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@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@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@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@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