernel_ulong_t)&soc_codec_dev_ak4642 },
+ { "ak4643", (kernel_ulong_t)&soc_codec_dev_ak4642 },
+ { "ak4648", (kernel_ulong_t)&soc_codec_dev_ak4648 },
+ { }
+};
+MODULE_DEVICE_TABLE(i2c, ak4642_i2c_id);
+
+static struct i2c_driver ak4642_i2c_driver = {
+ .driver = {
+ .name = "ak4642-codec",
+ .owner = THIS_MODULE,
+ },
+ .probe = ak4642_i2c_probe,
+ .remove = __devexit_p(ak4642_i2c_remove),
+ .id_table = ak4642_i2c_id,
+};
+#endif
+
+static int __init ak4642_modinit(void)
+{
+ int ret = 0;
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+ ret = i2c_add_driver(&ak4642_i2c_driver);
+#endif
+ return ret;
+
+}
+module_init(ak4642_modinit);
+
+static void __exit ak4642_exit(void)
+{
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+ i2c_del_driver(&ak4642_i2c_driver);
+#endif
+
+}
+module_exit(ak4642_exit);
+
+MODULE_DESCRIPTION("Soc AK4642 driver");
+MODULE_AUTHOR("Kuninori Morimoto <morimoto.kuninori@renesas.com>");
+MODULE_LICENSE("GPL");
generated by cgit (git 2.34.1) at 2024-12-20 07:47:53 +0000