/* * 88pm860x-codec.c -- 88PM860x ALSA SoC Audio Driver * * Copyright 2010 Marvell International Ltd. * Author: Haojian Zhuang * * 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 #include #include #include #include #include #include #include #include #include #include #include #include #include #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->dapm.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->dapm.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 (status & (HEADSET_STATUS | MIC_STATUS | SHORT_HS1 | SHORT_HS2 | SHORT_LO1 | SHORT_LO2)) trace_snd_soc_jack_irq(dev_name(pm860x->codec->dev)); 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); struct snd_soc_dapm_context *dapm = &codec->dapm; 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; } } 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; } snd_soc_add_controls(codec, pm860x_snd_controls, ARRAY_SIZE(pm860x_snd_controls)); snd_soc_dapm_new_controls(dapm, pm860x_dapm_widgets, ARRAY_SIZE(pm860x_dapm_widgets)); snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); return 0; out: while (--i >= 0) free_irq(pm860x->irq[i], pm860x); return ret; } 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 "); MODULE_LICENSE("GPL"); MODULE_ALIAS("platform:88pm860x-codec");