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[PATCH 1/3] ASoC: add 88pm860x codec driver
by Haojian Zhuang 12 Aug '10

12 Aug '10

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(a)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(a)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(a)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(a)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

1 0

[PATCH 1/3] ASoC: add 88pm860x codec driver
by Haojian Zhuang 12 Aug '10

12 Aug '10

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(a)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(a)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(a)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(a)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

1 0

[PATCH 1/3] ASoC: add 88pm860x codec driver
by Haojian Zhuang 12 Aug '10

12 Aug '10

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(a)marvell.com> --- sound/soc/codecs/88pm860x-codec.c | 1410 +++++++++++++++++++++++++++++++++++++ sound/soc/codecs/88pm860x-codec.h | 92 +++ sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + 4 files changed, 1508 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..1acb1f1 --- /dev/null +++ b/sound/soc/codecs/88pm860x-codec.c @@ -0,0 +1,1410 @@ +/* + * 88pm860x-codec.c -- 88PM860x ALSA SoC Audio Driver + * + * Copyright 2010 Marvell International Ltd. + * Author: Haojian Zhuang <haojian.zhuang(a)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 "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) + +/* 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 SHORT_LO2 (1 << 6) +#define SHORT_LO1 (1 << 4) +#define SHORT_HS2 (1 << 2) +#define SHORT_HS1 (1 << 0) + +enum { + FILTER_BYPASS = 0, + FILTER_LOW_PASS_1, + FILTER_LOW_PASS_2, + FILTER_HIGH_PASS_3, + FILTER_MAX, +}; + +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; + + 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 */ +static int pm860x_rsync_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + /* unmute DAC */ + 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_adc_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + unsigned int en1 = 0, en2 = 0; + + if (!strcmp(w->name, "Left ADC")) { + en1 = ADC_MOD_LEFT; + en2 = ADC_LEFT; + } + if (!strcmp(w->name, "Right ADC")) { + en1 = ADC_MOD_RIGHT; + en2 = ADC_RIGHT; + } + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + snd_soc_update_bits(codec, PM860X_ADC_EN_1, en1, en1); + snd_soc_update_bits(codec, PM860X_ADC_EN_2, en2, en2); + break; + case SND_SOC_DAPM_PRE_PMD: + snd_soc_update_bits(codec, PM860X_ADC_EN_1, en1, 0); + snd_soc_update_bits(codec, PM860X_ADC_EN_2, en2, 0); + break; + } + 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) { + dac |= MODULATOR; + /* mute */ + 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) { + /* mute */ + 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 int pm860x_mic1_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + /* Enable Mic1 Bias & MICDET, HSDET */ + snd_soc_update_bits(codec, PM860X_ADC_ANA_1, MIC1BIAS_MASK, + MIC1BIAS_MASK); + pm860x_set_bits(codec->control_data, REG_MIC_DET, + MICDET_MASK, MICDET_MASK); + pm860x_set_bits(codec->control_data, REG_HS_DET, + EN_HS_DET, EN_HS_DET); + break; + case SND_SOC_DAPM_PRE_PMD: + /* disable Mic1 Bias & MICDET, HSDET */ + pm860x_set_bits(codec->control_data, REG_MIC_DET, + MICDET_MASK, 0); + pm860x_set_bits(codec->control_data, REG_HS_DET, + EN_HS_DET, 0); + snd_soc_update_bits(codec, PM860X_ADC_ANA_1, MIC1BIAS_MASK, 0); + 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_ADC_E("Left ADC", NULL, SND_SOC_NOPM, 0, 0, + pm860x_adc_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_ADC_E("Right ADC", NULL, SND_SOC_NOPM, 0, 0, + pm860x_adc_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_PRE_PMD), + + 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_E("Mic1 Bias", SND_SOC_NOPM, 0, 0, + &pm860x_mic1_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + 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), + + SND_SOC_DAPM_POST("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"}, + + /* 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"}, + + {"HS1", NULL, "Headset1 PGA"}, + {"HS2", NULL, "Headset2 PGA"}, + {"LINEOUT1", NULL, "Lineout1 PGA"}, + {"LINEOUT2", NULL, "Lineout2 PGA"}, + {"LSP", NULL, "Speaker PGA"}, + {"LSN", NULL, "Speaker PGA"}, + {"EARP", NULL, "Earpiece PGA"}, + {"EARN", NULL, "Earpiece PGA"}, +}; + +static int pm860x_digital_mute(struct snd_soc_dai *codec_dai, int mute) +{ + struct snd_soc_codec *codec = codec_dai->codec; + int data = 0; + + if (mute) + data = DAC_MUTE; + snd_soc_update_bits(codec, PM860X_DAC_OFFSET, DAC_MUTE, data); + snd_soc_update_bits(codec, PM860X_EAR_CTRL_2, + RSYNC_CHANGE, RSYNC_CHANGE); + return 0; +} + +static int set_dai_fmt(struct pm860x_priv *pm860x, unsigned int fmt, + unsigned char *inf, unsigned char *mask) +{ + *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; + 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; + 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; + + ret = set_dai_fmt(pm860x, fmt, &inf, &mask); + if (!ret) + snd_soc_update_bits(codec, PM860X_PCM_IFACE_2, mask, inf); + return ret; +} + +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 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; + int ret; + + ret = set_dai_fmt(pm860x, fmt, &inf, &mask); + if (!ret) + snd_soc_update_bits(codec, PM860X_I2S_IFACE_2, mask, inf); + return ret; +} + +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; + + status = pm860x_reg_read(pm860x->i2c, REG_STATUS_1); + shrt = pm860x_reg_read(pm860x->i2c, REG_SHORTS); + + if (status & HEADSET_STATUS) + report |= PM860X_DET_HEADSET; + if (status & MIC_STATUS) + report |= PM860X_DET_MIC; + if (status & HOOK_STATUS) + report |= PM860X_DET_HOOK; + if (shrt & (SHORT_LO1 | SHORT_LO2)) + report |= PM860X_SHORT_LINEOUT; + if (shrt & (SHORT_HS1 | SHORT_HS2)) + report |= PM860X_SHORT_HEADSET; + dev_dbg(pm860x->codec->dev, "report:0x%x\n", report); + return IRQ_HANDLED; +} + +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(a)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..b034de2 --- /dev/null +++ b/sound/soc/codecs/88pm860x-codec.h @@ -0,0 +1,92 @@ +/* + * 88pm860x-codec.h -- 88PM860x ALSA SoC Audio Driver + * + * Copyright 2010 Marvell International Ltd. + * Haojian Zhuang <haojian.zhuang(a)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 + +#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 --0016e6d63f44021a1f048e002b69 Content-Type: text/x-patch; charset=US-ASCII; name="0001-ASoC-add-88pm860x-codec-driver.patch" Content-Disposition: attachment; filename="0001-ASoC-add-88pm860x-codec-driver.patch" Content-Transfer-Encoding: base64 X-Attachment-Id: f_gcyfxo2w0 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1 0

[PATCH 1/5] mfd: add codec resource into 88pm860x driver
by Haojian Zhuang 12 Aug '10

12 Aug '10

Add codec IRQ resources that are used in 88pm860x codec driver. Signed-off-by: Haojian Zhuang <haojian.zhuang(a)marvell.com> --- drivers/mfd/88pm860x-core.c | 44 +++++++++++++++++++++++++++++++++++++++++++ 1 files changed, 44 insertions(+), 0 deletions(-) diff --git a/drivers/mfd/88pm860x-core.c b/drivers/mfd/88pm860x-core.c index 1580f1f..fba00b4 100644 --- a/drivers/mfd/88pm860x-core.c +++ b/drivers/mfd/88pm860x-core.c @@ -158,6 +158,43 @@ static struct mfd_cell onkey_devs[] = { }, }; +static struct resource codec_resources[] = { + { + /* Headset microphone insertion or removal */ + .name = "micin", + .start = PM8607_IRQ_MICIN, + .end = PM8607_IRQ_MICIN, + .flags = IORESOURCE_IRQ, + }, { + /* Hook-switch press or release */ + .name = "hook", + .start = PM8607_IRQ_HOOK, + .end = PM8607_IRQ_HOOK, + .flags = IORESOURCE_IRQ, + }, { + /* Headset insertion or removal */ + .name = "headset", + .start = PM8607_IRQ_HEADSET, + .end = PM8607_IRQ_HEADSET, + .flags = IORESOURCE_IRQ, + }, { + /* Audio short */ + .name = "audio-short", + .start = PM8607_IRQ_AUDIO_SHORT, + .end = PM8607_IRQ_AUDIO_SHORT, + .flags = IORESOURCE_IRQ, + }, +}; + +static struct mfd_cell codec_devs[] = { + { + .name = "88pm860x-codec", + .num_resources = 4, + .resources = &codec_resources[0], + .id = -1, + }, +}; + static struct resource regulator_resources[] = { PM8607_REG_RESOURCE(BUCK1, BUCK1), PM8607_REG_RESOURCE(BUCK2, BUCK2), @@ -695,6 +732,13 @@ static void __devinit device_8607_init(struct pm860x_chip *chip, goto out_dev; } + ret = mfd_add_devices(chip->dev, 0, &codec_devs[0], + ARRAY_SIZE(codec_devs), + &codec_resources[0], 0); + if (ret < 0) { + dev_err(chip->dev, "Failed to add codec subdev\n"); + goto out_dev; + } return; out_dev: mfd_remove_devices(chip->dev); -- 1.5.6.5

1 0

[PATCH 1/1] mfd: add codec resource into 88pm860x driver
by Haojian Zhuang 12 Aug '10

12 Aug '10

Add codec IRQ resources that are used in 88pm860x codec driver. Signed-off-by: Haojian Zhuang <haojian.zhuang(a)marvell.com> --- drivers/mfd/88pm860x-core.c | 44 +++++++++++++++++++++++++++++++++++++++++++ 1 files changed, 44 insertions(+), 0 deletions(-) diff --git a/drivers/mfd/88pm860x-core.c b/drivers/mfd/88pm860x-core.c index 07933f3..4db10a1 100644 --- a/drivers/mfd/88pm860x-core.c +++ b/drivers/mfd/88pm860x-core.c @@ -158,6 +158,43 @@ static struct mfd_cell onkey_devs[] = { }, }; +static struct resource codec_resources[] = { + { + /* Headset microphone insertion or removal */ + .name = "micin", + .start = PM8607_IRQ_MICIN, + .end = PM8607_IRQ_MICIN, + .flags = IORESOURCE_IRQ, + }, { + /* Hook-switch press or release */ + .name = "hook", + .start = PM8607_IRQ_HOOK, + .end = PM8607_IRQ_HOOK, + .flags = IORESOURCE_IRQ, + }, { + /* Headset insertion or removal */ + .name = "headset", + .start = PM8607_IRQ_HEADSET, + .end = PM8607_IRQ_HEADSET, + .flags = IORESOURCE_IRQ, + }, { + /* Audio short */ + .name = "audio-short", + .start = PM8607_IRQ_AUDIO_SHORT, + .end = PM8607_IRQ_AUDIO_SHORT, + .flags = IORESOURCE_IRQ, + }, +}; + +static struct mfd_cell codec_devs[] = { + { + .name = "88pm860x-codec", + .num_resources = ARRAY_SIZE(codec_resources), + .resources = &codec_resources[0], + .id = -1, + }, +}; + static struct resource regulator_resources[] = { PM8607_REG_RESOURCE(BUCK1, BUCK1), PM8607_REG_RESOURCE(BUCK2, BUCK2), @@ -687,6 +724,13 @@ static void __devinit device_8607_init(struct pm860x_chip *chip, goto out_dev; } + ret = mfd_add_devices(chip->dev, 0, &codec_devs[0], + ARRAY_SIZE(codec_devs), + &codec_resources[0], 0); + if (ret < 0) { + dev_err(chip->dev, "Failed to add codec subdev\n"); + goto out_dev; + } return; out_dev: mfd_remove_devices(chip->dev); -- 1.5.6.5 --0016e6daaffbfa18ad048e7b0ed9 Content-Type: text/x-patch; charset=US-ASCII; name="0001-mfd-add-codec-resource-into-88pm860x-driver.patch" Content-Disposition: attachment; filename="0001-mfd-add-codec-resource-into-88pm860x-driver.patch" Content-Transfer-Encoding: base64 X-Attachment-Id: f_gd76bllc0 RnJvbSAyOGVkNGFmMzkyOTEyMGI5YjM4NTNmZmNiODA1NWViNmE4MWQzMWQ4IE1vbiBTZXAgMTcg MDA6MDA6MDAgMjAwMQpGcm9tOiBIYW9qaWFuIFpodWFuZyA8aGFvamlhbi56aHVhbmdAbWFydmVs bC5jb20+CkRhdGU6IFRodSwgMTIgQXVnIDIwMTAgMTE6NTk6MzMgKzA4MDAKU3ViamVjdDogW1BB VENIIDEvMV0gbWZkOiBhZGQgY29kZWMgcmVzb3VyY2UgaW50byA4OHBtODYweCBkcml2ZXIKCkFk ZCBjb2RlYyBJUlEgcmVzb3VyY2VzIHRoYXQgYXJlIHVzZWQgaW4gODhwbTg2MHggY29kZWMgZHJp dmVyLgoKU2lnbmVkLW9mZi1ieTogSGFvamlhbiBaaHVhbmcgPGhhb2ppYW4uemh1YW5nQG1hcnZl bGwuY29tPgotLS0KIGRyaXZlcnMvbWZkLzg4cG04NjB4LWNvcmUuYyB8ICAgNDQgKysrKysrKysr KysrKysrKysrKysrKysrKysrKysrKysrKysrKysrKysrKwogMSBmaWxlcyBjaGFuZ2VkLCA0NCBp bnNlcnRpb25zKCspLCAwIGRlbGV0aW9ucygtKQoKZGlmZiAtLWdpdCBhL2RyaXZlcnMvbWZkLzg4 cG04NjB4LWNvcmUuYyBiL2RyaXZlcnMvbWZkLzg4cG04NjB4LWNvcmUuYwppbmRleCAwNzkzM2Yz Li40ZGIxMGExIDEwMDY0NAotLS0gYS9kcml2ZXJzL21mZC84OHBtODYweC1jb3JlLmMKKysrIGIv ZHJpdmVycy9tZmQvODhwbTg2MHgtY29yZS5jCkBAIC0xNTgsNiArMTU4LDQzIEBAIHN0YXRpYyBz dHJ1Y3QgbWZkX2NlbGwgb25rZXlfZGV2c1tdID0gewogCX0sCiB9OwogCitzdGF0aWMgc3RydWN0 IHJlc291cmNlIGNvZGVjX3Jlc291cmNlc1tdID0geworCXsKKwkJLyogSGVhZHNldCBtaWNyb3Bo b25lIGluc2VydGlvbiBvciByZW1vdmFsICovCisJCS5uYW1lCQk9ICJtaWNpbiIsCisJCS5zdGFy dAkJPSBQTTg2MDdfSVJRX01JQ0lOLAorCQkuZW5kCQk9IFBNODYwN19JUlFfTUlDSU4sCisJCS5m bGFncwkJPSBJT1JFU09VUkNFX0lSUSwKKwl9LCB7CisJCS8qIEhvb2stc3dpdGNoIHByZXNzIG9y IHJlbGVhc2UgKi8KKwkJLm5hbWUJCT0gImhvb2siLAorCQkuc3RhcnQJCT0gUE04NjA3X0lSUV9I T09LLAorCQkuZW5kCQk9IFBNODYwN19JUlFfSE9PSywKKwkJLmZsYWdzCQk9IElPUkVTT1VSQ0Vf SVJRLAorCX0sIHsKKwkJLyogSGVhZHNldCBpbnNlcnRpb24gb3IgcmVtb3ZhbCAqLworCQkubmFt ZQkJPSAiaGVhZHNldCIsCisJCS5zdGFydAkJPSBQTTg2MDdfSVJRX0hFQURTRVQsCisJCS5lbmQJ CT0gUE04NjA3X0lSUV9IRUFEU0VULAorCQkuZmxhZ3MJCT0gSU9SRVNPVVJDRV9JUlEsCisJfSwg eworCQkvKiBBdWRpbyBzaG9ydCAqLworCQkubmFtZQkJPSAiYXVkaW8tc2hvcnQiLAorCQkuc3Rh cnQJCT0gUE04NjA3X0lSUV9BVURJT19TSE9SVCwKKwkJLmVuZAkJPSBQTTg2MDdfSVJRX0FVRElP X1NIT1JULAorCQkuZmxhZ3MJCT0gSU9SRVNPVVJDRV9JUlEsCisJfSwKK307CisKK3N0YXRpYyBz dHJ1Y3QgbWZkX2NlbGwgY29kZWNfZGV2c1tdID0geworCXsKKwkJLm5hbWUJCT0gIjg4cG04NjB4 LWNvZGVjIiwKKwkJLm51bV9yZXNvdXJjZXMJPSBBUlJBWV9TSVpFKGNvZGVjX3Jlc291cmNlcyks CisJCS5yZXNvdXJjZXMJPSAmY29kZWNfcmVzb3VyY2VzWzBdLAorCQkuaWQJCT0gLTEsCisJfSwK K307CisKIHN0YXRpYyBzdHJ1Y3QgcmVzb3VyY2UgcmVndWxhdG9yX3Jlc291cmNlc1tdID0gewog CVBNODYwN19SRUdfUkVTT1VSQ0UoQlVDSzEsIEJVQ0sxKSwKIAlQTTg2MDdfUkVHX1JFU09VUkNF KEJVQ0syLCBCVUNLMiksCkBAIC02ODcsNiArNzI0LDEzIEBAIHN0YXRpYyB2b2lkIF9fZGV2aW5p dCBkZXZpY2VfODYwN19pbml0KHN0cnVjdCBwbTg2MHhfY2hpcCAqY2hpcCwKIAkJZ290byBvdXRf ZGV2OwogCX0KIAorCXJldCA9IG1mZF9hZGRfZGV2aWNlcyhjaGlwLT5kZXYsIDAsICZjb2RlY19k ZXZzWzBdLAorCQkJICAgICAgQVJSQVlfU0laRShjb2RlY19kZXZzKSwKKwkJCSAgICAgICZjb2Rl Y19yZXNvdXJjZXNbMF0sIDApOworCWlmIChyZXQgPCAwKSB7CisJCWRldl9lcnIoY2hpcC0+ZGV2 LCAiRmFpbGVkIHRvIGFkZCBjb2RlYyBzdWJkZXZcbiIpOworCQlnb3RvIG91dF9kZXY7CisJfQog CXJldHVybjsKIG91dF9kZXY6CiAJbWZkX3JlbW92ZV9kZXZpY2VzKGNoaXAtPmRldik7Ci0tIAox LjUuNi41Cgo= --0016e6daaffbfa18ad048e7b0ed9--

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[PATCH 3/5] ASoC: update setting for pxa ssp slave mode
by Haojian Zhuang 12 Aug '10

12 Aug '10

SCFR bit is required to be always set if pxa ssp is in slave mode. This bit indicates clock input to SSPSCLK is only active during data transfers. Signed-off-by: Haojian Zhuang <haojian.zhuang(a)marvell.com> --- sound/soc/pxa/pxa-ssp.c | 23 +++++++++++++++-------- 1 files changed, 15 insertions(+), 8 deletions(-) diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index c4f480d..8dfbcda 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -462,9 +462,7 @@ static int pxa_ssp_set_dai_fmt(struct snd_soc_dai *cpu_dai, { struct ssp_priv *priv = snd_soc_dai_get_drvdata(cpu_dai); struct ssp_device *ssp = priv->ssp; - u32 sscr0; - u32 sscr1; - u32 sspsp; + u32 sscr0, sscr1, sspsp, scfr; /* check if we need to change anything at all */ if (priv->dai_fmt == fmt) @@ -479,16 +477,16 @@ static int pxa_ssp_set_dai_fmt(struct snd_soc_dai *cpu_dai, /* reset port settings */ sscr0 = pxa_ssp_read_reg(ssp, SSCR0) & - (SSCR0_ECS | SSCR0_NCS | SSCR0_MOD | SSCR0_ACS); + ~(SSCR0_ECS | SSCR0_NCS | SSCR0_MOD | SSCR0_ACS); sscr1 = SSCR1_RxTresh(8) | SSCR1_TxTresh(7); sspsp = 0; switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBM_CFM: - sscr1 |= SSCR1_SCLKDIR | SSCR1_SFRMDIR; + sscr1 |= SSCR1_SCLKDIR | SSCR1_SFRMDIR | SSCR1_SCFR; break; case SND_SOC_DAIFMT_CBM_CFS: - sscr1 |= SSCR1_SCLKDIR; + sscr1 |= SSCR1_SCLKDIR | SSCR1_SCFR; break; case SND_SOC_DAIFMT_CBS_CFS: break; @@ -534,6 +532,17 @@ static int pxa_ssp_set_dai_fmt(struct snd_soc_dai *cpu_dai, pxa_ssp_write_reg(ssp, SSCR1, sscr1); pxa_ssp_write_reg(ssp, SSPSP, sspsp); + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + case SND_SOC_DAIFMT_CBM_CFS: + scfr = pxa_ssp_read_reg(ssp, SSCR1) | SSCR1_SCFR; + pxa_ssp_write_reg(ssp, SSCR1, scfr); + + while (pxa_ssp_read_reg(ssp, SSSR) & SSSR_BSY) + cpu_relax(); + break; + } + dump_registers(ssp); /* Since we are configuring the timings for the format by hand @@ -583,10 +592,8 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, /* clear selected SSP bits */ sscr0 = pxa_ssp_read_reg(ssp, SSCR0) & ~(SSCR0_DSS | SSCR0_EDSS); - pxa_ssp_write_reg(ssp, SSCR0, sscr0); /* bit size */ - sscr0 = pxa_ssp_read_reg(ssp, SSCR0); switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: #ifdef CONFIG_PXA3xx -- 1.5.6.5

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Re: [alsa-devel] [PATCH] sound: usb: USB3 Super Speed patch
by Clemens Ladisch 11 Aug '10

11 Aug '10

Paul Zimmerman wrote: > This patch adds Super Speed support to the USB drivers under sound/. It adds > tests for USB_SPEED_SUPER to all the places that check for the USB speed. > +++ b/sound/usb/midi.c > @@ -834,7 +834,8 @@ static void snd_usbmidi_us122l_output(struct snd_usb_midi_out_endpoint *ep, > +++ b/sound/usb/misc/ua101.c > +++ b/sound/usb/usx2y/us122l.c > +++ b/sound/usb/usx2y/usb_stream.c These devices do not support super speed. > +++ b/sound/usb/pcm.c > - if (snd_usb_get_speed(subs->dev) != USB_SPEED_HIGH) > + if (snd_usb_get_speed(subs->dev) != USB_SPEED_HIGH && > + snd_usb_get_speed(subs->dev) != USB_SPEED_SUPER) > /* full speed devices have fixed data packet interval */ > ptmin = 1000; In places like this, it would be better to write something like if (snd_usb_get_speed(subs->dev) == USB_SPEED_FULL) or if (snd_usb_get_speed(subs->dev) < USB_SPEED_HIGH) Regards, Clemens

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[alsa-devel] ASoC: multi-component - two cleanup patches
by Sascha Hauer 11 Aug '10

11 Aug '10

Hi, Two small patches you might consider adding to the multi-component rework. Sascha

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[alsa-devel] i.MX SDMA support
by Sascha Hauer 11 Aug '10

11 Aug '10

Hi, I just posted a series adding SDMA support for i.MX to the Arm Linux Kernel mailing list. It also contains a patch for adding SDMA support to the i.MX SSI units. I haven't cc'd alsa-devel because most of the stuff is not sound related. I know at least Mark and Liam are interested, so I leave a pointer here. The sound stuff needs to be synchronized with the other patches, so it's probably best if all patches go via my tree. Sascha -- Pengutronix e.K. | | Industrial Linux Solutions | http://www.pengutronix.de/ | Peiner Str. 6-8, 31137 Hildesheim, Germany | Phone: +49-5121-206917-0 | Amtsgericht Hildesheim, HRA 2686 | Fax: +49-5121-206917-5555 |

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[alsa-devel] About the platform_name field in snd_soc_dai_link
by Sascha Hauer 11 Aug '10

11 Aug '10

Hi, I'm wondering what exactly a snd_soc_platform is. Apparently the snd_pcm_ops/pcm_new/pcm_free are specific to a platform. For my understanding these operations are more specific to a cpu_dai. Looking at the tree it seems that each cpu_dai has exactly one possible platform, which seems logical to me because the cpu_dai knows how to transfer the data. My problem on i.MX is that I currently have two possible cpu dais (imx-ssi.[01]) and each can be configured to use dma or fiq depending on the dma capabilities. So the cpu_dai knows which pcm_ops we have to use, but currently it's the soc glue code which has to decide in platform_name. Am I understanding something wrong here? Sascha -- Pengutronix e.K. | | Industrial Linux Solutions | http://www.pengutronix.de/ | Peiner Str. 6-8, 31137 Hildesheim, Germany | Phone: +49-5121-206917-0 | Amtsgericht Hildesheim, HRA 2686 | Fax: +49-5121-206917-5555 |

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