[PATCH 1/3] ASoC: add 88pm860x codec driver
Haojian Zhuang
haojian.zhuang at marvell.com
Thu Aug 12 06:01:47 CEST 2010
Add 88PM860x codec driver. 88PM860x codec supports two interfaces. And it
also supports headset/mic/hook/short detection.
Signed-off-by: Haojian Zhuang <haojian.zhuang at marvell.com>
---
sound/soc/codecs/88pm860x-codec.c | 1486 +++++++++++++++++++++++++++++++++++++
sound/soc/codecs/88pm860x-codec.h | 97 +++
sound/soc/codecs/Kconfig | 4 +
sound/soc/codecs/Makefile | 2 +
4 files changed, 1589 insertions(+), 0 deletions(-)
create mode 100644 sound/soc/codecs/88pm860x-codec.c
create mode 100644 sound/soc/codecs/88pm860x-codec.h
diff --git a/sound/soc/codecs/88pm860x-codec.c
b/sound/soc/codecs/88pm860x-codec.c
new file mode 100644
index 0000000..01d19e9
--- /dev/null
+++ b/sound/soc/codecs/88pm860x-codec.c
@@ -0,0 +1,1486 @@
+/*
+ * 88pm860x-codec.c -- 88PM860x ALSA SoC Audio Driver
+ *
+ * Copyright 2010 Marvell International Ltd.
+ * Author: Haojian Zhuang <haojian.zhuang at marvell.com>
+ *
+ * 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.
+ */
+
+#include <linux/kernel.h>
+#include <linux/module.h>
+#include <linux/i2c.h>
+#include <linux/platform_device.h>
+#include <linux/mfd/88pm860x.h>
+#include <linux/slab.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/tlv.h>
+#include <sound/initval.h>
+#include <sound/jack.h>
+
+#include "88pm860x-codec.h"
+
+#define MAX_NAME_LEN 20
+#define REG_CACHE_SIZE 0x40
+#define REG_CACHE_BASE 0xb0
+
+/* Status Register 1 (0x01) */
+#define REG_STATUS_1 0x01
+#define MIC_STATUS (1 << 7)
+#define HOOK_STATUS (1 << 6)
+#define HEADSET_STATUS (1 << 5)
+
+/* Mic Detection Register (0x37) */
+#define REG_MIC_DET 0x37
+#define CONTINUOUS_POLLING (3 << 1)
+#define EN_MIC_DET (1 << 0)
+#define MICDET_MASK 0x07
+
+/* Headset Detection Register (0x38) */
+#define REG_HS_DET 0x38
+#define EN_HS_DET (1 << 0)
+
+/* Misc2 Register (0x42) */
+#define REG_MISC2 0x42
+#define AUDIO_PLL (1 << 5)
+#define AUDIO_SECTION_RESET (1 << 4)
+#define AUDIO_SECTION_ON (1 << 3)
+
+/* PCM Interface Register 2 (0xb1) */
+#define PCM_INF2_BCLK (1 << 6) /* Bit clock polarity */
+#define PCM_INF2_FS (1 << 5) /* Frame Sync polarity */
+#define PCM_INF2_MASTER (1 << 4) /* Master / Slave */
+#define PCM_INF2_18WL (1 << 3) /* 18 / 16 bits */
+#define PCM_GENERAL_I2S 0
+#define PCM_EXACT_I2S 1
+#define PCM_LEFT_I2S 2
+#define PCM_RIGHT_I2S 3
+#define PCM_SHORT_FS 4
+#define PCM_LONG_FS 5
+#define PCM_MODE_MASK 7
+
+/* I2S Interface Register 4 (0xbe) */
+#define I2S_EQU_BYP (1 << 6)
+
+/* DAC Offset Register (0xcb) */
+#define DAC_MUTE (1 << 7)
+#define MUTE_LEFT (1 << 6)
+#define MUTE_RIGHT (1 << 2)
+
+/* ADC Analog Register 1 (0xd0) */
+#define REG_ADC_ANA_1 0xd0
+#define MIC1BIAS_MASK 0x60
+
+/* Earpiece/Speaker Control Register 2 (0xda) */
+#define REG_EAR2 0xda
+#define RSYNC_CHANGE (1 << 2)
+
+/* Audio Supplies Register 2 (0xdc) */
+#define REG_SUPPLIES2 0xdc
+#define LDO15_READY (1 << 4)
+#define LDO15_EN (1 << 3)
+#define CPUMP_READY (1 << 2)
+#define CPUMP_EN (1 << 1)
+#define AUDIO_EN (1 << 0)
+#define SUPPLY_MASK (LDO15_EN | CPUMP_EN | AUDIO_EN)
+
+/* Audio Enable Register 1 (0xdd) */
+#define ADC_MOD_RIGHT (1 << 1)
+#define ADC_MOD_LEFT (1 << 0)
+
+/* Audio Enable Register 2 (0xde) */
+#define ADC_LEFT (1 << 5)
+#define ADC_RIGHT (1 << 4)
+
+/* DAC Enable Register 2 (0xe1) */
+#define DAC_LEFT (1 << 5)
+#define DAC_RIGHT (1 << 4)
+#define MODULATOR (1 << 3)
+
+/* Shorts Register (0xeb) */
+#define REG_SHORTS 0xeb
+#define CLR_SHORT_LO2 (1 << 7)
+#define SHORT_LO2 (1 << 6)
+#define CLR_SHORT_LO1 (1 << 5)
+#define SHORT_LO1 (1 << 4)
+#define CLR_SHORT_HS2 (1 << 3)
+#define SHORT_HS2 (1 << 2)
+#define CLR_SHORT_HS1 (1 << 1)
+#define SHORT_HS1 (1 << 0)
+
+/*
+ * This widget should be just after DAC & PGA in DAPM power-on sequence and
+ * before DAC & PGA in DAPM power-off sequence.
+ */
+#define PM860X_DAPM_OUTPUT(wname, wevent) \
+{ .id = snd_soc_dapm_pga, .name = wname, .reg = SND_SOC_NOPM, \
+ .shift = 0, .invert = 0, .kcontrols = NULL, \
+ .num_kcontrols = 0, .event = wevent, \
+ .event_flags = SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD, }
+
+struct pm860x_det {
+ struct snd_soc_jack *hp_jack;
+ struct snd_soc_jack *mic_jack;
+ int hp_det;
+ int mic_det;
+ int hook_det;
+ int hs_shrt;
+ int lo_shrt;
+};
+
+struct pm860x_priv {
+ unsigned int sysclk;
+ unsigned int pcmclk;
+ unsigned int dir;
+ unsigned int filter;
+ struct snd_soc_codec *codec;
+ struct i2c_client *i2c;
+ struct pm860x_chip *chip;
+ struct pm860x_det det;
+
+ int irq[4];
+ unsigned char name[4][MAX_NAME_LEN];
+ unsigned char reg_cache[REG_CACHE_SIZE];
+};
+
+/* -9450dB to 0dB in 150dB steps ( mute instead of -9450dB) */
+static const DECLARE_TLV_DB_SCALE(dpga_tlv, -9450, 150, 1);
+
+/* -9dB to 0db in 3dB steps */
+static const DECLARE_TLV_DB_SCALE(adc_tlv, -900, 300, 0);
+
+/* {-23, -17, -13.5, -11, -9, -6, -3, 0}dB */
+static const unsigned int mic_tlv[] = {
+ TLV_DB_RANGE_HEAD(5),
+ 0, 0, TLV_DB_SCALE_ITEM(-2300, 0, 0),
+ 1, 1, TLV_DB_SCALE_ITEM(-1700, 0, 0),
+ 2, 2, TLV_DB_SCALE_ITEM(-1350, 0, 0),
+ 3, 3, TLV_DB_SCALE_ITEM(-1100, 0, 0),
+ 4, 7, TLV_DB_SCALE_ITEM(-900, 300, 0),
+};
+
+/* {0, 0, 0, -6, 0, 6, 12, 18}dB */
+static const unsigned int aux_tlv[] = {
+ TLV_DB_RANGE_HEAD(2),
+ 0, 2, TLV_DB_SCALE_ITEM(0, 0, 0),
+ 3, 7, TLV_DB_SCALE_ITEM(-600, 600, 0),
+};
+
+/* {-16, -13, -10, -7, -5.2, -3,3, -2.2, 0}dB, mute instead of -16dB */
+static const unsigned int out_tlv[] = {
+ TLV_DB_RANGE_HEAD(4),
+ 0, 3, TLV_DB_SCALE_ITEM(-1600, 300, 1),
+ 4, 4, TLV_DB_SCALE_ITEM(-520, 0, 0),
+ 5, 5, TLV_DB_SCALE_ITEM(-330, 0, 0),
+ 6, 7, TLV_DB_SCALE_ITEM(-220, 220, 0),
+};
+
+static const unsigned int st_tlv[] = {
+ TLV_DB_RANGE_HEAD(8),
+ 0, 1, TLV_DB_SCALE_ITEM(-12041, 602, 0),
+ 2, 3, TLV_DB_SCALE_ITEM(-11087, 250, 0),
+ 4, 5, TLV_DB_SCALE_ITEM(-10643, 158, 0),
+ 6, 7, TLV_DB_SCALE_ITEM(-10351, 116, 0),
+ 8, 9, TLV_DB_SCALE_ITEM(-10133, 92, 0),
+ 10, 13, TLV_DB_SCALE_ITEM(-9958, 70, 0),
+ 14, 17, TLV_DB_SCALE_ITEM(-9689, 53, 0),
+ 18, 271, TLV_DB_SCALE_ITEM(-9484, 37, 0),
+};
+
+/* Sidetone Gain = M * 2^(-5-N) */
+struct st_gain {
+ unsigned int db;
+ unsigned int m;
+ unsigned int n;
+};
+
+static struct st_gain st_table[] = {
+ {-12041, 1, 15}, {-11439, 1, 14}, {-11087, 3, 15}, {-10837, 1, 13},
+ {-10643, 5, 15}, {-10485, 3, 14}, {-10351, 7, 15}, {-10235, 1, 12},
+ {-10133, 9, 15}, {-10041, 5, 14}, { -9958, 11, 15}, { -9883, 3, 13},
+ { -9813, 13, 15}, { -9749, 7, 14}, { -9689, 15, 15}, { -9633, 1, 11},
+ { -9580, 17, 15}, { -9531, 9, 14}, { -9484, 19, 15}, { -9439, 5, 13},
+ { -9397, 21, 15}, { -9356, 11, 14}, { -9318, 23, 15}, { -9281, 3, 12},
+ { -9245, 25, 15}, { -9211, 13, 14}, { -9178, 27, 15}, { -9147, 7, 13},
+ { -9116, 29, 15}, { -9087, 15, 14}, { -9058, 31, 15}, { -9031, 1, 10},
+ { -8978, 17, 14}, { -8929, 9, 13}, { -8882, 19, 14}, { -8837, 5, 12},
+ { -8795, 21, 14}, { -8754, 11, 13}, { -8716, 23, 14}, { -8679, 3, 11},
+ { -8643, 25, 14}, { -8609, 13, 13}, { -8576, 27, 14}, { -8545, 7, 12},
+ { -8514, 29, 14}, { -8485, 15, 13}, { -8456, 31, 14}, { -8429, 1, 9},
+ { -8376, 17, 13}, { -8327, 9, 12}, { -8280, 19, 13}, { -8235, 5, 11},
+ { -8193, 21, 13}, { -8152, 11, 12}, { -8114, 23, 13}, { -8077, 3, 10},
+ { -8041, 25, 13}, { -8007, 13, 12}, { -7974, 27, 13}, { -7943, 7, 11},
+ { -7912, 29, 13}, { -7883, 15, 12}, { -7854, 31, 13}, { -7827, 1, 8},
+ { -7774, 17, 12}, { -7724, 9, 11}, { -7678, 19, 12}, { -7633, 5, 10},
+ { -7591, 21, 12}, { -7550, 11, 11}, { -7512, 23, 12}, { -7475, 3, 9},
+ { -7439, 25, 12}, { -7405, 13, 11}, { -7372, 27, 12}, { -7341, 7, 10},
+ { -7310, 29, 12}, { -7281, 15, 11}, { -7252, 31, 12}, { -7225, 1, 7},
+ { -7172, 17, 11}, { -7122, 9, 10}, { -7075, 19, 11}, { -7031, 5, 9},
+ { -6989, 21, 11}, { -6948, 11, 10}, { -6910, 23, 11}, { -6873, 3, 8},
+ { -6837, 25, 11}, { -6803, 13, 10}, { -6770, 27, 11}, { -6739, 7, 9},
+ { -6708, 29, 11}, { -6679, 15, 10}, { -6650, 31, 11}, { -6623, 1, 6},
+ { -6570, 17, 10}, { -6520, 9, 9}, { -6473, 19, 10}, { -6429, 5, 8},
+ { -6386, 21, 10}, { -6346, 11, 9}, { -6307, 23, 10}, { -6270, 3, 7},
+ { -6235, 25, 10}, { -6201, 13, 9}, { -6168, 27, 10}, { -6137, 7, 8},
+ { -6106, 29, 10}, { -6077, 15, 9}, { -6048, 31, 10}, { -6021, 1, 5},
+ { -5968, 17, 9}, { -5918, 9, 8}, { -5871, 19, 9}, { -5827, 5, 7},
+ { -5784, 21, 9}, { -5744, 11, 8}, { -5705, 23, 9}, { -5668, 3, 6},
+ { -5633, 25, 9}, { -5599, 13, 8}, { -5566, 27, 9}, { -5535, 7, 7},
+ { -5504, 29, 9}, { -5475, 15, 8}, { -5446, 31, 9}, { -5419, 1, 4},
+ { -5366, 17, 8}, { -5316, 9, 7}, { -5269, 19, 8}, { -5225, 5, 6},
+ { -5182, 21, 8}, { -5142, 11, 7}, { -5103, 23, 8}, { -5066, 3, 5},
+ { -5031, 25, 8}, { -4997, 13, 7}, { -4964, 27, 8}, { -4932, 7, 6},
+ { -4902, 29, 8}, { -4873, 15, 7}, { -4844, 31, 8}, { -4816, 1, 3},
+ { -4764, 17, 7}, { -4714, 9, 6}, { -4667, 19, 7}, { -4623, 5, 5},
+ { -4580, 21, 7}, { -4540, 11, 6}, { -4501, 23, 7}, { -4464, 3, 4},
+ { -4429, 25, 7}, { -4395, 13, 6}, { -4362, 27, 7}, { -4330, 7, 5},
+ { -4300, 29, 7}, { -4270, 15, 6}, { -4242, 31, 7}, { -4214, 1, 2},
+ { -4162, 17, 6}, { -4112, 9, 5}, { -4065, 19, 6}, { -4021, 5, 4},
+ { -3978, 21, 6}, { -3938, 11, 5}, { -3899, 23, 6}, { -3862, 3, 3},
+ { -3827, 25, 6}, { -3793, 13, 5}, { -3760, 27, 6}, { -3728, 7, 4},
+ { -3698, 29, 6}, { -3668, 15, 5}, { -3640, 31, 6}, { -3612, 1, 1},
+ { -3560, 17, 5}, { -3510, 9, 4}, { -3463, 19, 5}, { -3419, 5, 3},
+ { -3376, 21, 5}, { -3336, 11, 4}, { -3297, 23, 5}, { -3260, 3, 2},
+ { -3225, 25, 5}, { -3191, 13, 4}, { -3158, 27, 5}, { -3126, 7, 3},
+ { -3096, 29, 5}, { -3066, 15, 4}, { -3038, 31, 5}, { -3010, 1, 0},
+ { -2958, 17, 4}, { -2908, 9, 3}, { -2861, 19, 4}, { -2816, 5, 2},
+ { -2774, 21, 4}, { -2734, 11, 3}, { -2695, 23, 4}, { -2658, 3, 1},
+ { -2623, 25, 4}, { -2589, 13, 3}, { -2556, 27, 4}, { -2524, 7, 2},
+ { -2494, 29, 4}, { -2464, 15, 3}, { -2436, 31, 4}, { -2408, 2, 0},
+ { -2356, 17, 3}, { -2306, 9, 2}, { -2259, 19, 3}, { -2214, 5, 1},
+ { -2172, 21, 3}, { -2132, 11, 2}, { -2093, 23, 3}, { -2056, 3, 0},
+ { -2021, 25, 3}, { -1987, 13, 2}, { -1954, 27, 3}, { -1922, 7, 1},
+ { -1892, 29, 3}, { -1862, 15, 2}, { -1834, 31, 3}, { -1806, 4, 0},
+ { -1754, 17, 2}, { -1704, 9, 1}, { -1657, 19, 2}, { -1612, 5, 0},
+ { -1570, 21, 2}, { -1530, 11, 1}, { -1491, 23, 2}, { -1454, 6, 0},
+ { -1419, 25, 2}, { -1384, 13, 1}, { -1352, 27, 2}, { -1320, 7, 0},
+ { -1290, 29, 2}, { -1260, 15, 1}, { -1232, 31, 2}, { -1204, 8, 0},
+ { -1151, 17, 1}, { -1102, 9, 0}, { -1055, 19, 1}, { -1010, 10, 0},
+ { -968, 21, 1}, { -928, 11, 0}, { -889, 23, 1}, { -852, 12, 0},
+ { -816, 25, 1}, { -782, 13, 0}, { -750, 27, 1}, { -718, 14, 0},
+ { -688, 29, 1}, { -658, 15, 0}, { -630, 31, 1}, { -602, 16, 0},
+ { -549, 17, 0}, { -500, 18, 0}, { -453, 19, 0}, { -408, 20, 0},
+ { -366, 21, 0}, { -325, 22, 0}, { -287, 23, 0}, { -250, 24, 0},
+ { -214, 25, 0}, { -180, 26, 0}, { -148, 27, 0}, { -116, 28, 0},
+ { -86, 29, 0}, { -56, 30, 0}, { -28, 31, 0}, { 0, 0, 0},
+};
+
+static int pm860x_volatile(unsigned int reg)
+{
+ BUG_ON(reg >= REG_CACHE_SIZE);
+
+ switch (reg) {
+ case PM860X_AUDIO_SUPPLIES_2:
+ return 1;
+ }
+
+ return 0;
+}
+
+static unsigned int pm860x_read_reg_cache(struct snd_soc_codec *codec,
+ unsigned int reg)
+{
+ unsigned char *cache = codec->reg_cache;
+
+ BUG_ON(reg >= REG_CACHE_SIZE);
+
+ if (pm860x_volatile(reg))
+ return cache[reg];
+
+ reg += REG_CACHE_BASE;
+
+ return pm860x_reg_read(codec->control_data, reg);
+}
+
+static int pm860x_write_reg_cache(struct snd_soc_codec *codec,
+ unsigned int reg, unsigned int value)
+{
+ unsigned char *cache = codec->reg_cache;
+
+ BUG_ON(reg >= REG_CACHE_SIZE);
+
+ if (!pm860x_volatile(reg))
+ cache[reg] = (unsigned char)value;
+
+ reg += REG_CACHE_BASE;
+
+ return pm860x_reg_write(codec->control_data, reg, value);
+}
+
+static int snd_soc_get_volsw_2r_st(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct soc_mixer_control *mc =
+ (struct soc_mixer_control *)kcontrol->private_value;
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ unsigned int reg = mc->reg;
+ unsigned int reg2 = mc->rreg;
+ int val[2], val2[2], i;
+
+ val[0] = snd_soc_read(codec, reg) & 0x3f;
+ val[1] = (snd_soc_read(codec, PM860X_SIDETONE_SHIFT) >> 4) & 0xf;
+ val2[0] = snd_soc_read(codec, reg2) & 0x3f;
+ val2[1] = (snd_soc_read(codec, PM860X_SIDETONE_SHIFT)) & 0xf;
+
+ for (i = 0; i < ARRAY_SIZE(st_table); i++) {
+ if ((st_table[i].m == val[0]) && (st_table[i].n == val[1]))
+ ucontrol->value.integer.value[0] = i;
+ if ((st_table[i].m == val2[0]) && (st_table[i].n == val2[1]))
+ ucontrol->value.integer.value[1] = i;
+ }
+ return 0;
+}
+
+static int snd_soc_put_volsw_2r_st(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct soc_mixer_control *mc =
+ (struct soc_mixer_control *)kcontrol->private_value;
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ unsigned int reg = mc->reg;
+ unsigned int reg2 = mc->rreg;
+ int err;
+ unsigned int val, val2;
+
+ val = ucontrol->value.integer.value[0];
+ val2 = ucontrol->value.integer.value[1];
+
+ err = snd_soc_update_bits(codec, reg, 0x3f, st_table[val].m);
+ if (err < 0)
+ return err;
+ err = snd_soc_update_bits(codec, PM860X_SIDETONE_SHIFT, 0xf0,
+ st_table[val].n << 4);
+ if (err < 0)
+ return err;
+
+ err = snd_soc_update_bits(codec, reg2, 0x3f, st_table[val2].m);
+ if (err < 0)
+ return err;
+ err = snd_soc_update_bits(codec, PM860X_SIDETONE_SHIFT, 0x0f,
+ st_table[val2].n);
+ return err;
+}
+
+static int snd_soc_get_volsw_2r_out(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct soc_mixer_control *mc =
+ (struct soc_mixer_control *)kcontrol->private_value;
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ unsigned int reg = mc->reg;
+ unsigned int reg2 = mc->rreg;
+ unsigned int shift = mc->shift;
+ int max = mc->max, val, val2;
+ unsigned int mask = (1 << fls(max)) - 1;
+
+ val = snd_soc_read(codec, reg) >> shift;
+ val2 = snd_soc_read(codec, reg2) >> shift;
+ ucontrol->value.integer.value[0] = (max - val) & mask;
+ ucontrol->value.integer.value[1] = (max - val2) & mask;
+
+ return 0;
+}
+
+static int snd_soc_put_volsw_2r_out(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct soc_mixer_control *mc =
+ (struct soc_mixer_control *)kcontrol->private_value;
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ unsigned int reg = mc->reg;
+ unsigned int reg2 = mc->rreg;
+ unsigned int shift = mc->shift;
+ int max = mc->max;
+ unsigned int mask = (1 << fls(max)) - 1;
+ int err;
+ unsigned int val, val2, val_mask;
+
+ val_mask = mask << shift;
+ val = ((max - ucontrol->value.integer.value[0]) & mask);
+ val2 = ((max - ucontrol->value.integer.value[1]) & mask);
+
+ val = val << shift;
+ val2 = val2 << shift;
+
+ err = snd_soc_update_bits(codec, reg, val_mask, val);
+ if (err < 0)
+ return err;
+
+ err = snd_soc_update_bits(codec, reg2, val_mask, val2);
+ return err;
+}
+
+/* DAPM Widget Events */
+/*
+ * A lot registers are belong to RSYNC domain. It requires enabling RSYNC bit
+ * after updating these registers. Otherwise, these updated registers won't
+ * be effective.
+ */
+static int pm860x_rsync_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct snd_soc_codec *codec = w->codec;
+
+ /*
+ * In order to avoid current on the load, mute power-on and power-off
+ * should be transients.
+ * Unmute by DAC_MUTE. It should be unmuted when DAPM sequence is
+ * finished.
+ */
+ snd_soc_update_bits(codec, PM860X_DAC_OFFSET, DAC_MUTE, 0);
+ snd_soc_update_bits(codec, PM860X_EAR_CTRL_2,
+ RSYNC_CHANGE, RSYNC_CHANGE);
+ return 0;
+}
+
+static int pm860x_dac_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct snd_soc_codec *codec = w->codec;
+ unsigned int dac = 0;
+ int data;
+
+ if (!strcmp(w->name, "Left DAC"))
+ dac = DAC_LEFT;
+ if (!strcmp(w->name, "Right DAC"))
+ dac = DAC_RIGHT;
+ switch (event) {
+ case SND_SOC_DAPM_PRE_PMU:
+ if (dac) {
+ /* Auto mute in power-on sequence. */
+ dac |= MODULATOR;
+ snd_soc_update_bits(codec, PM860X_DAC_OFFSET,
+ DAC_MUTE, DAC_MUTE);
+ snd_soc_update_bits(codec, PM860X_EAR_CTRL_2,
+ RSYNC_CHANGE, RSYNC_CHANGE);
+ /* update dac */
+ snd_soc_update_bits(codec, PM860X_DAC_EN_2,
+ dac, dac);
+ }
+ break;
+ case SND_SOC_DAPM_PRE_PMD:
+ if (dac) {
+ /* Auto mute in power-off sequence. */
+ snd_soc_update_bits(codec, PM860X_DAC_OFFSET,
+ DAC_MUTE, DAC_MUTE);
+ snd_soc_update_bits(codec, PM860X_EAR_CTRL_2,
+ RSYNC_CHANGE, RSYNC_CHANGE);
+ /* update dac */
+ data = snd_soc_read(codec, PM860X_DAC_EN_2);
+ data &= ~dac;
+ if (!(data & (DAC_LEFT | DAC_RIGHT)))
+ data &= ~MODULATOR;
+ snd_soc_write(codec, PM860X_DAC_EN_2, data);
+ }
+ break;
+ }
+ return 0;
+}
+
+static const char *pm860x_opamp_texts[] = {"-50%", "-25%", "0%", "75%"};
+
+static const char *pm860x_pa_texts[] = {"-33%", "0%", "33%", "66%"};
+
+static const struct soc_enum pm860x_hs1_opamp_enum =
+ SOC_ENUM_SINGLE(PM860X_HS1_CTRL, 5, 4, pm860x_opamp_texts);
+
+static const struct soc_enum pm860x_hs2_opamp_enum =
+ SOC_ENUM_SINGLE(PM860X_HS2_CTRL, 5, 4, pm860x_opamp_texts);
+
+static const struct soc_enum pm860x_hs1_pa_enum =
+ SOC_ENUM_SINGLE(PM860X_HS1_CTRL, 3, 4, pm860x_pa_texts);
+
+static const struct soc_enum pm860x_hs2_pa_enum =
+ SOC_ENUM_SINGLE(PM860X_HS2_CTRL, 3, 4, pm860x_pa_texts);
+
+static const struct soc_enum pm860x_lo1_opamp_enum =
+ SOC_ENUM_SINGLE(PM860X_LO1_CTRL, 5, 4, pm860x_opamp_texts);
+
+static const struct soc_enum pm860x_lo2_opamp_enum =
+ SOC_ENUM_SINGLE(PM860X_LO2_CTRL, 5, 4, pm860x_opamp_texts);
+
+static const struct soc_enum pm860x_lo1_pa_enum =
+ SOC_ENUM_SINGLE(PM860X_LO1_CTRL, 3, 4, pm860x_pa_texts);
+
+static const struct soc_enum pm860x_lo2_pa_enum =
+ SOC_ENUM_SINGLE(PM860X_LO2_CTRL, 3, 4, pm860x_pa_texts);
+
+static const struct soc_enum pm860x_spk_pa_enum =
+ SOC_ENUM_SINGLE(PM860X_EAR_CTRL_1, 5, 4, pm860x_pa_texts);
+
+static const struct soc_enum pm860x_ear_pa_enum =
+ SOC_ENUM_SINGLE(PM860X_EAR_CTRL_2, 0, 4, pm860x_pa_texts);
+
+static const struct soc_enum pm860x_spk_ear_opamp_enum =
+ SOC_ENUM_SINGLE(PM860X_EAR_CTRL_1, 3, 4, pm860x_opamp_texts);
+
+static const struct snd_kcontrol_new pm860x_snd_controls[] = {
+ SOC_DOUBLE_R_TLV("ADC Capture Volume", PM860X_ADC_ANA_2,
+ PM860X_ADC_ANA_3, 6, 3, 0, adc_tlv),
+ SOC_DOUBLE_TLV("AUX Capture Volume", PM860X_ADC_ANA_3, 0, 3, 7, 0,
+ aux_tlv),
+ SOC_SINGLE_TLV("MIC1 Capture Volume", PM860X_ADC_ANA_2, 0, 7, 0,
+ mic_tlv),
+ SOC_SINGLE_TLV("MIC3 Capture Volume", PM860X_ADC_ANA_2, 3, 7, 0,
+ mic_tlv),
+ SOC_DOUBLE_R_EXT_TLV("Sidetone Volume", PM860X_SIDETONE_L_GAIN,
+ PM860X_SIDETONE_R_GAIN, 0, ARRAY_SIZE(st_table)-1,
+ 0, snd_soc_get_volsw_2r_st,
+ snd_soc_put_volsw_2r_st, st_tlv),
+ SOC_SINGLE_TLV("Speaker Playback Volume", PM860X_EAR_CTRL_1,
+ 0, 7, 0, out_tlv),
+ SOC_DOUBLE_R_TLV("Line Playback Volume", PM860X_LO1_CTRL,
+ PM860X_LO2_CTRL, 0, 7, 0, out_tlv),
+ SOC_DOUBLE_R_TLV("Headset Playback Volume", PM860X_HS1_CTRL,
+ PM860X_HS2_CTRL, 0, 7, 0, out_tlv),
+ SOC_DOUBLE_R_EXT_TLV("Hifi Left Playback Volume",
+ PM860X_HIFIL_GAIN_LEFT,
+ PM860X_HIFIL_GAIN_RIGHT, 0, 63, 0,
+ snd_soc_get_volsw_2r_out,
+ snd_soc_put_volsw_2r_out, dpga_tlv),
+ SOC_DOUBLE_R_EXT_TLV("Hifi Right Playback Volume",
+ PM860X_HIFIR_GAIN_LEFT,
+ PM860X_HIFIR_GAIN_RIGHT, 0, 63, 0,
+ snd_soc_get_volsw_2r_out,
+ snd_soc_put_volsw_2r_out, dpga_tlv),
+ SOC_DOUBLE_R_EXT_TLV("Lofi Playback Volume", PM860X_LOFI_GAIN_LEFT,
+ PM860X_LOFI_GAIN_RIGHT, 0, 63, 0,
+ snd_soc_get_volsw_2r_out,
+ snd_soc_put_volsw_2r_out, dpga_tlv),
+ SOC_ENUM("Headset1 Operational Amplifier Current",
+ pm860x_hs1_opamp_enum),
+ SOC_ENUM("Headset2 Operational Amplifier Current",
+ pm860x_hs2_opamp_enum),
+ SOC_ENUM("Headset1 Amplifier Current", pm860x_hs1_pa_enum),
+ SOC_ENUM("Headset2 Amplifier Current", pm860x_hs2_pa_enum),
+ SOC_ENUM("Lineout1 Operational Amplifier Current",
+ pm860x_lo1_opamp_enum),
+ SOC_ENUM("Lineout2 Operational Amplifier Current",
+ pm860x_lo2_opamp_enum),
+ SOC_ENUM("Lineout1 Amplifier Current", pm860x_lo1_pa_enum),
+ SOC_ENUM("Lineout2 Amplifier Current", pm860x_lo2_pa_enum),
+ SOC_ENUM("Speaker Operational Amplifier Current",
+ pm860x_spk_ear_opamp_enum),
+ SOC_ENUM("Speaker Amplifier Current", pm860x_spk_pa_enum),
+ SOC_ENUM("Earpiece Amplifier Current", pm860x_ear_pa_enum),
+};
+
+/*
+ * DAPM Controls
+ */
+
+/* PCM Switch / PCM Interface */
+static const struct snd_kcontrol_new pcm_switch_controls =
+ SOC_DAPM_SINGLE("Switch", PM860X_ADC_EN_2, 0, 1, 0);
+
+/* AUX1 Switch */
+static const struct snd_kcontrol_new aux1_switch_controls =
+ SOC_DAPM_SINGLE("Switch", PM860X_ANA_TO_ANA, 4, 1, 0);
+
+/* AUX2 Switch */
+static const struct snd_kcontrol_new aux2_switch_controls =
+ SOC_DAPM_SINGLE("Switch", PM860X_ANA_TO_ANA, 5, 1, 0);
+
+/* Left Ex. PA Switch */
+static const struct snd_kcontrol_new lepa_switch_controls =
+ SOC_DAPM_SINGLE("Switch", PM860X_DAC_EN_2, 2, 1, 0);
+
+/* Right Ex. PA Switch */
+static const struct snd_kcontrol_new repa_switch_controls =
+ SOC_DAPM_SINGLE("Switch", PM860X_DAC_EN_2, 1, 1, 0);
+
+/* PCM Mux / Mux7 */
+static const char *aif1_text[] = {
+ "PCM L", "PCM R",
+};
+
+static const struct soc_enum aif1_enum =
+ SOC_ENUM_SINGLE(PM860X_PCM_IFACE_3, 6, 2, aif1_text);
+
+static const struct snd_kcontrol_new aif1_mux =
+ SOC_DAPM_ENUM("PCM Mux", aif1_enum);
+
+/* I2S Mux / Mux9 */
+static const char *i2s_din_text[] = {
+ "DIN", "DIN1",
+};
+
+static const struct soc_enum i2s_din_enum =
+ SOC_ENUM_SINGLE(PM860X_I2S_IFACE_3, 1, 2, i2s_din_text);
+
+static const struct snd_kcontrol_new i2s_din_mux =
+ SOC_DAPM_ENUM("I2S DIN Mux", i2s_din_enum);
+
+/* I2S Mic Mux / Mux8 */
+static const char *i2s_mic_text[] = {
+ "Ex PA", "ADC",
+};
+
+static const struct soc_enum i2s_mic_enum =
+ SOC_ENUM_SINGLE(PM860X_I2S_IFACE_3, 4, 2, i2s_mic_text);
+
+static const struct snd_kcontrol_new i2s_mic_mux =
+ SOC_DAPM_ENUM("I2S Mic Mux", i2s_mic_enum);
+
+/* ADCL Mux / Mux2 */
+static const char *adcl_text[] = {
+ "ADCR", "ADCL",
+};
+
+static const struct soc_enum adcl_enum =
+ SOC_ENUM_SINGLE(PM860X_PCM_IFACE_3, 4, 2, adcl_text);
+
+static const struct snd_kcontrol_new adcl_mux =
+ SOC_DAPM_ENUM("ADC Left Mux", adcl_enum);
+
+/* ADCR Mux / Mux3 */
+static const char *adcr_text[] = {
+ "ADCL", "ADCR",
+};
+
+static const struct soc_enum adcr_enum =
+ SOC_ENUM_SINGLE(PM860X_PCM_IFACE_3, 2, 2, adcr_text);
+
+static const struct snd_kcontrol_new adcr_mux =
+ SOC_DAPM_ENUM("ADC Right Mux", adcr_enum);
+
+/* ADCR EC Mux / Mux6 */
+static const char *adcr_ec_text[] = {
+ "ADCR", "EC",
+};
+
+static const struct soc_enum adcr_ec_enum =
+ SOC_ENUM_SINGLE(PM860X_ADC_EN_2, 3, 2, adcr_ec_text);
+
+static const struct snd_kcontrol_new adcr_ec_mux =
+ SOC_DAPM_ENUM("ADCR EC Mux", adcr_ec_enum);
+
+/* EC Mux / Mux4 */
+static const char *ec_text[] = {
+ "Left", "Right", "Left + Right",
+};
+
+static const struct soc_enum ec_enum =
+ SOC_ENUM_SINGLE(PM860X_EC_PATH, 1, 3, ec_text);
+
+static const struct snd_kcontrol_new ec_mux =
+ SOC_DAPM_ENUM("EC Mux", ec_enum);
+
+static const char *dac_text[] = {
+ "No input", "Right", "Left", "No input",
+};
+
+/* DAC Headset 1 Mux / Mux10 */
+static const struct soc_enum dac_hs1_enum =
+ SOC_ENUM_SINGLE(PM860X_ANA_INPUT_SEL_1, 0, 4, dac_text);
+
+static const struct snd_kcontrol_new dac_hs1_mux =
+ SOC_DAPM_ENUM("DAC HS1 Mux", dac_hs1_enum);
+
+/* DAC Headset 2 Mux / Mux11 */
+static const struct soc_enum dac_hs2_enum =
+ SOC_ENUM_SINGLE(PM860X_ANA_INPUT_SEL_1, 2, 4, dac_text);
+
+static const struct snd_kcontrol_new dac_hs2_mux =
+ SOC_DAPM_ENUM("DAC HS2 Mux", dac_hs2_enum);
+
+/* DAC Lineout 1 Mux / Mux12 */
+static const struct soc_enum dac_lo1_enum =
+ SOC_ENUM_SINGLE(PM860X_ANA_INPUT_SEL_1, 4, 4, dac_text);
+
+static const struct snd_kcontrol_new dac_lo1_mux =
+ SOC_DAPM_ENUM("DAC LO1 Mux", dac_lo1_enum);
+
+/* DAC Lineout 2 Mux / Mux13 */
+static const struct soc_enum dac_lo2_enum =
+ SOC_ENUM_SINGLE(PM860X_ANA_INPUT_SEL_1, 6, 4, dac_text);
+
+static const struct snd_kcontrol_new dac_lo2_mux =
+ SOC_DAPM_ENUM("DAC LO2 Mux", dac_lo2_enum);
+
+/* DAC Spearker Earphone Mux / Mux14 */
+static const struct soc_enum dac_spk_ear_enum =
+ SOC_ENUM_SINGLE(PM860X_ANA_INPUT_SEL_2, 0, 4, dac_text);
+
+static const struct snd_kcontrol_new dac_spk_ear_mux =
+ SOC_DAPM_ENUM("DAC SP Mux", dac_spk_ear_enum);
+
+/* Headset 1 Mux / Mux15 */
+static const char *in_text[] = {
+ "Digital", "Analog",
+};
+
+static const struct soc_enum hs1_enum =
+ SOC_ENUM_SINGLE(PM860X_ANA_TO_ANA, 0, 2, in_text);
+
+static const struct snd_kcontrol_new hs1_mux =
+ SOC_DAPM_ENUM("Headset1 Mux", hs1_enum);
+
+/* Headset 2 Mux / Mux16 */
+static const struct soc_enum hs2_enum =
+ SOC_ENUM_SINGLE(PM860X_ANA_TO_ANA, 1, 2, in_text);
+
+static const struct snd_kcontrol_new hs2_mux =
+ SOC_DAPM_ENUM("Headset2 Mux", hs2_enum);
+
+/* Lineout 1 Mux / Mux17 */
+static const struct soc_enum lo1_enum =
+ SOC_ENUM_SINGLE(PM860X_ANA_TO_ANA, 2, 2, in_text);
+
+static const struct snd_kcontrol_new lo1_mux =
+ SOC_DAPM_ENUM("Lineout1 Mux", lo1_enum);
+
+/* Lineout 2 Mux / Mux18 */
+static const struct soc_enum lo2_enum =
+ SOC_ENUM_SINGLE(PM860X_ANA_TO_ANA, 3, 2, in_text);
+
+static const struct snd_kcontrol_new lo2_mux =
+ SOC_DAPM_ENUM("Lineout2 Mux", lo2_enum);
+
+/* Speaker Earpiece Demux */
+static const char *spk_text[] = {
+ "Earpiece", "Speaker",
+};
+
+static const struct soc_enum spk_enum =
+ SOC_ENUM_SINGLE(PM860X_ANA_TO_ANA, 6, 2, spk_text);
+
+static const struct snd_kcontrol_new spk_demux =
+ SOC_DAPM_ENUM("Speaker Earpiece Demux", spk_enum);
+
+/* MIC Mux / Mux1 */
+static const char *mic_text[] = {
+ "Mic 1", "Mic 2",
+};
+
+static const struct soc_enum mic_enum =
+ SOC_ENUM_SINGLE(PM860X_ADC_ANA_4, 4, 2, mic_text);
+
+static const struct snd_kcontrol_new mic_mux =
+ SOC_DAPM_ENUM("MIC Mux", mic_enum);
+
+static const struct snd_soc_dapm_widget pm860x_dapm_widgets[] = {
+ SND_SOC_DAPM_AIF_IN("PCM SDI", "PCM Playback", 0,
+ PM860X_ADC_EN_2, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("PCM SDO", "PCM Capture", 0,
+ PM860X_PCM_IFACE_3, 1, 1),
+
+
+ SND_SOC_DAPM_AIF_IN("I2S DIN", "I2S Playback", 0,
+ PM860X_DAC_EN_2, 0, 0),
+ SND_SOC_DAPM_AIF_IN("I2S DIN1", "I2S Playback", 0,
+ PM860X_DAC_EN_2, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("I2S DOUT", "I2S Capture", 0,
+ PM860X_I2S_IFACE_3, 5, 1),
+ SND_SOC_DAPM_MUX("I2S Mic Mux", SND_SOC_NOPM, 0, 0, &i2s_mic_mux),
+ SND_SOC_DAPM_MUX("ADC Left Mux", SND_SOC_NOPM, 0, 0, &adcl_mux),
+ SND_SOC_DAPM_MUX("ADC Right Mux", SND_SOC_NOPM, 0, 0, &adcr_mux),
+ SND_SOC_DAPM_MUX("EC Mux", SND_SOC_NOPM, 0, 0, &ec_mux),
+ SND_SOC_DAPM_MUX("ADCR EC Mux", SND_SOC_NOPM, 0, 0, &adcr_ec_mux),
+ SND_SOC_DAPM_SWITCH("Left EPA", SND_SOC_NOPM, 0, 0,
+ &lepa_switch_controls),
+ SND_SOC_DAPM_SWITCH("Right EPA", SND_SOC_NOPM, 0, 0,
+ &repa_switch_controls),
+
+ SND_SOC_DAPM_REG(snd_soc_dapm_supply, "Left ADC MOD", PM860X_ADC_EN_1,
+ 0, 1, 1, 0),
+ SND_SOC_DAPM_REG(snd_soc_dapm_supply, "Right ADC MOD", PM860X_ADC_EN_1,
+ 1, 1, 1, 0),
+ SND_SOC_DAPM_ADC("Left ADC", NULL, PM860X_ADC_EN_2, 5, 0),
+ SND_SOC_DAPM_ADC("Right ADC", NULL, PM860X_ADC_EN_2, 4, 0),
+
+ SND_SOC_DAPM_SWITCH("AUX1 Switch", SND_SOC_NOPM, 0, 0,
+ &aux1_switch_controls),
+ SND_SOC_DAPM_SWITCH("AUX2 Switch", SND_SOC_NOPM, 0, 0,
+ &aux2_switch_controls),
+
+ SND_SOC_DAPM_MUX("MIC Mux", SND_SOC_NOPM, 0, 0, &mic_mux),
+ SND_SOC_DAPM_MICBIAS("Mic1 Bias", PM860X_ADC_ANA_1, 2, 0),
+ SND_SOC_DAPM_MICBIAS("Mic3 Bias", PM860X_ADC_ANA_1, 7, 0),
+ SND_SOC_DAPM_PGA("MIC1 Volume", PM860X_ADC_EN_1, 2, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("MIC3 Volume", PM860X_ADC_EN_1, 3, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("AUX1 Volume", PM860X_ADC_EN_1, 4, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("AUX2 Volume", PM860X_ADC_EN_1, 5, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("Sidetone PGA", PM860X_ADC_EN_2, 1, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("Lofi PGA", PM860X_ADC_EN_2, 2, 0, NULL, 0),
+
+ SND_SOC_DAPM_INPUT("AUX1"),
+ SND_SOC_DAPM_INPUT("AUX2"),
+ SND_SOC_DAPM_INPUT("MIC1P"),
+ SND_SOC_DAPM_INPUT("MIC1N"),
+ SND_SOC_DAPM_INPUT("MIC2P"),
+ SND_SOC_DAPM_INPUT("MIC2N"),
+ SND_SOC_DAPM_INPUT("MIC3P"),
+ SND_SOC_DAPM_INPUT("MIC3N"),
+
+ SND_SOC_DAPM_DAC_E("Left DAC", NULL, SND_SOC_NOPM, 0, 0,
+ pm860x_dac_event,
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_PRE_PMD),
+ SND_SOC_DAPM_DAC_E("Right DAC", NULL, SND_SOC_NOPM, 0, 0,
+ pm860x_dac_event,
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_PRE_PMD),
+
+ SND_SOC_DAPM_MUX("I2S DIN Mux", SND_SOC_NOPM, 0, 0, &i2s_din_mux),
+ SND_SOC_DAPM_MUX("DAC HS1 Mux", SND_SOC_NOPM, 0, 0, &dac_hs1_mux),
+ SND_SOC_DAPM_MUX("DAC HS2 Mux", SND_SOC_NOPM, 0, 0, &dac_hs2_mux),
+ SND_SOC_DAPM_MUX("DAC LO1 Mux", SND_SOC_NOPM, 0, 0, &dac_lo1_mux),
+ SND_SOC_DAPM_MUX("DAC LO2 Mux", SND_SOC_NOPM, 0, 0, &dac_lo2_mux),
+ SND_SOC_DAPM_MUX("DAC SP Mux", SND_SOC_NOPM, 0, 0, &dac_spk_ear_mux),
+ SND_SOC_DAPM_MUX("Headset1 Mux", SND_SOC_NOPM, 0, 0, &hs1_mux),
+ SND_SOC_DAPM_MUX("Headset2 Mux", SND_SOC_NOPM, 0, 0, &hs2_mux),
+ SND_SOC_DAPM_MUX("Lineout1 Mux", SND_SOC_NOPM, 0, 0, &lo1_mux),
+ SND_SOC_DAPM_MUX("Lineout2 Mux", SND_SOC_NOPM, 0, 0, &lo2_mux),
+ SND_SOC_DAPM_MUX("Speaker Earpiece Demux", SND_SOC_NOPM, 0, 0,
+ &spk_demux),
+
+
+ SND_SOC_DAPM_PGA("Headset1 PGA", PM860X_DAC_EN_1, 0, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("Headset2 PGA", PM860X_DAC_EN_1, 1, 0, NULL, 0),
+ SND_SOC_DAPM_OUTPUT("HS1"),
+ SND_SOC_DAPM_OUTPUT("HS2"),
+ SND_SOC_DAPM_PGA("Lineout1 PGA", PM860X_DAC_EN_1, 2, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("Lineout2 PGA", PM860X_DAC_EN_1, 3, 0, NULL, 0),
+ SND_SOC_DAPM_OUTPUT("LINEOUT1"),
+ SND_SOC_DAPM_OUTPUT("LINEOUT2"),
+ SND_SOC_DAPM_PGA("Earpiece PGA", PM860X_DAC_EN_1, 4, 0, NULL, 0),
+ SND_SOC_DAPM_OUTPUT("EARP"),
+ SND_SOC_DAPM_OUTPUT("EARN"),
+ SND_SOC_DAPM_PGA("Speaker PGA", PM860X_DAC_EN_1, 5, 0, NULL, 0),
+ SND_SOC_DAPM_OUTPUT("LSP"),
+ SND_SOC_DAPM_OUTPUT("LSN"),
+ SND_SOC_DAPM_REG(snd_soc_dapm_supply, "VCODEC", PM860X_AUDIO_SUPPLIES_2,
+ 0, SUPPLY_MASK, SUPPLY_MASK, 0),
+
+ PM860X_DAPM_OUTPUT("RSYNC", pm860x_rsync_event),
+};
+
+static const struct snd_soc_dapm_route audio_map[] = {
+ /* supply */
+ {"Left DAC", NULL, "VCODEC"},
+ {"Right DAC", NULL, "VCODEC"},
+ {"Left ADC", NULL, "VCODEC"},
+ {"Right ADC", NULL, "VCODEC"},
+ {"Left ADC", NULL, "Left ADC MOD"},
+ {"Right ADC", NULL, "Right ADC MOD"},
+
+ /* PCM/AIF1 Inputs */
+ {"PCM SDO", NULL, "ADC Left Mux"},
+ {"PCM SDO", NULL, "ADCR EC Mux"},
+
+ /* PCM/AFI2 Outputs */
+ {"Lofi PGA", NULL, "PCM SDI"},
+ {"Lofi PGA", NULL, "Sidetone PGA"},
+ {"Left DAC", NULL, "Lofi PGA"},
+ {"Right DAC", NULL, "Lofi PGA"},
+
+ /* I2S/AIF2 Inputs */
+ {"MIC Mux", "Mic 1", "MIC1P"},
+ {"MIC Mux", "Mic 1", "MIC1N"},
+ {"MIC Mux", "Mic 2", "MIC2P"},
+ {"MIC Mux", "Mic 2", "MIC2N"},
+ {"MIC1 Volume", NULL, "MIC Mux"},
+ {"MIC3 Volume", NULL, "MIC3P"},
+ {"MIC3 Volume", NULL, "MIC3N"},
+ {"Left ADC", NULL, "MIC1 Volume"},
+ {"Right ADC", NULL, "MIC3 Volume"},
+ {"ADC Left Mux", "ADCR", "Right ADC"},
+ {"ADC Left Mux", "ADCL", "Left ADC"},
+ {"ADC Right Mux", "ADCL", "Left ADC"},
+ {"ADC Right Mux", "ADCR", "Right ADC"},
+ {"Left EPA", "Switch", "Left DAC"},
+ {"Right EPA", "Switch", "Right DAC"},
+ {"EC Mux", "Left", "Left DAC"},
+ {"EC Mux", "Right", "Right DAC"},
+ {"EC Mux", "Left + Right", "Left DAC"},
+ {"EC Mux", "Left + Right", "Right DAC"},
+ {"ADCR EC Mux", "ADCR", "ADC Right Mux"},
+ {"ADCR EC Mux", "EC", "EC Mux"},
+ {"I2S Mic Mux", "Ex PA", "Left EPA"},
+ {"I2S Mic Mux", "Ex PA", "Right EPA"},
+ {"I2S Mic Mux", "ADC", "ADC Left Mux"},
+ {"I2S Mic Mux", "ADC", "ADCR EC Mux"},
+ {"I2S DOUT", NULL, "I2S Mic Mux"},
+
+ /* I2S/AIF2 Outputs */
+ {"I2S DIN Mux", "DIN", "I2S DIN"},
+ {"I2S DIN Mux", "DIN1", "I2S DIN1"},
+ {"Left DAC", NULL, "I2S DIN Mux"},
+ {"Right DAC", NULL, "I2S DIN Mux"},
+ {"DAC HS1 Mux", "Left", "Left DAC"},
+ {"DAC HS1 Mux", "Right", "Right DAC"},
+ {"DAC HS2 Mux", "Left", "Left DAC"},
+ {"DAC HS2 Mux", "Right", "Right DAC"},
+ {"DAC LO1 Mux", "Left", "Left DAC"},
+ {"DAC LO1 Mux", "Right", "Right DAC"},
+ {"DAC LO2 Mux", "Left", "Left DAC"},
+ {"DAC LO2 Mux", "Right", "Right DAC"},
+ {"Headset1 Mux", "Digital", "DAC HS1 Mux"},
+ {"Headset2 Mux", "Digital", "DAC HS2 Mux"},
+ {"Lineout1 Mux", "Digital", "DAC LO1 Mux"},
+ {"Lineout2 Mux", "Digital", "DAC LO2 Mux"},
+ {"Headset1 PGA", NULL, "Headset1 Mux"},
+ {"Headset2 PGA", NULL, "Headset2 Mux"},
+ {"Lineout1 PGA", NULL, "Lineout1 Mux"},
+ {"Lineout2 PGA", NULL, "Lineout2 Mux"},
+ {"DAC SP Mux", "Left", "Left DAC"},
+ {"DAC SP Mux", "Right", "Right DAC"},
+ {"Speaker Earpiece Demux", "Speaker", "DAC SP Mux"},
+ {"Speaker PGA", NULL, "Speaker Earpiece Demux"},
+ {"Earpiece PGA", NULL, "Speaker Earpiece Demux"},
+
+ {"RSYNC", NULL, "Headset1 PGA"},
+ {"RSYNC", NULL, "Headset2 PGA"},
+ {"RSYNC", NULL, "Lineout1 PGA"},
+ {"RSYNC", NULL, "Lineout2 PGA"},
+ {"RSYNC", NULL, "Speaker PGA"},
+ {"RSYNC", NULL, "Speaker PGA"},
+ {"RSYNC", NULL, "Earpiece PGA"},
+ {"RSYNC", NULL, "Earpiece PGA"},
+
+ {"HS1", NULL, "RSYNC"},
+ {"HS2", NULL, "RSYNC"},
+ {"LINEOUT1", NULL, "RSYNC"},
+ {"LINEOUT2", NULL, "RSYNC"},
+ {"LSP", NULL, "RSYNC"},
+ {"LSN", NULL, "RSYNC"},
+ {"EARP", NULL, "RSYNC"},
+ {"EARN", NULL, "RSYNC"},
+};
+
+/*
+ * Use MUTE_LEFT & MUTE_RIGHT to implement digital mute.
+ * These bits can also be used to mute.
+ */
+static int pm860x_digital_mute(struct snd_soc_dai *codec_dai, int mute)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ int data = 0, mask = MUTE_LEFT | MUTE_RIGHT;
+
+ if (mute)
+ data = mask;
+ snd_soc_update_bits(codec, PM860X_DAC_OFFSET, mask, data);
+ snd_soc_update_bits(codec, PM860X_EAR_CTRL_2,
+ RSYNC_CHANGE, RSYNC_CHANGE);
+ return 0;
+}
+
+static int pm860x_pcm_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;
+ unsigned char inf = 0, mask = 0;
+
+ /* bit size */
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+ inf &= ~PCM_INF2_18WL;
+ break;
+ case SNDRV_PCM_FORMAT_S18_3LE:
+ inf |= PCM_INF2_18WL;
+ break;
+ default:
+ return -EINVAL;
+ }
+ mask |= PCM_INF2_18WL;
+ snd_soc_update_bits(codec, PM860X_PCM_IFACE_2, mask, inf);
+
+ /* sample rate */
+ switch (params_rate(params)) {
+ case 8000:
+ inf = 0;
+ break;
+ case 16000:
+ inf = 3;
+ break;
+ case 32000:
+ inf = 6;
+ break;
+ case 48000:
+ inf = 8;
+ break;
+ default:
+ return -EINVAL;
+ }
+ snd_soc_update_bits(codec, PM860X_PCM_RATE, 0x0f, inf);
+
+ return 0;
+}
+
+static int pm860x_pcm_set_dai_fmt(struct snd_soc_dai *codec_dai,
+ unsigned int fmt)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ struct pm860x_priv *pm860x = snd_soc_codec_get_drvdata(codec);
+ unsigned char inf = 0, mask = 0;
+ int ret = -EINVAL;
+
+ mask |= PCM_INF2_BCLK | PCM_INF2_FS | PCM_INF2_MASTER;
+
+ /* set master/slave audio interface */
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBM_CFM:
+ case SND_SOC_DAIFMT_CBM_CFS:
+ if (pm860x->dir == PM860X_CLK_DIR_OUT) {
+ inf |= PCM_INF2_MASTER;
+ ret = 0;
+ }
+ break;
+ case SND_SOC_DAIFMT_CBS_CFS:
+ if (pm860x->dir == PM860X_CLK_DIR_IN) {
+ inf &= ~PCM_INF2_MASTER;
+ ret = 0;
+ }
+ break;
+ }
+
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ inf |= PCM_EXACT_I2S;
+ ret = 0;
+ break;
+ }
+ mask |= PCM_MODE_MASK;
+ if (ret)
+ return ret;
+ snd_soc_update_bits(codec, PM860X_PCM_IFACE_2, mask, inf);
+ return 0;
+}
+
+static int pm860x_set_dai_sysclk(struct snd_soc_dai *codec_dai,
+ int clk_id, unsigned int freq, int dir)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ struct pm860x_priv *pm860x = snd_soc_codec_get_drvdata(codec);
+
+ if (dir == PM860X_CLK_DIR_OUT)
+ pm860x->dir = PM860X_CLK_DIR_OUT;
+ else {
+ pm860x->dir = PM860X_CLK_DIR_IN;
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static int pm860x_i2s_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;
+ unsigned char inf;
+
+ /* bit size */
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+ inf = 0;
+ break;
+ case SNDRV_PCM_FORMAT_S18_3LE:
+ inf = PCM_INF2_18WL;
+ break;
+ default:
+ return -EINVAL;
+ }
+ snd_soc_update_bits(codec, PM860X_I2S_IFACE_2, PCM_INF2_18WL, inf);
+
+ /* sample rate */
+ switch (params_rate(params)) {
+ case 8000:
+ inf = 0;
+ break;
+ case 11025:
+ inf = 1;
+ break;
+ case 16000:
+ inf = 3;
+ break;
+ case 22050:
+ inf = 4;
+ break;
+ case 32000:
+ inf = 6;
+ break;
+ case 44100:
+ inf = 7;
+ break;
+ case 48000:
+ inf = 8;
+ break;
+ default:
+ return -EINVAL;
+ }
+ snd_soc_update_bits(codec, PM860X_I2S_IFACE_4, 0xf, inf);
+
+ return 0;
+}
+
+static int pm860x_i2s_set_dai_fmt(struct snd_soc_dai *codec_dai,
+ unsigned int fmt)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ struct pm860x_priv *pm860x = snd_soc_codec_get_drvdata(codec);
+ unsigned char inf = 0, mask = 0;
+
+ mask |= PCM_INF2_BCLK | PCM_INF2_FS | PCM_INF2_MASTER;
+
+ /* set master/slave audio interface */
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBM_CFM:
+ if (pm860x->dir == PM860X_CLK_DIR_OUT)
+ inf |= PCM_INF2_MASTER;
+ else
+ return -EINVAL;
+ break;
+ case SND_SOC_DAIFMT_CBS_CFS:
+ if (pm860x->dir == PM860X_CLK_DIR_IN)
+ inf &= ~PCM_INF2_MASTER;
+ else
+ return -EINVAL;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ inf |= PCM_EXACT_I2S;
+ break;
+ default:
+ return -EINVAL;
+ }
+ mask |= PCM_MODE_MASK;
+ snd_soc_update_bits(codec, PM860X_I2S_IFACE_2, mask, inf);
+ return 0;
+}
+
+static int pm860x_set_bias_level(struct snd_soc_codec *codec,
+ enum snd_soc_bias_level level)
+{
+ int data;
+
+ switch (level) {
+ case SND_SOC_BIAS_ON:
+ break;
+
+ case SND_SOC_BIAS_PREPARE:
+ break;
+
+ case SND_SOC_BIAS_STANDBY:
+ if (codec->bias_level == SND_SOC_BIAS_OFF) {
+ /* Enable Audio PLL & Audio section */
+ data = AUDIO_PLL | AUDIO_SECTION_RESET
+ | AUDIO_SECTION_ON;
+ pm860x_reg_write(codec->control_data, REG_MISC2, data);
+ }
+ break;
+
+ case SND_SOC_BIAS_OFF:
+ data = AUDIO_PLL | AUDIO_SECTION_RESET | AUDIO_SECTION_ON;
+ pm860x_set_bits(codec->control_data, REG_MISC2, data, 0);
+ break;
+ }
+ codec->bias_level = level;
+ return 0;
+}
+
+static struct snd_soc_dai_ops pm860x_pcm_dai_ops = {
+ .digital_mute = pm860x_digital_mute,
+ .hw_params = pm860x_pcm_hw_params,
+ .set_fmt = pm860x_pcm_set_dai_fmt,
+ .set_sysclk = pm860x_set_dai_sysclk,
+};
+
+static struct snd_soc_dai_ops pm860x_i2s_dai_ops = {
+ .digital_mute = pm860x_digital_mute,
+ .hw_params = pm860x_i2s_hw_params,
+ .set_fmt = pm860x_i2s_set_dai_fmt,
+ .set_sysclk = pm860x_set_dai_sysclk,
+};
+
+#define PM860X_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 | \
+ SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_48000)
+
+static struct snd_soc_dai_driver pm860x_dai[] = {
+ {
+ /* DAI PCM */
+ .name = "88pm860x-pcm",
+ .id = 1,
+ .playback = {
+ .stream_name = "PCM Playback",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = PM860X_RATES,
+ .formats = SNDRV_PCM_FORMAT_S16_LE | \
+ SNDRV_PCM_FORMAT_S18_3LE,
+ },
+ .capture = {
+ .stream_name = "PCM Capture",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = PM860X_RATES,
+ .formats = SNDRV_PCM_FORMAT_S16_LE | \
+ SNDRV_PCM_FORMAT_S18_3LE,
+ },
+ .ops = &pm860x_pcm_dai_ops,
+ }, {
+ /* DAI I2S */
+ .name = "88pm860x-i2s",
+ .id = 2,
+ .playback = {
+ .stream_name = "I2S Playback",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_8000_48000,
+ .formats = SNDRV_PCM_FORMAT_S16_LE | \
+ SNDRV_PCM_FORMAT_S18_3LE,
+ },
+ .capture = {
+ .stream_name = "I2S Capture",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_8000_48000,
+ .formats = SNDRV_PCM_FORMAT_S16_LE | \
+ SNDRV_PCM_FORMAT_S18_3LE,
+ },
+ .ops = &pm860x_i2s_dai_ops,
+ },
+};
+
+static irqreturn_t pm860x_codec_handler(int irq, void *data)
+{
+ struct pm860x_priv *pm860x = data;
+ int status, shrt, report = 0, mic_report = 0;
+ int mask;
+
+ status = pm860x_reg_read(pm860x->i2c, REG_STATUS_1);
+ shrt = pm860x_reg_read(pm860x->i2c, REG_SHORTS);
+ mask = pm860x->det.hs_shrt | pm860x->det.hook_det | pm860x->det.lo_shrt
+ | pm860x->det.hp_det;
+
+ if ((pm860x->det.hp_det & SND_JACK_HEADPHONE)
+ && (status & HEADSET_STATUS))
+ report |= SND_JACK_HEADPHONE;
+
+ if ((pm860x->det.mic_det & SND_JACK_MICROPHONE)
+ && (status & MIC_STATUS))
+ mic_report |= SND_JACK_MICROPHONE;
+
+ if (pm860x->det.hs_shrt && (shrt & (SHORT_HS1 | SHORT_HS2)))
+ report |= pm860x->det.hs_shrt;
+
+ if (pm860x->det.hook_det && (status & HOOK_STATUS))
+ report |= pm860x->det.hook_det;
+
+ if (pm860x->det.lo_shrt && (shrt & (SHORT_LO1 | SHORT_LO2)))
+ report |= pm860x->det.lo_shrt;
+
+ if (report)
+ snd_soc_jack_report(pm860x->det.hp_jack, report, mask);
+ if (mic_report)
+ snd_soc_jack_report(pm860x->det.mic_jack, SND_JACK_MICROPHONE,
+ SND_JACK_MICROPHONE);
+
+ dev_dbg(pm860x->codec->dev, "headphone report:0x%x, mask:%x\n",
+ report, mask);
+ dev_dbg(pm860x->codec->dev, "microphone report:0x%x\n", mic_report);
+ return IRQ_HANDLED;
+}
+
+int pm860x_hs_jack_detect(struct snd_soc_codec *codec,
+ struct snd_soc_jack *jack,
+ int det, int hook, int hs_shrt, int lo_shrt)
+{
+ struct pm860x_priv *pm860x = snd_soc_codec_get_drvdata(codec);
+ int data;
+
+ pm860x->det.hp_jack = jack;
+ pm860x->det.hp_det = det;
+ pm860x->det.hook_det = hook;
+ pm860x->det.hs_shrt = hs_shrt;
+ pm860x->det.lo_shrt = lo_shrt;
+
+ if (det & SND_JACK_HEADPHONE)
+ pm860x_set_bits(codec->control_data, REG_HS_DET,
+ EN_HS_DET, EN_HS_DET);
+ /* headset short detect */
+ if (hs_shrt) {
+ data = CLR_SHORT_HS2 | CLR_SHORT_HS1;
+ pm860x_set_bits(codec->control_data, REG_SHORTS, data, data);
+ }
+ /* Lineout short detect */
+ if (lo_shrt) {
+ data = CLR_SHORT_LO2 | CLR_SHORT_LO1;
+ pm860x_set_bits(codec->control_data, REG_SHORTS, data, data);
+ }
+
+ /* sync status */
+ pm860x_codec_handler(0, pm860x);
+ return 0;
+}
+EXPORT_SYMBOL_GPL(pm860x_hs_jack_detect);
+
+int pm860x_mic_jack_detect(struct snd_soc_codec *codec,
+ struct snd_soc_jack *jack, int det)
+{
+ struct pm860x_priv *pm860x = snd_soc_codec_get_drvdata(codec);
+
+ pm860x->det.mic_jack = jack;
+ pm860x->det.mic_det = det;
+
+ if (det & SND_JACK_MICROPHONE)
+ pm860x_set_bits(codec->control_data, REG_MIC_DET,
+ MICDET_MASK, MICDET_MASK);
+
+ /* sync status */
+ pm860x_codec_handler(0, pm860x);
+ return 0;
+}
+EXPORT_SYMBOL_GPL(pm860x_mic_jack_detect);
+
+static int pm860x_probe(struct snd_soc_codec *codec)
+{
+ struct pm860x_priv *pm860x = snd_soc_codec_get_drvdata(codec);
+ int i, ret;
+
+ pm860x->codec = codec;
+
+ codec->control_data = pm860x->i2c;
+
+ for (i = 0; i < 4; i++) {
+ ret = request_threaded_irq(pm860x->irq[i], NULL,
+ pm860x_codec_handler, IRQF_ONESHOT,
+ pm860x->name[i], pm860x);
+ if (ret < 0) {
+ dev_err(codec->dev, "Failed to request IRQ!\n");
+ goto out_irq;
+ }
+ }
+
+ pm860x_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+ ret = pm860x_bulk_read(codec->control_data, REG_CACHE_BASE,
+ REG_CACHE_SIZE, codec->reg_cache);
+ if (ret < 0) {
+ dev_err(codec->dev, "Failed to fill register cache: %d\n",
+ ret);
+ goto out_codec;
+ }
+
+ snd_soc_add_controls(codec, pm860x_snd_controls,
+ ARRAY_SIZE(pm860x_snd_controls));
+ snd_soc_dapm_new_controls(codec, pm860x_dapm_widgets,
+ ARRAY_SIZE(pm860x_dapm_widgets));
+ snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+ return 0;
+
+out_codec:
+ i = 3;
+out_irq:
+ for (; i >= 0; i--)
+ free_irq(pm860x->irq[i], pm860x);
+ return -EINVAL;
+}
+
+static int pm860x_remove(struct snd_soc_codec *codec)
+{
+ struct pm860x_priv *pm860x = snd_soc_codec_get_drvdata(codec);
+ int i;
+
+ for (i = 3; i >= 0; i--)
+ free_irq(pm860x->irq[i], pm860x);
+ pm860x_set_bias_level(codec, SND_SOC_BIAS_OFF);
+ return 0;
+}
+
+static struct snd_soc_codec_driver soc_codec_dev_pm860x = {
+ .probe = pm860x_probe,
+ .remove = pm860x_remove,
+ .read = pm860x_read_reg_cache,
+ .write = pm860x_write_reg_cache,
+ .reg_cache_size = REG_CACHE_SIZE,
+ .reg_word_size = sizeof(u8),
+ .set_bias_level = pm860x_set_bias_level,
+};
+
+static int __devinit pm860x_codec_probe(struct platform_device *pdev)
+{
+ struct pm860x_chip *chip = dev_get_drvdata(pdev->dev.parent);
+ struct pm860x_priv *pm860x;
+ struct resource *res;
+ int i, ret;
+
+ pm860x = kzalloc(sizeof(struct pm860x_priv), GFP_KERNEL);
+ if (pm860x == NULL)
+ return -ENOMEM;
+
+ pm860x->chip = chip;
+ pm860x->i2c = (chip->id == CHIP_PM8607) ? chip->client
+ : chip->companion;
+ platform_set_drvdata(pdev, pm860x);
+
+ for (i = 0; i < 4; i++) {
+ res = platform_get_resource(pdev, IORESOURCE_IRQ, i);
+ if (!res) {
+ dev_err(&pdev->dev, "Failed to get IRQ resources\n");
+ goto out;
+ }
+ pm860x->irq[i] = res->start + chip->irq_base;
+ strncpy(pm860x->name[i], res->name, MAX_NAME_LEN);
+ }
+
+ ret = snd_soc_register_codec(&pdev->dev, &soc_codec_dev_pm860x,
+ pm860x_dai, ARRAY_SIZE(pm860x_dai));
+ if (ret) {
+ dev_err(&pdev->dev, "Failed to register codec\n");
+ goto out;
+ }
+ return ret;
+
+out:
+ platform_set_drvdata(pdev, NULL);
+ kfree(pm860x);
+ return -EINVAL;
+}
+
+static int __devexit pm860x_codec_remove(struct platform_device *pdev)
+{
+ struct pm860x_priv *pm860x = platform_get_drvdata(pdev);
+
+ snd_soc_unregister_codec(&pdev->dev);
+ platform_set_drvdata(pdev, NULL);
+ kfree(pm860x);
+ return 0;
+}
+
+static struct platform_driver pm860x_codec_driver = {
+ .driver = {
+ .name = "88pm860x-codec",
+ .owner = THIS_MODULE,
+ },
+ .probe = pm860x_codec_probe,
+ .remove = __devexit_p(pm860x_codec_remove),
+};
+
+static __init int pm860x_init(void)
+{
+ return platform_driver_register(&pm860x_codec_driver);
+}
+module_init(pm860x_init);
+
+static __exit void pm860x_exit(void)
+{
+ platform_driver_unregister(&pm860x_codec_driver);
+}
+module_exit(pm860x_exit);
+
+MODULE_DESCRIPTION("ASoC 88PM860x driver");
+MODULE_AUTHOR("Haojian Zhuang <haojian.zhuang at marvell.com>");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:88pm860x-codec");
+
diff --git a/sound/soc/codecs/88pm860x-codec.h
b/sound/soc/codecs/88pm860x-codec.h
new file mode 100644
index 0000000..3364ba4
--- /dev/null
+++ b/sound/soc/codecs/88pm860x-codec.h
@@ -0,0 +1,97 @@
+/*
+ * 88pm860x-codec.h -- 88PM860x ALSA SoC Audio Driver
+ *
+ * Copyright 2010 Marvell International Ltd.
+ * Haojian Zhuang <haojian.zhuang at marvell.com>
+ *
+ * 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.
+ */
+
+#ifndef __88PM860X_H
+#define __88PM860X_H
+
+/* The offset of these registers are 0xb0 */
+#define PM860X_PCM_IFACE_1 0x00
+#define PM860X_PCM_IFACE_2 0x01
+#define PM860X_PCM_IFACE_3 0x02
+#define PM860X_PCM_RATE 0x03
+#define PM860X_EC_PATH 0x04
+#define PM860X_SIDETONE_L_GAIN 0x05
+#define PM860X_SIDETONE_R_GAIN 0x06
+#define PM860X_SIDETONE_SHIFT 0x07
+#define PM860X_ADC_OFFSET_1 0x08
+#define PM860X_ADC_OFFSET_2 0x09
+#define PM860X_DMIC_DELAY 0x0a
+
+#define PM860X_I2S_IFACE_1 0x0b
+#define PM860X_I2S_IFACE_2 0x0c
+#define PM860X_I2S_IFACE_3 0x0d
+#define PM860X_I2S_IFACE_4 0x0e
+#define PM860X_EQUALIZER_N0_1 0x0f
+#define PM860X_EQUALIZER_N0_2 0x10
+#define PM860X_EQUALIZER_N1_1 0x11
+#define PM860X_EQUALIZER_N1_2 0x12
+#define PM860X_EQUALIZER_D1_1 0x13
+#define PM860X_EQUALIZER_D1_2 0x14
+#define PM860X_LOFI_GAIN_LEFT 0x15
+#define PM860X_LOFI_GAIN_RIGHT 0x16
+#define PM860X_HIFIL_GAIN_LEFT 0x17
+#define PM860X_HIFIL_GAIN_RIGHT 0x18
+#define PM860X_HIFIR_GAIN_LEFT 0x19
+#define PM860X_HIFIR_GAIN_RIGHT 0x1a
+#define PM860X_DAC_OFFSET 0x1b
+#define PM860X_OFFSET_LEFT_1 0x1c
+#define PM860X_OFFSET_LEFT_2 0x1d
+#define PM860X_OFFSET_RIGHT_1 0x1e
+#define PM860X_OFFSET_RIGHT_2 0x1f
+#define PM860X_ADC_ANA_1 0x20
+#define PM860X_ADC_ANA_2 0x21
+#define PM860X_ADC_ANA_3 0x22
+#define PM860X_ADC_ANA_4 0x23
+#define PM860X_ANA_TO_ANA 0x24
+#define PM860X_HS1_CTRL 0x25
+#define PM860X_HS2_CTRL 0x26
+#define PM860X_LO1_CTRL 0x27
+#define PM860X_LO2_CTRL 0x28
+#define PM860X_EAR_CTRL_1 0x29
+#define PM860X_EAR_CTRL_2 0x2a
+#define PM860X_AUDIO_SUPPLIES_1 0x2b
+#define PM860X_AUDIO_SUPPLIES_2 0x2c
+#define PM860X_ADC_EN_1 0x2d
+#define PM860X_ADC_EN_2 0x2e
+#define PM860X_DAC_EN_1 0x2f
+#define PM860X_DAC_EN_2 0x31
+#define PM860X_AUDIO_CAL_1 0x32
+#define PM860X_AUDIO_CAL_2 0x33
+#define PM860X_AUDIO_CAL_3 0x34
+#define PM860X_AUDIO_CAL_4 0x35
+#define PM860X_AUDIO_CAL_5 0x36
+#define PM860X_ANA_INPUT_SEL_1 0x37
+#define PM860X_ANA_INPUT_SEL_2 0x38
+
+#define PM860X_PCM_IFACE_4 0x39
+#define PM860X_I2S_IFACE_5 0x3a
+
+#define PM860X_SHORTS 0x3b
+#define PM860X_PLL_ADJ_1 0x3c
+#define PM860X_PLL_ADJ_2 0x3d
+
+/* bits definition */
+#define PM860X_CLK_DIR_IN 0
+#define PM860X_CLK_DIR_OUT 1
+
+#define PM860X_DET_HEADSET (1 << 0)
+#define PM860X_DET_MIC (1 << 1)
+#define PM860X_DET_HOOK (1 << 2)
+#define PM860X_SHORT_HEADSET (1 << 3)
+#define PM860X_SHORT_LINEOUT (1 << 4)
+#define PM860X_DET_MASK 0x1F
+
+extern int pm860x_hs_jack_detect(struct snd_soc_codec *, struct snd_soc_jack *,
+ int, int, int, int);
+extern int pm860x_mic_jack_detect(struct snd_soc_codec *, struct
snd_soc_jack *,
+ int);
+
+#endif /* __88PM860X_H */
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index bfdd92b..a3cfc18 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -10,6 +10,7 @@ config SND_SOC_I2C_AND_SPI
config SND_SOC_ALL_CODECS
tristate "Build all ASoC CODEC drivers"
+ select SND_SOC_88PM860X if MFD_88PM860X
select SND_SOC_L3
select SND_SOC_AC97_CODEC if SND_SOC_AC97_BUS
select SND_SOC_AD1836 if SPI_MASTER
@@ -85,6 +86,9 @@ config SND_SOC_ALL_CODECS
If unsure select "N".
+config SND_SOC_88PM860X
+ tristate
+
config SND_SOC_WM_HUBS
tristate
default y if SND_SOC_WM8993=y || SND_SOC_WM8994=y
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index 9c3c39f..b9c4358 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -1,3 +1,4 @@
+snd-soc-88pm860x-objs := 88pm860x-codec.o
snd-soc-ac97-objs := ac97.o
snd-soc-ad1836-objs := ad1836.o
snd-soc-ad193x-objs := ad193x.o
@@ -67,6 +68,7 @@ snd-soc-tpa6130a2-objs := tpa6130a2.o
snd-soc-wm2000-objs := wm2000.o
snd-soc-wm9090-objs := wm9090.o
+obj-$(CONFIG_SND_SOC_88PM860X) += snd-soc-88pm860x.o
obj-$(CONFIG_SND_SOC_AC97_CODEC) += snd-soc-ac97.o
obj-$(CONFIG_SND_SOC_AD1836) += snd-soc-ad1836.o
obj-$(CONFIG_SND_SOC_AD193X) += snd-soc-ad193x.o
--
1.5.6.5
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