diff options
Diffstat (limited to 'ANDROID_3.4.5/sound/soc/codecs/tlv320aic3x.c')
-rw-r--r-- | ANDROID_3.4.5/sound/soc/codecs/tlv320aic3x.c | 1505 |
1 files changed, 1505 insertions, 0 deletions
diff --git a/ANDROID_3.4.5/sound/soc/codecs/tlv320aic3x.c b/ANDROID_3.4.5/sound/soc/codecs/tlv320aic3x.c new file mode 100644 index 00000000..b8f0262a --- /dev/null +++ b/ANDROID_3.4.5/sound/soc/codecs/tlv320aic3x.c @@ -0,0 +1,1505 @@ +/* + * ALSA SoC TLV320AIC3X codec driver + * + * Author: Vladimir Barinov, <vbarinov@embeddedalley.com> + * Copyright: (C) 2007 MontaVista Software, Inc., <source@mvista.com> + * + * Based on sound/soc/codecs/wm8753.c by Liam Girdwood + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * Notes: + * The AIC3X is a driver for a low power stereo audio + * codecs aic31, aic32, aic33, aic3007. + * + * It supports full aic33 codec functionality. + * The compatibility with aic32, aic31 and aic3007 is as follows: + * aic32/aic3007 | aic31 + * --------------------------------------- + * MONO_LOUT -> N/A | MONO_LOUT -> N/A + * | IN1L -> LINE1L + * | IN1R -> LINE1R + * | IN2L -> LINE2L + * | IN2R -> LINE2R + * | MIC3L/R -> N/A + * truncated internal functionality in + * accordance with documentation + * --------------------------------------- + * + * Hence the machine layer should disable unsupported inputs/outputs by + * snd_soc_dapm_disable_pin(codec, "MONO_LOUT"), etc. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/gpio.h> +#include <linux/regulator/consumer.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/initval.h> +#include <sound/tlv.h> +#include <sound/tlv320aic3x.h> + +#include "tlv320aic3x.h" + +#define AIC3X_NUM_SUPPLIES 4 +static const char *aic3x_supply_names[AIC3X_NUM_SUPPLIES] = { + "IOVDD", /* I/O Voltage */ + "DVDD", /* Digital Core Voltage */ + "AVDD", /* Analog DAC Voltage */ + "DRVDD", /* ADC Analog and Output Driver Voltage */ +}; + +static LIST_HEAD(reset_list); + +struct aic3x_priv; + +struct aic3x_disable_nb { + struct notifier_block nb; + struct aic3x_priv *aic3x; +}; + +/* codec private data */ +struct aic3x_priv { + struct snd_soc_codec *codec; + struct regulator_bulk_data supplies[AIC3X_NUM_SUPPLIES]; + struct aic3x_disable_nb disable_nb[AIC3X_NUM_SUPPLIES]; + enum snd_soc_control_type control_type; + struct aic3x_setup_data *setup; + unsigned int sysclk; + struct list_head list; + int master; + int gpio_reset; + int power; +#define AIC3X_MODEL_3X 0 +#define AIC3X_MODEL_33 1 +#define AIC3X_MODEL_3007 2 + u16 model; +}; + +/* + * AIC3X register cache + * We can't read the AIC3X register space when we are + * using 2 wire for device control, so we cache them instead. + * There is no point in caching the reset register + */ +static const u8 aic3x_reg[AIC3X_CACHEREGNUM] = { + 0x00, 0x00, 0x00, 0x10, /* 0 */ + 0x04, 0x00, 0x00, 0x00, /* 4 */ + 0x00, 0x00, 0x00, 0x01, /* 8 */ + 0x00, 0x00, 0x00, 0x80, /* 12 */ + 0x80, 0xff, 0xff, 0x78, /* 16 */ + 0x78, 0x78, 0x78, 0x78, /* 20 */ + 0x78, 0x00, 0x00, 0xfe, /* 24 */ + 0x00, 0x00, 0xfe, 0x00, /* 28 */ + 0x18, 0x18, 0x00, 0x00, /* 32 */ + 0x00, 0x00, 0x00, 0x00, /* 36 */ + 0x00, 0x00, 0x00, 0x80, /* 40 */ + 0x80, 0x00, 0x00, 0x00, /* 44 */ + 0x00, 0x00, 0x00, 0x04, /* 48 */ + 0x00, 0x00, 0x00, 0x00, /* 52 */ + 0x00, 0x00, 0x04, 0x00, /* 56 */ + 0x00, 0x00, 0x00, 0x00, /* 60 */ + 0x00, 0x04, 0x00, 0x00, /* 64 */ + 0x00, 0x00, 0x00, 0x00, /* 68 */ + 0x04, 0x00, 0x00, 0x00, /* 72 */ + 0x00, 0x00, 0x00, 0x00, /* 76 */ + 0x00, 0x00, 0x00, 0x00, /* 80 */ + 0x00, 0x00, 0x00, 0x00, /* 84 */ + 0x00, 0x00, 0x00, 0x00, /* 88 */ + 0x00, 0x00, 0x00, 0x00, /* 92 */ + 0x00, 0x00, 0x00, 0x00, /* 96 */ + 0x00, 0x00, 0x02, /* 100 */ +}; + +#define SOC_DAPM_SINGLE_AIC3X(xname, reg, shift, mask, invert) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .info = snd_soc_info_volsw, \ + .get = snd_soc_dapm_get_volsw, .put = snd_soc_dapm_put_volsw_aic3x, \ + .private_value = SOC_SINGLE_VALUE(reg, shift, mask, invert) } + +/* + * All input lines are connected when !0xf and disconnected with 0xf bit field, + * so we have to use specific dapm_put call for input mixer + */ +static int snd_soc_dapm_put_volsw_aic3x(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol); + struct snd_soc_dapm_widget *widget = wlist->widgets[0]; + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + unsigned int reg = mc->reg; + unsigned int shift = mc->shift; + int max = mc->max; + unsigned int mask = (1 << fls(max)) - 1; + unsigned int invert = mc->invert; + unsigned short val, val_mask; + int ret; + struct snd_soc_dapm_path *path; + int found = 0; + + val = (ucontrol->value.integer.value[0] & mask); + + mask = 0xf; + if (val) + val = mask; + + if (invert) + val = mask - val; + val_mask = mask << shift; + val = val << shift; + + mutex_lock(&widget->codec->mutex); + + if (snd_soc_test_bits(widget->codec, reg, val_mask, val)) { + /* find dapm widget path assoc with kcontrol */ + list_for_each_entry(path, &widget->dapm->card->paths, list) { + if (path->kcontrol != kcontrol) + continue; + + /* found, now check type */ + found = 1; + if (val) + /* new connection */ + path->connect = invert ? 0 : 1; + else + /* old connection must be powered down */ + path->connect = invert ? 1 : 0; + + dapm_mark_dirty(path->source, "tlv320aic3x source"); + dapm_mark_dirty(path->sink, "tlv320aic3x sink"); + + break; + } + + if (found) + snd_soc_dapm_sync(widget->dapm); + } + + ret = snd_soc_update_bits(widget->codec, reg, val_mask, val); + + mutex_unlock(&widget->codec->mutex); + return ret; +} + +static const char *aic3x_left_dac_mux[] = { "DAC_L1", "DAC_L3", "DAC_L2" }; +static const char *aic3x_right_dac_mux[] = { "DAC_R1", "DAC_R3", "DAC_R2" }; +static const char *aic3x_left_hpcom_mux[] = + { "differential of HPLOUT", "constant VCM", "single-ended" }; +static const char *aic3x_right_hpcom_mux[] = + { "differential of HPROUT", "constant VCM", "single-ended", + "differential of HPLCOM", "external feedback" }; +static const char *aic3x_linein_mode_mux[] = { "single-ended", "differential" }; +static const char *aic3x_adc_hpf[] = + { "Disabled", "0.0045xFs", "0.0125xFs", "0.025xFs" }; + +#define LDAC_ENUM 0 +#define RDAC_ENUM 1 +#define LHPCOM_ENUM 2 +#define RHPCOM_ENUM 3 +#define LINE1L_2_L_ENUM 4 +#define LINE1L_2_R_ENUM 5 +#define LINE1R_2_L_ENUM 6 +#define LINE1R_2_R_ENUM 7 +#define LINE2L_ENUM 8 +#define LINE2R_ENUM 9 +#define ADC_HPF_ENUM 10 + +static const struct soc_enum aic3x_enum[] = { + SOC_ENUM_SINGLE(DAC_LINE_MUX, 6, 3, aic3x_left_dac_mux), + SOC_ENUM_SINGLE(DAC_LINE_MUX, 4, 3, aic3x_right_dac_mux), + SOC_ENUM_SINGLE(HPLCOM_CFG, 4, 3, aic3x_left_hpcom_mux), + SOC_ENUM_SINGLE(HPRCOM_CFG, 3, 5, aic3x_right_hpcom_mux), + SOC_ENUM_SINGLE(LINE1L_2_LADC_CTRL, 7, 2, aic3x_linein_mode_mux), + SOC_ENUM_SINGLE(LINE1L_2_RADC_CTRL, 7, 2, aic3x_linein_mode_mux), + SOC_ENUM_SINGLE(LINE1R_2_LADC_CTRL, 7, 2, aic3x_linein_mode_mux), + SOC_ENUM_SINGLE(LINE1R_2_RADC_CTRL, 7, 2, aic3x_linein_mode_mux), + SOC_ENUM_SINGLE(LINE2L_2_LADC_CTRL, 7, 2, aic3x_linein_mode_mux), + SOC_ENUM_SINGLE(LINE2R_2_RADC_CTRL, 7, 2, aic3x_linein_mode_mux), + SOC_ENUM_DOUBLE(AIC3X_CODEC_DFILT_CTRL, 6, 4, 4, aic3x_adc_hpf), +}; + +/* + * DAC digital volumes. From -63.5 to 0 dB in 0.5 dB steps + */ +static DECLARE_TLV_DB_SCALE(dac_tlv, -6350, 50, 0); +/* ADC PGA gain volumes. From 0 to 59.5 dB in 0.5 dB steps */ +static DECLARE_TLV_DB_SCALE(adc_tlv, 0, 50, 0); +/* + * Output stage volumes. From -78.3 to 0 dB. Muted below -78.3 dB. + * Step size is approximately 0.5 dB over most of the scale but increasing + * near the very low levels. + * Define dB scale so that it is mostly correct for range about -55 to 0 dB + * but having increasing dB difference below that (and where it doesn't count + * so much). This setting shows -50 dB (actual is -50.3 dB) for register + * value 100 and -58.5 dB (actual is -78.3 dB) for register value 117. + */ +static DECLARE_TLV_DB_SCALE(output_stage_tlv, -5900, 50, 1); + +static const struct snd_kcontrol_new aic3x_snd_controls[] = { + /* Output */ + SOC_DOUBLE_R_TLV("PCM Playback Volume", + LDAC_VOL, RDAC_VOL, 0, 0x7f, 1, dac_tlv), + + /* + * Output controls that map to output mixer switches. Note these are + * only for swapped L-to-R and R-to-L routes. See below stereo controls + * for direct L-to-L and R-to-R routes. + */ + SOC_SINGLE_TLV("Left Line Mixer Line2R Bypass Volume", + LINE2R_2_LLOPM_VOL, 0, 118, 1, output_stage_tlv), + SOC_SINGLE_TLV("Left Line Mixer PGAR Bypass Volume", + PGAR_2_LLOPM_VOL, 0, 118, 1, output_stage_tlv), + SOC_SINGLE_TLV("Left Line Mixer DACR1 Playback Volume", + DACR1_2_LLOPM_VOL, 0, 118, 1, output_stage_tlv), + + SOC_SINGLE_TLV("Right Line Mixer Line2L Bypass Volume", + LINE2L_2_RLOPM_VOL, 0, 118, 1, output_stage_tlv), + SOC_SINGLE_TLV("Right Line Mixer PGAL Bypass Volume", + PGAL_2_RLOPM_VOL, 0, 118, 1, output_stage_tlv), + SOC_SINGLE_TLV("Right Line Mixer DACL1 Playback Volume", + DACL1_2_RLOPM_VOL, 0, 118, 1, output_stage_tlv), + + SOC_SINGLE_TLV("Left HP Mixer Line2R Bypass Volume", + LINE2R_2_HPLOUT_VOL, 0, 118, 1, output_stage_tlv), + SOC_SINGLE_TLV("Left HP Mixer PGAR Bypass Volume", + PGAR_2_HPLOUT_VOL, 0, 118, 1, output_stage_tlv), + SOC_SINGLE_TLV("Left HP Mixer DACR1 Playback Volume", + DACR1_2_HPLOUT_VOL, 0, 118, 1, output_stage_tlv), + + SOC_SINGLE_TLV("Right HP Mixer Line2L Bypass Volume", + LINE2L_2_HPROUT_VOL, 0, 118, 1, output_stage_tlv), + SOC_SINGLE_TLV("Right HP Mixer PGAL Bypass Volume", + PGAL_2_HPROUT_VOL, 0, 118, 1, output_stage_tlv), + SOC_SINGLE_TLV("Right HP Mixer DACL1 Playback Volume", + DACL1_2_HPROUT_VOL, 0, 118, 1, output_stage_tlv), + + SOC_SINGLE_TLV("Left HPCOM Mixer Line2R Bypass Volume", + LINE2R_2_HPLCOM_VOL, 0, 118, 1, output_stage_tlv), + SOC_SINGLE_TLV("Left HPCOM Mixer PGAR Bypass Volume", + PGAR_2_HPLCOM_VOL, 0, 118, 1, output_stage_tlv), + SOC_SINGLE_TLV("Left HPCOM Mixer DACR1 Playback Volume", + DACR1_2_HPLCOM_VOL, 0, 118, 1, output_stage_tlv), + + SOC_SINGLE_TLV("Right HPCOM Mixer Line2L Bypass Volume", + LINE2L_2_HPRCOM_VOL, 0, 118, 1, output_stage_tlv), + SOC_SINGLE_TLV("Right HPCOM Mixer PGAL Bypass Volume", + PGAL_2_HPRCOM_VOL, 0, 118, 1, output_stage_tlv), + SOC_SINGLE_TLV("Right HPCOM Mixer DACL1 Playback Volume", + DACL1_2_HPRCOM_VOL, 0, 118, 1, output_stage_tlv), + + /* Stereo output controls for direct L-to-L and R-to-R routes */ + SOC_DOUBLE_R_TLV("Line Line2 Bypass Volume", + LINE2L_2_LLOPM_VOL, LINE2R_2_RLOPM_VOL, + 0, 118, 1, output_stage_tlv), + SOC_DOUBLE_R_TLV("Line PGA Bypass Volume", + PGAL_2_LLOPM_VOL, PGAR_2_RLOPM_VOL, + 0, 118, 1, output_stage_tlv), + SOC_DOUBLE_R_TLV("Line DAC Playback Volume", + DACL1_2_LLOPM_VOL, DACR1_2_RLOPM_VOL, + 0, 118, 1, output_stage_tlv), + + SOC_DOUBLE_R_TLV("Mono Line2 Bypass Volume", + LINE2L_2_MONOLOPM_VOL, LINE2R_2_MONOLOPM_VOL, + 0, 118, 1, output_stage_tlv), + SOC_DOUBLE_R_TLV("Mono PGA Bypass Volume", + PGAL_2_MONOLOPM_VOL, PGAR_2_MONOLOPM_VOL, + 0, 118, 1, output_stage_tlv), + SOC_DOUBLE_R_TLV("Mono DAC Playback Volume", + DACL1_2_MONOLOPM_VOL, DACR1_2_MONOLOPM_VOL, + 0, 118, 1, output_stage_tlv), + + SOC_DOUBLE_R_TLV("HP Line2 Bypass Volume", + LINE2L_2_HPLOUT_VOL, LINE2R_2_HPROUT_VOL, + 0, 118, 1, output_stage_tlv), + SOC_DOUBLE_R_TLV("HP PGA Bypass Volume", + PGAL_2_HPLOUT_VOL, PGAR_2_HPROUT_VOL, + 0, 118, 1, output_stage_tlv), + SOC_DOUBLE_R_TLV("HP DAC Playback Volume", + DACL1_2_HPLOUT_VOL, DACR1_2_HPROUT_VOL, + 0, 118, 1, output_stage_tlv), + + SOC_DOUBLE_R_TLV("HPCOM Line2 Bypass Volume", + LINE2L_2_HPLCOM_VOL, LINE2R_2_HPRCOM_VOL, + 0, 118, 1, output_stage_tlv), + SOC_DOUBLE_R_TLV("HPCOM PGA Bypass Volume", + PGAL_2_HPLCOM_VOL, PGAR_2_HPRCOM_VOL, + 0, 118, 1, output_stage_tlv), + SOC_DOUBLE_R_TLV("HPCOM DAC Playback Volume", + DACL1_2_HPLCOM_VOL, DACR1_2_HPRCOM_VOL, + 0, 118, 1, output_stage_tlv), + + /* Output pin mute controls */ + SOC_DOUBLE_R("Line Playback Switch", LLOPM_CTRL, RLOPM_CTRL, 3, + 0x01, 0), + SOC_SINGLE("Mono Playback Switch", MONOLOPM_CTRL, 3, 0x01, 0), + SOC_DOUBLE_R("HP Playback Switch", HPLOUT_CTRL, HPROUT_CTRL, 3, + 0x01, 0), + SOC_DOUBLE_R("HPCOM Playback Switch", HPLCOM_CTRL, HPRCOM_CTRL, 3, + 0x01, 0), + + /* + * Note: enable Automatic input Gain Controller with care. It can + * adjust PGA to max value when ADC is on and will never go back. + */ + SOC_DOUBLE_R("AGC Switch", LAGC_CTRL_A, RAGC_CTRL_A, 7, 0x01, 0), + + /* Input */ + SOC_DOUBLE_R_TLV("PGA Capture Volume", LADC_VOL, RADC_VOL, + 0, 119, 0, adc_tlv), + SOC_DOUBLE_R("PGA Capture Switch", LADC_VOL, RADC_VOL, 7, 0x01, 1), + + SOC_ENUM("ADC HPF Cut-off", aic3x_enum[ADC_HPF_ENUM]), +}; + +/* + * Class-D amplifier gain. From 0 to 18 dB in 6 dB steps + */ +static DECLARE_TLV_DB_SCALE(classd_amp_tlv, 0, 600, 0); + +static const struct snd_kcontrol_new aic3x_classd_amp_gain_ctrl = + SOC_DOUBLE_TLV("Class-D Amplifier Gain", CLASSD_CTRL, 6, 4, 3, 0, classd_amp_tlv); + +/* Left DAC Mux */ +static const struct snd_kcontrol_new aic3x_left_dac_mux_controls = +SOC_DAPM_ENUM("Route", aic3x_enum[LDAC_ENUM]); + +/* Right DAC Mux */ +static const struct snd_kcontrol_new aic3x_right_dac_mux_controls = +SOC_DAPM_ENUM("Route", aic3x_enum[RDAC_ENUM]); + +/* Left HPCOM Mux */ +static const struct snd_kcontrol_new aic3x_left_hpcom_mux_controls = +SOC_DAPM_ENUM("Route", aic3x_enum[LHPCOM_ENUM]); + +/* Right HPCOM Mux */ +static const struct snd_kcontrol_new aic3x_right_hpcom_mux_controls = +SOC_DAPM_ENUM("Route", aic3x_enum[RHPCOM_ENUM]); + +/* Left Line Mixer */ +static const struct snd_kcontrol_new aic3x_left_line_mixer_controls[] = { + SOC_DAPM_SINGLE("Line2L Bypass Switch", LINE2L_2_LLOPM_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("PGAL Bypass Switch", PGAL_2_LLOPM_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("DACL1 Switch", DACL1_2_LLOPM_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("Line2R Bypass Switch", LINE2R_2_LLOPM_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("PGAR Bypass Switch", PGAR_2_LLOPM_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("DACR1 Switch", DACR1_2_LLOPM_VOL, 7, 1, 0), +}; + +/* Right Line Mixer */ +static const struct snd_kcontrol_new aic3x_right_line_mixer_controls[] = { + SOC_DAPM_SINGLE("Line2L Bypass Switch", LINE2L_2_RLOPM_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("PGAL Bypass Switch", PGAL_2_RLOPM_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("DACL1 Switch", DACL1_2_RLOPM_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("Line2R Bypass Switch", LINE2R_2_RLOPM_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("PGAR Bypass Switch", PGAR_2_RLOPM_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("DACR1 Switch", DACR1_2_RLOPM_VOL, 7, 1, 0), +}; + +/* Mono Mixer */ +static const struct snd_kcontrol_new aic3x_mono_mixer_controls[] = { + SOC_DAPM_SINGLE("Line2L Bypass Switch", LINE2L_2_MONOLOPM_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("PGAL Bypass Switch", PGAL_2_MONOLOPM_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("DACL1 Switch", DACL1_2_MONOLOPM_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("Line2R Bypass Switch", LINE2R_2_MONOLOPM_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("PGAR Bypass Switch", PGAR_2_MONOLOPM_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("DACR1 Switch", DACR1_2_MONOLOPM_VOL, 7, 1, 0), +}; + +/* Left HP Mixer */ +static const struct snd_kcontrol_new aic3x_left_hp_mixer_controls[] = { + SOC_DAPM_SINGLE("Line2L Bypass Switch", LINE2L_2_HPLOUT_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("PGAL Bypass Switch", PGAL_2_HPLOUT_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("DACL1 Switch", DACL1_2_HPLOUT_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("Line2R Bypass Switch", LINE2R_2_HPLOUT_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("PGAR Bypass Switch", PGAR_2_HPLOUT_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("DACR1 Switch", DACR1_2_HPLOUT_VOL, 7, 1, 0), +}; + +/* Right HP Mixer */ +static const struct snd_kcontrol_new aic3x_right_hp_mixer_controls[] = { + SOC_DAPM_SINGLE("Line2L Bypass Switch", LINE2L_2_HPROUT_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("PGAL Bypass Switch", PGAL_2_HPROUT_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("DACL1 Switch", DACL1_2_HPROUT_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("Line2R Bypass Switch", LINE2R_2_HPROUT_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("PGAR Bypass Switch", PGAR_2_HPROUT_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("DACR1 Switch", DACR1_2_HPROUT_VOL, 7, 1, 0), +}; + +/* Left HPCOM Mixer */ +static const struct snd_kcontrol_new aic3x_left_hpcom_mixer_controls[] = { + SOC_DAPM_SINGLE("Line2L Bypass Switch", LINE2L_2_HPLCOM_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("PGAL Bypass Switch", PGAL_2_HPLCOM_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("DACL1 Switch", DACL1_2_HPLCOM_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("Line2R Bypass Switch", LINE2R_2_HPLCOM_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("PGAR Bypass Switch", PGAR_2_HPLCOM_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("DACR1 Switch", DACR1_2_HPLCOM_VOL, 7, 1, 0), +}; + +/* Right HPCOM Mixer */ +static const struct snd_kcontrol_new aic3x_right_hpcom_mixer_controls[] = { + SOC_DAPM_SINGLE("Line2L Bypass Switch", LINE2L_2_HPRCOM_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("PGAL Bypass Switch", PGAL_2_HPRCOM_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("DACL1 Switch", DACL1_2_HPRCOM_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("Line2R Bypass Switch", LINE2R_2_HPRCOM_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("PGAR Bypass Switch", PGAR_2_HPRCOM_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("DACR1 Switch", DACR1_2_HPRCOM_VOL, 7, 1, 0), +}; + +/* Left PGA Mixer */ +static const struct snd_kcontrol_new aic3x_left_pga_mixer_controls[] = { + SOC_DAPM_SINGLE_AIC3X("Line1L Switch", LINE1L_2_LADC_CTRL, 3, 1, 1), + SOC_DAPM_SINGLE_AIC3X("Line1R Switch", LINE1R_2_LADC_CTRL, 3, 1, 1), + SOC_DAPM_SINGLE_AIC3X("Line2L Switch", LINE2L_2_LADC_CTRL, 3, 1, 1), + SOC_DAPM_SINGLE_AIC3X("Mic3L Switch", MIC3LR_2_LADC_CTRL, 4, 1, 1), + SOC_DAPM_SINGLE_AIC3X("Mic3R Switch", MIC3LR_2_LADC_CTRL, 0, 1, 1), +}; + +/* Right PGA Mixer */ +static const struct snd_kcontrol_new aic3x_right_pga_mixer_controls[] = { + SOC_DAPM_SINGLE_AIC3X("Line1R Switch", LINE1R_2_RADC_CTRL, 3, 1, 1), + SOC_DAPM_SINGLE_AIC3X("Line1L Switch", LINE1L_2_RADC_CTRL, 3, 1, 1), + SOC_DAPM_SINGLE_AIC3X("Line2R Switch", LINE2R_2_RADC_CTRL, 3, 1, 1), + SOC_DAPM_SINGLE_AIC3X("Mic3L Switch", MIC3LR_2_RADC_CTRL, 4, 1, 1), + SOC_DAPM_SINGLE_AIC3X("Mic3R Switch", MIC3LR_2_RADC_CTRL, 0, 1, 1), +}; + +/* Left Line1 Mux */ +static const struct snd_kcontrol_new aic3x_left_line1l_mux_controls = +SOC_DAPM_ENUM("Route", aic3x_enum[LINE1L_2_L_ENUM]); +static const struct snd_kcontrol_new aic3x_right_line1l_mux_controls = +SOC_DAPM_ENUM("Route", aic3x_enum[LINE1L_2_R_ENUM]); + +/* Right Line1 Mux */ +static const struct snd_kcontrol_new aic3x_right_line1r_mux_controls = +SOC_DAPM_ENUM("Route", aic3x_enum[LINE1R_2_R_ENUM]); +static const struct snd_kcontrol_new aic3x_left_line1r_mux_controls = +SOC_DAPM_ENUM("Route", aic3x_enum[LINE1R_2_L_ENUM]); + +/* Left Line2 Mux */ +static const struct snd_kcontrol_new aic3x_left_line2_mux_controls = +SOC_DAPM_ENUM("Route", aic3x_enum[LINE2L_ENUM]); + +/* Right Line2 Mux */ +static const struct snd_kcontrol_new aic3x_right_line2_mux_controls = +SOC_DAPM_ENUM("Route", aic3x_enum[LINE2R_ENUM]); + +static const struct snd_soc_dapm_widget aic3x_dapm_widgets[] = { + /* Left DAC to Left Outputs */ + SND_SOC_DAPM_DAC("Left DAC", "Left Playback", DAC_PWR, 7, 0), + SND_SOC_DAPM_MUX("Left DAC Mux", SND_SOC_NOPM, 0, 0, + &aic3x_left_dac_mux_controls), + SND_SOC_DAPM_MUX("Left HPCOM Mux", SND_SOC_NOPM, 0, 0, + &aic3x_left_hpcom_mux_controls), + SND_SOC_DAPM_PGA("Left Line Out", LLOPM_CTRL, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("Left HP Out", HPLOUT_CTRL, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("Left HP Com", HPLCOM_CTRL, 0, 0, NULL, 0), + + /* Right DAC to Right Outputs */ + SND_SOC_DAPM_DAC("Right DAC", "Right Playback", DAC_PWR, 6, 0), + SND_SOC_DAPM_MUX("Right DAC Mux", SND_SOC_NOPM, 0, 0, + &aic3x_right_dac_mux_controls), + SND_SOC_DAPM_MUX("Right HPCOM Mux", SND_SOC_NOPM, 0, 0, + &aic3x_right_hpcom_mux_controls), + SND_SOC_DAPM_PGA("Right Line Out", RLOPM_CTRL, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("Right HP Out", HPROUT_CTRL, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("Right HP Com", HPRCOM_CTRL, 0, 0, NULL, 0), + + /* Mono Output */ + SND_SOC_DAPM_PGA("Mono Out", MONOLOPM_CTRL, 0, 0, NULL, 0), + + /* Inputs to Left ADC */ + SND_SOC_DAPM_ADC("Left ADC", "Left Capture", LINE1L_2_LADC_CTRL, 2, 0), + SND_SOC_DAPM_MIXER("Left PGA Mixer", SND_SOC_NOPM, 0, 0, + &aic3x_left_pga_mixer_controls[0], + ARRAY_SIZE(aic3x_left_pga_mixer_controls)), + SND_SOC_DAPM_MUX("Left Line1L Mux", SND_SOC_NOPM, 0, 0, + &aic3x_left_line1l_mux_controls), + SND_SOC_DAPM_MUX("Left Line1R Mux", SND_SOC_NOPM, 0, 0, + &aic3x_left_line1r_mux_controls), + SND_SOC_DAPM_MUX("Left Line2L Mux", SND_SOC_NOPM, 0, 0, + &aic3x_left_line2_mux_controls), + + /* Inputs to Right ADC */ + SND_SOC_DAPM_ADC("Right ADC", "Right Capture", + LINE1R_2_RADC_CTRL, 2, 0), + SND_SOC_DAPM_MIXER("Right PGA Mixer", SND_SOC_NOPM, 0, 0, + &aic3x_right_pga_mixer_controls[0], + ARRAY_SIZE(aic3x_right_pga_mixer_controls)), + SND_SOC_DAPM_MUX("Right Line1L Mux", SND_SOC_NOPM, 0, 0, + &aic3x_right_line1l_mux_controls), + SND_SOC_DAPM_MUX("Right Line1R Mux", SND_SOC_NOPM, 0, 0, + &aic3x_right_line1r_mux_controls), + SND_SOC_DAPM_MUX("Right Line2R Mux", SND_SOC_NOPM, 0, 0, + &aic3x_right_line2_mux_controls), + + /* + * Not a real mic bias widget but similar function. This is for dynamic + * control of GPIO1 digital mic modulator clock output function when + * using digital mic. + */ + SND_SOC_DAPM_REG(snd_soc_dapm_micbias, "GPIO1 dmic modclk", + AIC3X_GPIO1_REG, 4, 0xf, + AIC3X_GPIO1_FUNC_DIGITAL_MIC_MODCLK, + AIC3X_GPIO1_FUNC_DISABLED), + + /* + * Also similar function like mic bias. Selects digital mic with + * configurable oversampling rate instead of ADC converter. + */ + SND_SOC_DAPM_REG(snd_soc_dapm_micbias, "DMic Rate 128", + AIC3X_ASD_INTF_CTRLA, 0, 3, 1, 0), + SND_SOC_DAPM_REG(snd_soc_dapm_micbias, "DMic Rate 64", + AIC3X_ASD_INTF_CTRLA, 0, 3, 2, 0), + SND_SOC_DAPM_REG(snd_soc_dapm_micbias, "DMic Rate 32", + AIC3X_ASD_INTF_CTRLA, 0, 3, 3, 0), + + /* Mic Bias */ + SND_SOC_DAPM_REG(snd_soc_dapm_micbias, "Mic Bias 2V", + MICBIAS_CTRL, 6, 3, 1, 0), + SND_SOC_DAPM_REG(snd_soc_dapm_micbias, "Mic Bias 2.5V", + MICBIAS_CTRL, 6, 3, 2, 0), + SND_SOC_DAPM_REG(snd_soc_dapm_micbias, "Mic Bias AVDD", + MICBIAS_CTRL, 6, 3, 3, 0), + + /* Output mixers */ + SND_SOC_DAPM_MIXER("Left Line Mixer", SND_SOC_NOPM, 0, 0, + &aic3x_left_line_mixer_controls[0], + ARRAY_SIZE(aic3x_left_line_mixer_controls)), + SND_SOC_DAPM_MIXER("Right Line Mixer", SND_SOC_NOPM, 0, 0, + &aic3x_right_line_mixer_controls[0], + ARRAY_SIZE(aic3x_right_line_mixer_controls)), + SND_SOC_DAPM_MIXER("Mono Mixer", SND_SOC_NOPM, 0, 0, + &aic3x_mono_mixer_controls[0], + ARRAY_SIZE(aic3x_mono_mixer_controls)), + SND_SOC_DAPM_MIXER("Left HP Mixer", SND_SOC_NOPM, 0, 0, + &aic3x_left_hp_mixer_controls[0], + ARRAY_SIZE(aic3x_left_hp_mixer_controls)), + SND_SOC_DAPM_MIXER("Right HP Mixer", SND_SOC_NOPM, 0, 0, + &aic3x_right_hp_mixer_controls[0], + ARRAY_SIZE(aic3x_right_hp_mixer_controls)), + SND_SOC_DAPM_MIXER("Left HPCOM Mixer", SND_SOC_NOPM, 0, 0, + &aic3x_left_hpcom_mixer_controls[0], + ARRAY_SIZE(aic3x_left_hpcom_mixer_controls)), + SND_SOC_DAPM_MIXER("Right HPCOM Mixer", SND_SOC_NOPM, 0, 0, + &aic3x_right_hpcom_mixer_controls[0], + ARRAY_SIZE(aic3x_right_hpcom_mixer_controls)), + + SND_SOC_DAPM_OUTPUT("LLOUT"), + SND_SOC_DAPM_OUTPUT("RLOUT"), + SND_SOC_DAPM_OUTPUT("MONO_LOUT"), + SND_SOC_DAPM_OUTPUT("HPLOUT"), + SND_SOC_DAPM_OUTPUT("HPROUT"), + SND_SOC_DAPM_OUTPUT("HPLCOM"), + SND_SOC_DAPM_OUTPUT("HPRCOM"), + + SND_SOC_DAPM_INPUT("MIC3L"), + SND_SOC_DAPM_INPUT("MIC3R"), + SND_SOC_DAPM_INPUT("LINE1L"), + SND_SOC_DAPM_INPUT("LINE1R"), + SND_SOC_DAPM_INPUT("LINE2L"), + SND_SOC_DAPM_INPUT("LINE2R"), + + /* + * Virtual output pin to detection block inside codec. This can be + * used to keep codec bias on if gpio or detection features are needed. + * Force pin on or construct a path with an input jack and mic bias + * widgets. + */ + SND_SOC_DAPM_OUTPUT("Detection"), +}; + +static const struct snd_soc_dapm_widget aic3007_dapm_widgets[] = { + /* Class-D outputs */ + SND_SOC_DAPM_PGA("Left Class-D Out", CLASSD_CTRL, 3, 0, NULL, 0), + SND_SOC_DAPM_PGA("Right Class-D Out", CLASSD_CTRL, 2, 0, NULL, 0), + + SND_SOC_DAPM_OUTPUT("SPOP"), + SND_SOC_DAPM_OUTPUT("SPOM"), +}; + +static const struct snd_soc_dapm_route intercon[] = { + /* Left Input */ + {"Left Line1L Mux", "single-ended", "LINE1L"}, + {"Left Line1L Mux", "differential", "LINE1L"}, + + {"Left Line2L Mux", "single-ended", "LINE2L"}, + {"Left Line2L Mux", "differential", "LINE2L"}, + + {"Left PGA Mixer", "Line1L Switch", "Left Line1L Mux"}, + {"Left PGA Mixer", "Line1R Switch", "Left Line1R Mux"}, + {"Left PGA Mixer", "Line2L Switch", "Left Line2L Mux"}, + {"Left PGA Mixer", "Mic3L Switch", "MIC3L"}, + {"Left PGA Mixer", "Mic3R Switch", "MIC3R"}, + + {"Left ADC", NULL, "Left PGA Mixer"}, + {"Left ADC", NULL, "GPIO1 dmic modclk"}, + + /* Right Input */ + {"Right Line1R Mux", "single-ended", "LINE1R"}, + {"Right Line1R Mux", "differential", "LINE1R"}, + + {"Right Line2R Mux", "single-ended", "LINE2R"}, + {"Right Line2R Mux", "differential", "LINE2R"}, + + {"Right PGA Mixer", "Line1L Switch", "Right Line1L Mux"}, + {"Right PGA Mixer", "Line1R Switch", "Right Line1R Mux"}, + {"Right PGA Mixer", "Line2R Switch", "Right Line2R Mux"}, + {"Right PGA Mixer", "Mic3L Switch", "MIC3L"}, + {"Right PGA Mixer", "Mic3R Switch", "MIC3R"}, + + {"Right ADC", NULL, "Right PGA Mixer"}, + {"Right ADC", NULL, "GPIO1 dmic modclk"}, + + /* + * Logical path between digital mic enable and GPIO1 modulator clock + * output function + */ + {"GPIO1 dmic modclk", NULL, "DMic Rate 128"}, + {"GPIO1 dmic modclk", NULL, "DMic Rate 64"}, + {"GPIO1 dmic modclk", NULL, "DMic Rate 32"}, + + /* Left DAC Output */ + {"Left DAC Mux", "DAC_L1", "Left DAC"}, + {"Left DAC Mux", "DAC_L2", "Left DAC"}, + {"Left DAC Mux", "DAC_L3", "Left DAC"}, + + /* Right DAC Output */ + {"Right DAC Mux", "DAC_R1", "Right DAC"}, + {"Right DAC Mux", "DAC_R2", "Right DAC"}, + {"Right DAC Mux", "DAC_R3", "Right DAC"}, + + /* Left Line Output */ + {"Left Line Mixer", "Line2L Bypass Switch", "Left Line2L Mux"}, + {"Left Line Mixer", "PGAL Bypass Switch", "Left PGA Mixer"}, + {"Left Line Mixer", "DACL1 Switch", "Left DAC Mux"}, + {"Left Line Mixer", "Line2R Bypass Switch", "Right Line2R Mux"}, + {"Left Line Mixer", "PGAR Bypass Switch", "Right PGA Mixer"}, + {"Left Line Mixer", "DACR1 Switch", "Right DAC Mux"}, + + {"Left Line Out", NULL, "Left Line Mixer"}, + {"Left Line Out", NULL, "Left DAC Mux"}, + {"LLOUT", NULL, "Left Line Out"}, + + /* Right Line Output */ + {"Right Line Mixer", "Line2L Bypass Switch", "Left Line2L Mux"}, + {"Right Line Mixer", "PGAL Bypass Switch", "Left PGA Mixer"}, + {"Right Line Mixer", "DACL1 Switch", "Left DAC Mux"}, + {"Right Line Mixer", "Line2R Bypass Switch", "Right Line2R Mux"}, + {"Right Line Mixer", "PGAR Bypass Switch", "Right PGA Mixer"}, + {"Right Line Mixer", "DACR1 Switch", "Right DAC Mux"}, + + {"Right Line Out", NULL, "Right Line Mixer"}, + {"Right Line Out", NULL, "Right DAC Mux"}, + {"RLOUT", NULL, "Right Line Out"}, + + /* Mono Output */ + {"Mono Mixer", "Line2L Bypass Switch", "Left Line2L Mux"}, + {"Mono Mixer", "PGAL Bypass Switch", "Left PGA Mixer"}, + {"Mono Mixer", "DACL1 Switch", "Left DAC Mux"}, + {"Mono Mixer", "Line2R Bypass Switch", "Right Line2R Mux"}, + {"Mono Mixer", "PGAR Bypass Switch", "Right PGA Mixer"}, + {"Mono Mixer", "DACR1 Switch", "Right DAC Mux"}, + + {"Mono Out", NULL, "Mono Mixer"}, + {"MONO_LOUT", NULL, "Mono Out"}, + + /* Left HP Output */ + {"Left HP Mixer", "Line2L Bypass Switch", "Left Line2L Mux"}, + {"Left HP Mixer", "PGAL Bypass Switch", "Left PGA Mixer"}, + {"Left HP Mixer", "DACL1 Switch", "Left DAC Mux"}, + {"Left HP Mixer", "Line2R Bypass Switch", "Right Line2R Mux"}, + {"Left HP Mixer", "PGAR Bypass Switch", "Right PGA Mixer"}, + {"Left HP Mixer", "DACR1 Switch", "Right DAC Mux"}, + + {"Left HP Out", NULL, "Left HP Mixer"}, + {"Left HP Out", NULL, "Left DAC Mux"}, + {"HPLOUT", NULL, "Left HP Out"}, + + /* Right HP Output */ + {"Right HP Mixer", "Line2L Bypass Switch", "Left Line2L Mux"}, + {"Right HP Mixer", "PGAL Bypass Switch", "Left PGA Mixer"}, + {"Right HP Mixer", "DACL1 Switch", "Left DAC Mux"}, + {"Right HP Mixer", "Line2R Bypass Switch", "Right Line2R Mux"}, + {"Right HP Mixer", "PGAR Bypass Switch", "Right PGA Mixer"}, + {"Right HP Mixer", "DACR1 Switch", "Right DAC Mux"}, + + {"Right HP Out", NULL, "Right HP Mixer"}, + {"Right HP Out", NULL, "Right DAC Mux"}, + {"HPROUT", NULL, "Right HP Out"}, + + /* Left HPCOM Output */ + {"Left HPCOM Mixer", "Line2L Bypass Switch", "Left Line2L Mux"}, + {"Left HPCOM Mixer", "PGAL Bypass Switch", "Left PGA Mixer"}, + {"Left HPCOM Mixer", "DACL1 Switch", "Left DAC Mux"}, + {"Left HPCOM Mixer", "Line2R Bypass Switch", "Right Line2R Mux"}, + {"Left HPCOM Mixer", "PGAR Bypass Switch", "Right PGA Mixer"}, + {"Left HPCOM Mixer", "DACR1 Switch", "Right DAC Mux"}, + + {"Left HPCOM Mux", "differential of HPLOUT", "Left HP Mixer"}, + {"Left HPCOM Mux", "constant VCM", "Left HPCOM Mixer"}, + {"Left HPCOM Mux", "single-ended", "Left HPCOM Mixer"}, + {"Left HP Com", NULL, "Left HPCOM Mux"}, + {"HPLCOM", NULL, "Left HP Com"}, + + /* Right HPCOM Output */ + {"Right HPCOM Mixer", "Line2L Bypass Switch", "Left Line2L Mux"}, + {"Right HPCOM Mixer", "PGAL Bypass Switch", "Left PGA Mixer"}, + {"Right HPCOM Mixer", "DACL1 Switch", "Left DAC Mux"}, + {"Right HPCOM Mixer", "Line2R Bypass Switch", "Right Line2R Mux"}, + {"Right HPCOM Mixer", "PGAR Bypass Switch", "Right PGA Mixer"}, + {"Right HPCOM Mixer", "DACR1 Switch", "Right DAC Mux"}, + + {"Right HPCOM Mux", "differential of HPROUT", "Right HP Mixer"}, + {"Right HPCOM Mux", "constant VCM", "Right HPCOM Mixer"}, + {"Right HPCOM Mux", "single-ended", "Right HPCOM Mixer"}, + {"Right HPCOM Mux", "differential of HPLCOM", "Left HPCOM Mixer"}, + {"Right HPCOM Mux", "external feedback", "Right HPCOM Mixer"}, + {"Right HP Com", NULL, "Right HPCOM Mux"}, + {"HPRCOM", NULL, "Right HP Com"}, +}; + +static const struct snd_soc_dapm_route intercon_3007[] = { + /* Class-D outputs */ + {"Left Class-D Out", NULL, "Left Line Out"}, + {"Right Class-D Out", NULL, "Left Line Out"}, + {"SPOP", NULL, "Left Class-D Out"}, + {"SPOM", NULL, "Right Class-D Out"}, +}; + +static int aic3x_add_widgets(struct snd_soc_codec *codec) +{ + struct aic3x_priv *aic3x = snd_soc_codec_get_drvdata(codec); + struct snd_soc_dapm_context *dapm = &codec->dapm; + + snd_soc_dapm_new_controls(dapm, aic3x_dapm_widgets, + ARRAY_SIZE(aic3x_dapm_widgets)); + + /* set up audio path interconnects */ + snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon)); + + if (aic3x->model == AIC3X_MODEL_3007) { + snd_soc_dapm_new_controls(dapm, aic3007_dapm_widgets, + ARRAY_SIZE(aic3007_dapm_widgets)); + snd_soc_dapm_add_routes(dapm, intercon_3007, + ARRAY_SIZE(intercon_3007)); + } + + return 0; +} + +static int aic3x_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec *codec =rtd->codec; + struct aic3x_priv *aic3x = snd_soc_codec_get_drvdata(codec); + int codec_clk = 0, bypass_pll = 0, fsref, last_clk = 0; + u8 data, j, r, p, pll_q, pll_p = 1, pll_r = 1, pll_j = 1; + u16 d, pll_d = 1; + int clk; + + /* select data word length */ + data = snd_soc_read(codec, AIC3X_ASD_INTF_CTRLB) & (~(0x3 << 4)); + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + break; + case SNDRV_PCM_FORMAT_S20_3LE: + data |= (0x01 << 4); + break; + case SNDRV_PCM_FORMAT_S24_LE: + data |= (0x02 << 4); + break; + case SNDRV_PCM_FORMAT_S32_LE: + data |= (0x03 << 4); + break; + } + snd_soc_write(codec, AIC3X_ASD_INTF_CTRLB, data); + + /* Fsref can be 44100 or 48000 */ + fsref = (params_rate(params) % 11025 == 0) ? 44100 : 48000; + + /* Try to find a value for Q which allows us to bypass the PLL and + * generate CODEC_CLK directly. */ + for (pll_q = 2; pll_q < 18; pll_q++) + if (aic3x->sysclk / (128 * pll_q) == fsref) { + bypass_pll = 1; + break; + } + + if (bypass_pll) { + pll_q &= 0xf; + snd_soc_write(codec, AIC3X_PLL_PROGA_REG, pll_q << PLLQ_SHIFT); + snd_soc_write(codec, AIC3X_GPIOB_REG, CODEC_CLKIN_CLKDIV); + /* disable PLL if it is bypassed */ + snd_soc_update_bits(codec, AIC3X_PLL_PROGA_REG, PLL_ENABLE, 0); + + } else { + snd_soc_write(codec, AIC3X_GPIOB_REG, CODEC_CLKIN_PLLDIV); + /* enable PLL when it is used */ + snd_soc_update_bits(codec, AIC3X_PLL_PROGA_REG, + PLL_ENABLE, PLL_ENABLE); + } + + /* Route Left DAC to left channel input and + * right DAC to right channel input */ + data = (LDAC2LCH | RDAC2RCH); + data |= (fsref == 44100) ? FSREF_44100 : FSREF_48000; + if (params_rate(params) >= 64000) + data |= DUAL_RATE_MODE; + snd_soc_write(codec, AIC3X_CODEC_DATAPATH_REG, data); + + /* codec sample rate select */ + data = (fsref * 20) / params_rate(params); + if (params_rate(params) < 64000) + data /= 2; + data /= 5; + data -= 2; + data |= (data << 4); + snd_soc_write(codec, AIC3X_SAMPLE_RATE_SEL_REG, data); + + if (bypass_pll) + return 0; + + /* Use PLL, compute appropriate setup for j, d, r and p, the closest + * one wins the game. Try with d==0 first, next with d!=0. + * Constraints for j are according to the datasheet. + * The sysclk is divided by 1000 to prevent integer overflows. + */ + + codec_clk = (2048 * fsref) / (aic3x->sysclk / 1000); + + for (r = 1; r <= 16; r++) + for (p = 1; p <= 8; p++) { + for (j = 4; j <= 55; j++) { + /* This is actually 1000*((j+(d/10000))*r)/p + * The term had to be converted to get + * rid of the division by 10000; d = 0 here + */ + int tmp_clk = (1000 * j * r) / p; + + /* Check whether this values get closer than + * the best ones we had before + */ + if (abs(codec_clk - tmp_clk) < + abs(codec_clk - last_clk)) { + pll_j = j; pll_d = 0; + pll_r = r; pll_p = p; + last_clk = tmp_clk; + } + + /* Early exit for exact matches */ + if (tmp_clk == codec_clk) + goto found; + } + } + + /* try with d != 0 */ + for (p = 1; p <= 8; p++) { + j = codec_clk * p / 1000; + + if (j < 4 || j > 11) + continue; + + /* do not use codec_clk here since we'd loose precision */ + d = ((2048 * p * fsref) - j * aic3x->sysclk) + * 100 / (aic3x->sysclk/100); + + clk = (10000 * j + d) / (10 * p); + + /* check whether this values get closer than the best + * ones we had before */ + if (abs(codec_clk - clk) < abs(codec_clk - last_clk)) { + pll_j = j; pll_d = d; pll_r = 1; pll_p = p; + last_clk = clk; + } + + /* Early exit for exact matches */ + if (clk == codec_clk) + goto found; + } + + if (last_clk == 0) { + printk(KERN_ERR "%s(): unable to setup PLL\n", __func__); + return -EINVAL; + } + +found: + snd_soc_update_bits(codec, AIC3X_PLL_PROGA_REG, PLLP_MASK, pll_p); + snd_soc_write(codec, AIC3X_OVRF_STATUS_AND_PLLR_REG, + pll_r << PLLR_SHIFT); + snd_soc_write(codec, AIC3X_PLL_PROGB_REG, pll_j << PLLJ_SHIFT); + snd_soc_write(codec, AIC3X_PLL_PROGC_REG, + (pll_d >> 6) << PLLD_MSB_SHIFT); + snd_soc_write(codec, AIC3X_PLL_PROGD_REG, + (pll_d & 0x3F) << PLLD_LSB_SHIFT); + + return 0; +} + +static int aic3x_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + u8 ldac_reg = snd_soc_read(codec, LDAC_VOL) & ~MUTE_ON; + u8 rdac_reg = snd_soc_read(codec, RDAC_VOL) & ~MUTE_ON; + + if (mute) { + snd_soc_write(codec, LDAC_VOL, ldac_reg | MUTE_ON); + snd_soc_write(codec, RDAC_VOL, rdac_reg | MUTE_ON); + } else { + snd_soc_write(codec, LDAC_VOL, ldac_reg); + snd_soc_write(codec, RDAC_VOL, rdac_reg); + } + + return 0; +} + +static int aic3x_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 aic3x_priv *aic3x = snd_soc_codec_get_drvdata(codec); + + aic3x->sysclk = freq; + return 0; +} + +static int aic3x_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct aic3x_priv *aic3x = snd_soc_codec_get_drvdata(codec); + u8 iface_areg, iface_breg; + int delay = 0; + + iface_areg = snd_soc_read(codec, AIC3X_ASD_INTF_CTRLA) & 0x3f; + iface_breg = snd_soc_read(codec, AIC3X_ASD_INTF_CTRLB) & 0x3f; + + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + aic3x->master = 1; + iface_areg |= BIT_CLK_MASTER | WORD_CLK_MASTER; + break; + case SND_SOC_DAIFMT_CBS_CFS: + aic3x->master = 0; + iface_areg &= ~(BIT_CLK_MASTER | WORD_CLK_MASTER); + break; + default: + return -EINVAL; + } + + /* + * match both interface format and signal polarities since they + * are fixed + */ + switch (fmt & (SND_SOC_DAIFMT_FORMAT_MASK | + SND_SOC_DAIFMT_INV_MASK)) { + case (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF): + break; + case (SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_IB_NF): + delay = 1; + case (SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_IB_NF): + iface_breg |= (0x01 << 6); + break; + case (SND_SOC_DAIFMT_RIGHT_J | SND_SOC_DAIFMT_NB_NF): + iface_breg |= (0x02 << 6); + break; + case (SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_NB_NF): + iface_breg |= (0x03 << 6); + break; + default: + return -EINVAL; + } + + /* set iface */ + snd_soc_write(codec, AIC3X_ASD_INTF_CTRLA, iface_areg); + snd_soc_write(codec, AIC3X_ASD_INTF_CTRLB, iface_breg); + snd_soc_write(codec, AIC3X_ASD_INTF_CTRLC, delay); + + return 0; +} + +static int aic3x_init_3007(struct snd_soc_codec *codec) +{ + u8 tmp1, tmp2, *cache = codec->reg_cache; + + /* + * There is no need to cache writes to undocumented page 0xD but + * respective page 0 register cache entries must be preserved + */ + tmp1 = cache[0xD]; + tmp2 = cache[0x8]; + /* Class-D speaker driver init; datasheet p. 46 */ + snd_soc_write(codec, AIC3X_PAGE_SELECT, 0x0D); + snd_soc_write(codec, 0xD, 0x0D); + snd_soc_write(codec, 0x8, 0x5C); + snd_soc_write(codec, 0x8, 0x5D); + snd_soc_write(codec, 0x8, 0x5C); + snd_soc_write(codec, AIC3X_PAGE_SELECT, 0x00); + cache[0xD] = tmp1; + cache[0x8] = tmp2; + + return 0; +} + +static int aic3x_regulator_event(struct notifier_block *nb, + unsigned long event, void *data) +{ + struct aic3x_disable_nb *disable_nb = + container_of(nb, struct aic3x_disable_nb, nb); + struct aic3x_priv *aic3x = disable_nb->aic3x; + + if (event & REGULATOR_EVENT_DISABLE) { + /* + * Put codec to reset and require cache sync as at least one + * of the supplies was disabled + */ + if (gpio_is_valid(aic3x->gpio_reset)) + gpio_set_value(aic3x->gpio_reset, 0); + aic3x->codec->cache_sync = 1; + } + + return 0; +} + +static int aic3x_set_power(struct snd_soc_codec *codec, int power) +{ + struct aic3x_priv *aic3x = snd_soc_codec_get_drvdata(codec); + int i, ret; + u8 *cache = codec->reg_cache; + + if (power) { + ret = regulator_bulk_enable(ARRAY_SIZE(aic3x->supplies), + aic3x->supplies); + if (ret) + goto out; + aic3x->power = 1; + /* + * Reset release and cache sync is necessary only if some + * supply was off or if there were cached writes + */ + if (!codec->cache_sync) + goto out; + + if (gpio_is_valid(aic3x->gpio_reset)) { + udelay(1); + gpio_set_value(aic3x->gpio_reset, 1); + } + + /* Sync reg_cache with the hardware */ + codec->cache_only = 0; + for (i = AIC3X_SAMPLE_RATE_SEL_REG; i < ARRAY_SIZE(aic3x_reg); i++) + snd_soc_write(codec, i, cache[i]); + if (aic3x->model == AIC3X_MODEL_3007) + aic3x_init_3007(codec); + codec->cache_sync = 0; + } else { + /* + * Do soft reset to this codec instance in order to clear + * possible VDD leakage currents in case the supply regulators + * remain on + */ + snd_soc_write(codec, AIC3X_RESET, SOFT_RESET); + codec->cache_sync = 1; + aic3x->power = 0; + /* HW writes are needless when bias is off */ + codec->cache_only = 1; + ret = regulator_bulk_disable(ARRAY_SIZE(aic3x->supplies), + aic3x->supplies); + } +out: + return ret; +} + +static int aic3x_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct aic3x_priv *aic3x = snd_soc_codec_get_drvdata(codec); + + switch (level) { + case SND_SOC_BIAS_ON: + break; + case SND_SOC_BIAS_PREPARE: + if (codec->dapm.bias_level == SND_SOC_BIAS_STANDBY && + aic3x->master) { + /* enable pll */ + snd_soc_update_bits(codec, AIC3X_PLL_PROGA_REG, + PLL_ENABLE, PLL_ENABLE); + } + break; + case SND_SOC_BIAS_STANDBY: + if (!aic3x->power) + aic3x_set_power(codec, 1); + if (codec->dapm.bias_level == SND_SOC_BIAS_PREPARE && + aic3x->master) { + /* disable pll */ + snd_soc_update_bits(codec, AIC3X_PLL_PROGA_REG, + PLL_ENABLE, 0); + } + break; + case SND_SOC_BIAS_OFF: + if (aic3x->power) + aic3x_set_power(codec, 0); + break; + } + codec->dapm.bias_level = level; + + return 0; +} + +void aic3x_set_headset_detection(struct snd_soc_codec *codec, int detect, + int headset_debounce, int button_debounce) +{ + u8 val; + + val = ((detect & AIC3X_HEADSET_DETECT_MASK) + << AIC3X_HEADSET_DETECT_SHIFT) | + ((headset_debounce & AIC3X_HEADSET_DEBOUNCE_MASK) + << AIC3X_HEADSET_DEBOUNCE_SHIFT) | + ((button_debounce & AIC3X_BUTTON_DEBOUNCE_MASK) + << AIC3X_BUTTON_DEBOUNCE_SHIFT); + + if (detect & AIC3X_HEADSET_DETECT_MASK) + val |= AIC3X_HEADSET_DETECT_ENABLED; + + snd_soc_write(codec, AIC3X_HEADSET_DETECT_CTRL_A, val); +} + +#define AIC3X_RATES SNDRV_PCM_RATE_8000_96000 +#define AIC3X_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE) + +static const struct snd_soc_dai_ops aic3x_dai_ops = { + .hw_params = aic3x_hw_params, + .digital_mute = aic3x_mute, + .set_sysclk = aic3x_set_dai_sysclk, + .set_fmt = aic3x_set_dai_fmt, +}; + +static struct snd_soc_dai_driver aic3x_dai = { + .name = "tlv320aic3x-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = AIC3X_RATES, + .formats = AIC3X_FORMATS,}, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = AIC3X_RATES, + .formats = AIC3X_FORMATS,}, + .ops = &aic3x_dai_ops, + .symmetric_rates = 1, +}; + +static int aic3x_suspend(struct snd_soc_codec *codec) +{ + aic3x_set_bias_level(codec, SND_SOC_BIAS_OFF); + + return 0; +} + +static int aic3x_resume(struct snd_soc_codec *codec) +{ + aic3x_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + return 0; +} + +/* + * initialise the AIC3X driver + * register the mixer and dsp interfaces with the kernel + */ +static int aic3x_init(struct snd_soc_codec *codec) +{ + struct aic3x_priv *aic3x = snd_soc_codec_get_drvdata(codec); + + snd_soc_write(codec, AIC3X_PAGE_SELECT, PAGE0_SELECT); + snd_soc_write(codec, AIC3X_RESET, SOFT_RESET); + + /* DAC default volume and mute */ + snd_soc_write(codec, LDAC_VOL, DEFAULT_VOL | MUTE_ON); + snd_soc_write(codec, RDAC_VOL, DEFAULT_VOL | MUTE_ON); + + /* DAC to HP default volume and route to Output mixer */ + snd_soc_write(codec, DACL1_2_HPLOUT_VOL, DEFAULT_VOL | ROUTE_ON); + snd_soc_write(codec, DACR1_2_HPROUT_VOL, DEFAULT_VOL | ROUTE_ON); + snd_soc_write(codec, DACL1_2_HPLCOM_VOL, DEFAULT_VOL | ROUTE_ON); + snd_soc_write(codec, DACR1_2_HPRCOM_VOL, DEFAULT_VOL | ROUTE_ON); + /* DAC to Line Out default volume and route to Output mixer */ + snd_soc_write(codec, DACL1_2_LLOPM_VOL, DEFAULT_VOL | ROUTE_ON); + snd_soc_write(codec, DACR1_2_RLOPM_VOL, DEFAULT_VOL | ROUTE_ON); + /* DAC to Mono Line Out default volume and route to Output mixer */ + snd_soc_write(codec, DACL1_2_MONOLOPM_VOL, DEFAULT_VOL | ROUTE_ON); + snd_soc_write(codec, DACR1_2_MONOLOPM_VOL, DEFAULT_VOL | ROUTE_ON); + + /* unmute all outputs */ + snd_soc_update_bits(codec, LLOPM_CTRL, UNMUTE, UNMUTE); + snd_soc_update_bits(codec, RLOPM_CTRL, UNMUTE, UNMUTE); + snd_soc_update_bits(codec, MONOLOPM_CTRL, UNMUTE, UNMUTE); + snd_soc_update_bits(codec, HPLOUT_CTRL, UNMUTE, UNMUTE); + snd_soc_update_bits(codec, HPROUT_CTRL, UNMUTE, UNMUTE); + snd_soc_update_bits(codec, HPLCOM_CTRL, UNMUTE, UNMUTE); + snd_soc_update_bits(codec, HPRCOM_CTRL, UNMUTE, UNMUTE); + + /* ADC default volume and unmute */ + snd_soc_write(codec, LADC_VOL, DEFAULT_GAIN); + snd_soc_write(codec, RADC_VOL, DEFAULT_GAIN); + /* By default route Line1 to ADC PGA mixer */ + snd_soc_write(codec, LINE1L_2_LADC_CTRL, 0x0); + snd_soc_write(codec, LINE1R_2_RADC_CTRL, 0x0); + + /* PGA to HP Bypass default volume, disconnect from Output Mixer */ + snd_soc_write(codec, PGAL_2_HPLOUT_VOL, DEFAULT_VOL); + snd_soc_write(codec, PGAR_2_HPROUT_VOL, DEFAULT_VOL); + snd_soc_write(codec, PGAL_2_HPLCOM_VOL, DEFAULT_VOL); + snd_soc_write(codec, PGAR_2_HPRCOM_VOL, DEFAULT_VOL); + /* PGA to Line Out default volume, disconnect from Output Mixer */ + snd_soc_write(codec, PGAL_2_LLOPM_VOL, DEFAULT_VOL); + snd_soc_write(codec, PGAR_2_RLOPM_VOL, DEFAULT_VOL); + /* PGA to Mono Line Out default volume, disconnect from Output Mixer */ + snd_soc_write(codec, PGAL_2_MONOLOPM_VOL, DEFAULT_VOL); + snd_soc_write(codec, PGAR_2_MONOLOPM_VOL, DEFAULT_VOL); + + /* Line2 to HP Bypass default volume, disconnect from Output Mixer */ + snd_soc_write(codec, LINE2L_2_HPLOUT_VOL, DEFAULT_VOL); + snd_soc_write(codec, LINE2R_2_HPROUT_VOL, DEFAULT_VOL); + snd_soc_write(codec, LINE2L_2_HPLCOM_VOL, DEFAULT_VOL); + snd_soc_write(codec, LINE2R_2_HPRCOM_VOL, DEFAULT_VOL); + /* Line2 Line Out default volume, disconnect from Output Mixer */ + snd_soc_write(codec, LINE2L_2_LLOPM_VOL, DEFAULT_VOL); + snd_soc_write(codec, LINE2R_2_RLOPM_VOL, DEFAULT_VOL); + /* Line2 to Mono Out default volume, disconnect from Output Mixer */ + snd_soc_write(codec, LINE2L_2_MONOLOPM_VOL, DEFAULT_VOL); + snd_soc_write(codec, LINE2R_2_MONOLOPM_VOL, DEFAULT_VOL); + + if (aic3x->model == AIC3X_MODEL_3007) { + aic3x_init_3007(codec); + snd_soc_write(codec, CLASSD_CTRL, 0); + } + + return 0; +} + +static bool aic3x_is_shared_reset(struct aic3x_priv *aic3x) +{ + struct aic3x_priv *a; + + list_for_each_entry(a, &reset_list, list) { + if (gpio_is_valid(aic3x->gpio_reset) && + aic3x->gpio_reset == a->gpio_reset) + return true; + } + + return false; +} + +static int aic3x_probe(struct snd_soc_codec *codec) +{ + struct aic3x_priv *aic3x = snd_soc_codec_get_drvdata(codec); + int ret, i; + + INIT_LIST_HEAD(&aic3x->list); + aic3x->codec = codec; + + ret = snd_soc_codec_set_cache_io(codec, 8, 8, aic3x->control_type); + if (ret != 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } + + if (gpio_is_valid(aic3x->gpio_reset) && + !aic3x_is_shared_reset(aic3x)) { + ret = gpio_request(aic3x->gpio_reset, "tlv320aic3x reset"); + if (ret != 0) + goto err_gpio; + gpio_direction_output(aic3x->gpio_reset, 0); + } + + for (i = 0; i < ARRAY_SIZE(aic3x->supplies); i++) + aic3x->supplies[i].supply = aic3x_supply_names[i]; + + ret = regulator_bulk_get(codec->dev, ARRAY_SIZE(aic3x->supplies), + aic3x->supplies); + if (ret != 0) { + dev_err(codec->dev, "Failed to request supplies: %d\n", ret); + goto err_get; + } + for (i = 0; i < ARRAY_SIZE(aic3x->supplies); i++) { + aic3x->disable_nb[i].nb.notifier_call = aic3x_regulator_event; + aic3x->disable_nb[i].aic3x = aic3x; + ret = regulator_register_notifier(aic3x->supplies[i].consumer, + &aic3x->disable_nb[i].nb); + if (ret) { + dev_err(codec->dev, + "Failed to request regulator notifier: %d\n", + ret); + goto err_notif; + } + } + + codec->cache_only = 1; + aic3x_init(codec); + + if (aic3x->setup) { + /* setup GPIO functions */ + snd_soc_write(codec, AIC3X_GPIO1_REG, + (aic3x->setup->gpio_func[0] & 0xf) << 4); + snd_soc_write(codec, AIC3X_GPIO2_REG, + (aic3x->setup->gpio_func[1] & 0xf) << 4); + } + + snd_soc_add_codec_controls(codec, aic3x_snd_controls, + ARRAY_SIZE(aic3x_snd_controls)); + if (aic3x->model == AIC3X_MODEL_3007) + snd_soc_add_codec_controls(codec, &aic3x_classd_amp_gain_ctrl, 1); + + aic3x_add_widgets(codec); + list_add(&aic3x->list, &reset_list); + + return 0; + +err_notif: + while (i--) + regulator_unregister_notifier(aic3x->supplies[i].consumer, + &aic3x->disable_nb[i].nb); + regulator_bulk_free(ARRAY_SIZE(aic3x->supplies), aic3x->supplies); +err_get: + if (gpio_is_valid(aic3x->gpio_reset) && + !aic3x_is_shared_reset(aic3x)) + gpio_free(aic3x->gpio_reset); +err_gpio: + return ret; +} + +static int aic3x_remove(struct snd_soc_codec *codec) +{ + struct aic3x_priv *aic3x = snd_soc_codec_get_drvdata(codec); + int i; + + aic3x_set_bias_level(codec, SND_SOC_BIAS_OFF); + list_del(&aic3x->list); + if (gpio_is_valid(aic3x->gpio_reset) && + !aic3x_is_shared_reset(aic3x)) { + gpio_set_value(aic3x->gpio_reset, 0); + gpio_free(aic3x->gpio_reset); + } + for (i = 0; i < ARRAY_SIZE(aic3x->supplies); i++) + regulator_unregister_notifier(aic3x->supplies[i].consumer, + &aic3x->disable_nb[i].nb); + regulator_bulk_free(ARRAY_SIZE(aic3x->supplies), aic3x->supplies); + + return 0; +} + +static struct snd_soc_codec_driver soc_codec_dev_aic3x = { + .set_bias_level = aic3x_set_bias_level, + .idle_bias_off = true, + .reg_cache_size = ARRAY_SIZE(aic3x_reg), + .reg_word_size = sizeof(u8), + .reg_cache_default = aic3x_reg, + .probe = aic3x_probe, + .remove = aic3x_remove, + .suspend = aic3x_suspend, + .resume = aic3x_resume, +}; + +/* + * AIC3X 2 wire address can be up to 4 devices with device addresses + * 0x18, 0x19, 0x1A, 0x1B + */ + +static const struct i2c_device_id aic3x_i2c_id[] = { + { "tlv320aic3x", AIC3X_MODEL_3X }, + { "tlv320aic33", AIC3X_MODEL_33 }, + { "tlv320aic3007", AIC3X_MODEL_3007 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, aic3x_i2c_id); + +/* + * If the i2c layer weren't so broken, we could pass this kind of data + * around + */ +static int aic3x_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct aic3x_pdata *pdata = i2c->dev.platform_data; + struct aic3x_priv *aic3x; + int ret; + + aic3x = devm_kzalloc(&i2c->dev, sizeof(struct aic3x_priv), GFP_KERNEL); + if (aic3x == NULL) { + dev_err(&i2c->dev, "failed to create private data\n"); + return -ENOMEM; + } + + aic3x->control_type = SND_SOC_I2C; + + i2c_set_clientdata(i2c, aic3x); + if (pdata) { + aic3x->gpio_reset = pdata->gpio_reset; + aic3x->setup = pdata->setup; + } else { + aic3x->gpio_reset = -1; + } + + aic3x->model = id->driver_data; + + ret = snd_soc_register_codec(&i2c->dev, + &soc_codec_dev_aic3x, &aic3x_dai, 1); + return ret; +} + +static int aic3x_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + return 0; +} + +/* machine i2c codec control layer */ +static struct i2c_driver aic3x_i2c_driver = { + .driver = { + .name = "tlv320aic3x-codec", + .owner = THIS_MODULE, + }, + .probe = aic3x_i2c_probe, + .remove = aic3x_i2c_remove, + .id_table = aic3x_i2c_id, +}; + +static int __init aic3x_modinit(void) +{ + int ret = 0; + ret = i2c_add_driver(&aic3x_i2c_driver); + if (ret != 0) { + printk(KERN_ERR "Failed to register TLV320AIC3x I2C driver: %d\n", + ret); + } + return ret; +} +module_init(aic3x_modinit); + +static void __exit aic3x_exit(void) +{ + i2c_del_driver(&aic3x_i2c_driver); +} +module_exit(aic3x_exit); + +MODULE_DESCRIPTION("ASoC TLV320AIC3X codec driver"); +MODULE_AUTHOR("Vladimir Barinov"); +MODULE_LICENSE("GPL"); |