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alc5623.c

/*
 * alc5623.c  --  alc562[123] ALSA Soc Audio driver
 *
 * Copyright 2008 Realtek Microelectronics
 * Author: flove <flove@realtek.com> Ethan <eku@marvell.com>
 *
 * Copyright 2010 Arnaud Patard <arnaud.patard@rtp-net.org>
 *
 *
 * Based on WM8753.c
 *
 * This program is free software; you can redistribute it and/or modify
 * it under the terms of the GNU General Public License version 2 as
 * published by the Free Software Foundation.
 *
 */

#include <linux/module.h>
#include <linux/kernel.h>
#include <linux/init.h>
#include <linux/delay.h>
#include <linux/pm.h>
#include <linux/i2c.h>
#include <linux/slab.h>
#include <linux/platform_device.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
#include <sound/tlv.h>
#include <sound/soc.h>
#include <sound/initval.h>
#include <sound/alc5623.h>

#include "alc5623.h"

static int caps_charge = 2000;
module_param(caps_charge, int, 0);
MODULE_PARM_DESC(caps_charge, "ALC5623 cap charge time (msecs)");

/* codec private data */
00041 struct alc5623_priv {
      enum snd_soc_control_type control_type;
      void *control_data;
      struct mutex mutex;
      u8 id;
      unsigned int sysclk;
      u16 reg_cache[ALC5623_VENDOR_ID2+2];
      unsigned int add_ctrl;
      unsigned int jack_det_ctrl;
};

static void alc5623_fill_cache(struct snd_soc_codec *codec)
{
      int i, step = codec->driver->reg_cache_step;
      u16 *cache = codec->reg_cache;

      /* not really efficient ... */
      for (i = 0 ; i < codec->driver->reg_cache_size ; i += step)
            cache[i] = codec->hw_read(codec, i);
}

static inline int alc5623_reset(struct snd_soc_codec *codec)
{
      return snd_soc_write(codec, ALC5623_RESET, 0);
}

static int amp_mixer_event(struct snd_soc_dapm_widget *w,
      struct snd_kcontrol *kcontrol, int event)
{
      /* to power-on/off class-d amp generators/speaker */
      /* need to write to 'index-46h' register :        */
      /* so write index num (here 0x46) to reg 0x6a     */
      /* and then 0xffff/0 to reg 0x6c                  */
      snd_soc_write(w->codec, ALC5623_HID_CTRL_INDEX, 0x46);

      switch (event) {
      case SND_SOC_DAPM_PRE_PMU:
            snd_soc_write(w->codec, ALC5623_HID_CTRL_DATA, 0xFFFF);
            break;
      case SND_SOC_DAPM_POST_PMD:
            snd_soc_write(w->codec, ALC5623_HID_CTRL_DATA, 0);
            break;
      }

      return 0;
}

/*
 * ALC5623 Controls
 */

static const DECLARE_TLV_DB_SCALE(vol_tlv, -3450, 150, 0);
static const DECLARE_TLV_DB_SCALE(hp_tlv, -4650, 150, 0);
static const DECLARE_TLV_DB_SCALE(adc_rec_tlv, -1650, 150, 0);
static const unsigned int boost_tlv[] = {
      TLV_DB_RANGE_HEAD(3),
      0, 0, TLV_DB_SCALE_ITEM(0, 0, 0),
      1, 1, TLV_DB_SCALE_ITEM(2000, 0, 0),
      2, 2, TLV_DB_SCALE_ITEM(3000, 0, 0),
};
static const DECLARE_TLV_DB_SCALE(dig_tlv, 0, 600, 0);

static const struct snd_kcontrol_new rt5621_vol_snd_controls[] = {
      SOC_DOUBLE_TLV("Speaker Playback Volume",
                  ALC5623_SPK_OUT_VOL, 8, 0, 31, 1, hp_tlv),
      SOC_DOUBLE("Speaker Playback Switch",
                  ALC5623_SPK_OUT_VOL, 15, 7, 1, 1),
      SOC_DOUBLE_TLV("Headphone Playback Volume",
                  ALC5623_HP_OUT_VOL, 8, 0, 31, 1, hp_tlv),
      SOC_DOUBLE("Headphone Playback Switch",
                  ALC5623_HP_OUT_VOL, 15, 7, 1, 1),
};

static const struct snd_kcontrol_new rt5622_vol_snd_controls[] = {
      SOC_DOUBLE_TLV("Speaker Playback Volume",
                  ALC5623_SPK_OUT_VOL, 8, 0, 31, 1, hp_tlv),
      SOC_DOUBLE("Speaker Playback Switch",
                  ALC5623_SPK_OUT_VOL, 15, 7, 1, 1),
      SOC_DOUBLE_TLV("Line Playback Volume",
                  ALC5623_HP_OUT_VOL, 8, 0, 31, 1, hp_tlv),
      SOC_DOUBLE("Line Playback Switch",
                  ALC5623_HP_OUT_VOL, 15, 7, 1, 1),
};

static const struct snd_kcontrol_new alc5623_vol_snd_controls[] = {
      SOC_DOUBLE_TLV("Line Playback Volume",
                  ALC5623_SPK_OUT_VOL, 8, 0, 31, 1, hp_tlv),
      SOC_DOUBLE("Line Playback Switch",
                  ALC5623_SPK_OUT_VOL, 15, 7, 1, 1),
      SOC_DOUBLE_TLV("Headphone Playback Volume",
                  ALC5623_HP_OUT_VOL, 8, 0, 31, 1, hp_tlv),
      SOC_DOUBLE("Headphone Playback Switch",
                  ALC5623_HP_OUT_VOL, 15, 7, 1, 1),
};

static const struct snd_kcontrol_new alc5623_snd_controls[] = {
      SOC_DOUBLE_TLV("Auxout Playback Volume",
                  ALC5623_MONO_AUX_OUT_VOL, 8, 0, 31, 1, hp_tlv),
      SOC_DOUBLE("Auxout Playback Switch",
                  ALC5623_MONO_AUX_OUT_VOL, 15, 7, 1, 1),
      SOC_DOUBLE_TLV("PCM Playback Volume",
                  ALC5623_STEREO_DAC_VOL, 8, 0, 31, 1, vol_tlv),
      SOC_DOUBLE_TLV("AuxI Capture Volume",
                  ALC5623_AUXIN_VOL, 8, 0, 31, 1, vol_tlv),
      SOC_DOUBLE_TLV("LineIn Capture Volume",
                  ALC5623_LINE_IN_VOL, 8, 0, 31, 1, vol_tlv),
      SOC_SINGLE_TLV("Mic1 Capture Volume",
                  ALC5623_MIC_VOL, 8, 31, 1, vol_tlv),
      SOC_SINGLE_TLV("Mic2 Capture Volume",
                  ALC5623_MIC_VOL, 0, 31, 1, vol_tlv),
      SOC_DOUBLE_TLV("Rec Capture Volume",
                  ALC5623_ADC_REC_GAIN, 7, 0, 31, 0, adc_rec_tlv),
      SOC_SINGLE_TLV("Mic 1 Boost Volume",
                  ALC5623_MIC_CTRL, 10, 2, 0, boost_tlv),
      SOC_SINGLE_TLV("Mic 2 Boost Volume",
                  ALC5623_MIC_CTRL, 8, 2, 0, boost_tlv),
      SOC_SINGLE_TLV("Digital Boost Volume",
                  ALC5623_ADD_CTRL_REG, 4, 3, 0, dig_tlv),
};

/*
 * DAPM Controls
 */
static const struct snd_kcontrol_new alc5623_hp_mixer_controls[] = {
SOC_DAPM_SINGLE("LI2HP Playback Switch", ALC5623_LINE_IN_VOL, 15, 1, 1),
SOC_DAPM_SINGLE("AUXI2HP Playback Switch", ALC5623_AUXIN_VOL, 15, 1, 1),
SOC_DAPM_SINGLE("MIC12HP Playback Switch", ALC5623_MIC_ROUTING_CTRL, 15, 1, 1),
SOC_DAPM_SINGLE("MIC22HP Playback Switch", ALC5623_MIC_ROUTING_CTRL, 7, 1, 1),
SOC_DAPM_SINGLE("DAC2HP Playback Switch", ALC5623_STEREO_DAC_VOL, 15, 1, 1),
};

static const struct snd_kcontrol_new alc5623_hpl_mixer_controls[] = {
SOC_DAPM_SINGLE("ADC2HP_L Playback Switch", ALC5623_ADC_REC_GAIN, 15, 1, 1),
};

static const struct snd_kcontrol_new alc5623_hpr_mixer_controls[] = {
SOC_DAPM_SINGLE("ADC2HP_R Playback Switch", ALC5623_ADC_REC_GAIN, 14, 1, 1),
};

static const struct snd_kcontrol_new alc5623_mono_mixer_controls[] = {
SOC_DAPM_SINGLE("ADC2MONO_L Playback Switch", ALC5623_ADC_REC_GAIN, 13, 1, 1),
SOC_DAPM_SINGLE("ADC2MONO_R Playback Switch", ALC5623_ADC_REC_GAIN, 12, 1, 1),
SOC_DAPM_SINGLE("LI2MONO Playback Switch", ALC5623_LINE_IN_VOL, 13, 1, 1),
SOC_DAPM_SINGLE("AUXI2MONO Playback Switch", ALC5623_AUXIN_VOL, 13, 1, 1),
SOC_DAPM_SINGLE("MIC12MONO Playback Switch", ALC5623_MIC_ROUTING_CTRL, 13, 1, 1),
SOC_DAPM_SINGLE("MIC22MONO Playback Switch", ALC5623_MIC_ROUTING_CTRL, 5, 1, 1),
SOC_DAPM_SINGLE("DAC2MONO Playback Switch", ALC5623_STEREO_DAC_VOL, 13, 1, 1),
};

static const struct snd_kcontrol_new alc5623_speaker_mixer_controls[] = {
SOC_DAPM_SINGLE("LI2SPK Playback Switch", ALC5623_LINE_IN_VOL, 14, 1, 1),
SOC_DAPM_SINGLE("AUXI2SPK Playback Switch", ALC5623_AUXIN_VOL, 14, 1, 1),
SOC_DAPM_SINGLE("MIC12SPK Playback Switch", ALC5623_MIC_ROUTING_CTRL, 14, 1, 1),
SOC_DAPM_SINGLE("MIC22SPK Playback Switch", ALC5623_MIC_ROUTING_CTRL, 6, 1, 1),
SOC_DAPM_SINGLE("DAC2SPK Playback Switch", ALC5623_STEREO_DAC_VOL, 14, 1, 1),
};

/* Left Record Mixer */
static const struct snd_kcontrol_new alc5623_captureL_mixer_controls[] = {
SOC_DAPM_SINGLE("Mic1 Capture Switch", ALC5623_ADC_REC_MIXER, 14, 1, 1),
SOC_DAPM_SINGLE("Mic2 Capture Switch", ALC5623_ADC_REC_MIXER, 13, 1, 1),
SOC_DAPM_SINGLE("LineInL Capture Switch", ALC5623_ADC_REC_MIXER, 12, 1, 1),
SOC_DAPM_SINGLE("Left AuxI Capture Switch", ALC5623_ADC_REC_MIXER, 11, 1, 1),
SOC_DAPM_SINGLE("HPMixerL Capture Switch", ALC5623_ADC_REC_MIXER, 10, 1, 1),
SOC_DAPM_SINGLE("SPKMixer Capture Switch", ALC5623_ADC_REC_MIXER, 9, 1, 1),
SOC_DAPM_SINGLE("MonoMixer Capture Switch", ALC5623_ADC_REC_MIXER, 8, 1, 1),
};

/* Right Record Mixer */
static const struct snd_kcontrol_new alc5623_captureR_mixer_controls[] = {
SOC_DAPM_SINGLE("Mic1 Capture Switch", ALC5623_ADC_REC_MIXER, 6, 1, 1),
SOC_DAPM_SINGLE("Mic2 Capture Switch", ALC5623_ADC_REC_MIXER, 5, 1, 1),
SOC_DAPM_SINGLE("LineInR Capture Switch", ALC5623_ADC_REC_MIXER, 4, 1, 1),
SOC_DAPM_SINGLE("Right AuxI Capture Switch", ALC5623_ADC_REC_MIXER, 3, 1, 1),
SOC_DAPM_SINGLE("HPMixerR Capture Switch", ALC5623_ADC_REC_MIXER, 2, 1, 1),
SOC_DAPM_SINGLE("SPKMixer Capture Switch", ALC5623_ADC_REC_MIXER, 1, 1, 1),
SOC_DAPM_SINGLE("MonoMixer Capture Switch", ALC5623_ADC_REC_MIXER, 0, 1, 1),
};

static const char *alc5623_spk_n_sour_sel[] = {
            "RN/-R", "RP/+R", "LN/-R", "Vmid" };
static const char *alc5623_hpl_out_input_sel[] = {
            "Vmid", "HP Left Mix"};
static const char *alc5623_hpr_out_input_sel[] = {
            "Vmid", "HP Right Mix"};
static const char *alc5623_spkout_input_sel[] = {
            "Vmid", "HPOut Mix", "Speaker Mix", "Mono Mix"};
static const char *alc5623_aux_out_input_sel[] = {
            "Vmid", "HPOut Mix", "Speaker Mix", "Mono Mix"};

/* auxout output mux */
static const struct soc_enum alc5623_aux_out_input_enum =
SOC_ENUM_SINGLE(ALC5623_OUTPUT_MIXER_CTRL, 6, 4, alc5623_aux_out_input_sel);
static const struct snd_kcontrol_new alc5623_auxout_mux_controls =
SOC_DAPM_ENUM("Route", alc5623_aux_out_input_enum);

/* speaker output mux */
static const struct soc_enum alc5623_spkout_input_enum =
SOC_ENUM_SINGLE(ALC5623_OUTPUT_MIXER_CTRL, 10, 4, alc5623_spkout_input_sel);
static const struct snd_kcontrol_new alc5623_spkout_mux_controls =
SOC_DAPM_ENUM("Route", alc5623_spkout_input_enum);

/* headphone left output mux */
static const struct soc_enum alc5623_hpl_out_input_enum =
SOC_ENUM_SINGLE(ALC5623_OUTPUT_MIXER_CTRL, 9, 2, alc5623_hpl_out_input_sel);
static const struct snd_kcontrol_new alc5623_hpl_out_mux_controls =
SOC_DAPM_ENUM("Route", alc5623_hpl_out_input_enum);

/* headphone right output mux */
static const struct soc_enum alc5623_hpr_out_input_enum =
SOC_ENUM_SINGLE(ALC5623_OUTPUT_MIXER_CTRL, 8, 2, alc5623_hpr_out_input_sel);
static const struct snd_kcontrol_new alc5623_hpr_out_mux_controls =
SOC_DAPM_ENUM("Route", alc5623_hpr_out_input_enum);

/* speaker output N select */
static const struct soc_enum alc5623_spk_n_sour_enum =
SOC_ENUM_SINGLE(ALC5623_OUTPUT_MIXER_CTRL, 14, 4, alc5623_spk_n_sour_sel);
static const struct snd_kcontrol_new alc5623_spkoutn_mux_controls =
SOC_DAPM_ENUM("Route", alc5623_spk_n_sour_enum);

static const struct snd_soc_dapm_widget alc5623_dapm_widgets[] = {
/* Muxes */
SND_SOC_DAPM_MUX("AuxOut Mux", SND_SOC_NOPM, 0, 0,
      &alc5623_auxout_mux_controls),
SND_SOC_DAPM_MUX("SpeakerOut Mux", SND_SOC_NOPM, 0, 0,
      &alc5623_spkout_mux_controls),
SND_SOC_DAPM_MUX("Left Headphone Mux", SND_SOC_NOPM, 0, 0,
      &alc5623_hpl_out_mux_controls),
SND_SOC_DAPM_MUX("Right Headphone Mux", SND_SOC_NOPM, 0, 0,
      &alc5623_hpr_out_mux_controls),
SND_SOC_DAPM_MUX("SpeakerOut N Mux", SND_SOC_NOPM, 0, 0,
      &alc5623_spkoutn_mux_controls),

/* output mixers */
SND_SOC_DAPM_MIXER("HP Mix", SND_SOC_NOPM, 0, 0,
      &alc5623_hp_mixer_controls[0],
      ARRAY_SIZE(alc5623_hp_mixer_controls)),
SND_SOC_DAPM_MIXER("HPR Mix", ALC5623_PWR_MANAG_ADD2, 4, 0,
      &alc5623_hpr_mixer_controls[0],
      ARRAY_SIZE(alc5623_hpr_mixer_controls)),
SND_SOC_DAPM_MIXER("HPL Mix", ALC5623_PWR_MANAG_ADD2, 5, 0,
      &alc5623_hpl_mixer_controls[0],
      ARRAY_SIZE(alc5623_hpl_mixer_controls)),
SND_SOC_DAPM_MIXER("HPOut Mix", SND_SOC_NOPM, 0, 0, NULL, 0),
SND_SOC_DAPM_MIXER("Mono Mix", ALC5623_PWR_MANAG_ADD2, 2, 0,
      &alc5623_mono_mixer_controls[0],
      ARRAY_SIZE(alc5623_mono_mixer_controls)),
SND_SOC_DAPM_MIXER("Speaker Mix", ALC5623_PWR_MANAG_ADD2, 3, 0,
      &alc5623_speaker_mixer_controls[0],
      ARRAY_SIZE(alc5623_speaker_mixer_controls)),

/* input mixers */
SND_SOC_DAPM_MIXER("Left Capture Mix", ALC5623_PWR_MANAG_ADD2, 1, 0,
      &alc5623_captureL_mixer_controls[0],
      ARRAY_SIZE(alc5623_captureL_mixer_controls)),
SND_SOC_DAPM_MIXER("Right Capture Mix", ALC5623_PWR_MANAG_ADD2, 0, 0,
      &alc5623_captureR_mixer_controls[0],
      ARRAY_SIZE(alc5623_captureR_mixer_controls)),

SND_SOC_DAPM_DAC("Left DAC", "Left HiFi Playback",
      ALC5623_PWR_MANAG_ADD2, 9, 0),
SND_SOC_DAPM_DAC("Right DAC", "Right HiFi Playback",
      ALC5623_PWR_MANAG_ADD2, 8, 0),
SND_SOC_DAPM_MIXER("I2S Mix", ALC5623_PWR_MANAG_ADD1, 15, 0, NULL, 0),
SND_SOC_DAPM_MIXER("AuxI Mix", SND_SOC_NOPM, 0, 0, NULL, 0),
SND_SOC_DAPM_MIXER("Line Mix", SND_SOC_NOPM, 0, 0, NULL, 0),
SND_SOC_DAPM_ADC("Left ADC", "Left HiFi Capture",
      ALC5623_PWR_MANAG_ADD2, 7, 0),
SND_SOC_DAPM_ADC("Right ADC", "Right HiFi Capture",
      ALC5623_PWR_MANAG_ADD2, 6, 0),
SND_SOC_DAPM_PGA("Left Headphone", ALC5623_PWR_MANAG_ADD3, 10, 0, NULL, 0),
SND_SOC_DAPM_PGA("Right Headphone", ALC5623_PWR_MANAG_ADD3, 9, 0, NULL, 0),
SND_SOC_DAPM_PGA("SpeakerOut", ALC5623_PWR_MANAG_ADD3, 12, 0, NULL, 0),
SND_SOC_DAPM_PGA("Left AuxOut", ALC5623_PWR_MANAG_ADD3, 14, 0, NULL, 0),
SND_SOC_DAPM_PGA("Right AuxOut", ALC5623_PWR_MANAG_ADD3, 13, 0, NULL, 0),
SND_SOC_DAPM_PGA("Left LineIn", ALC5623_PWR_MANAG_ADD3, 7, 0, NULL, 0),
SND_SOC_DAPM_PGA("Right LineIn", ALC5623_PWR_MANAG_ADD3, 6, 0, NULL, 0),
SND_SOC_DAPM_PGA("Left AuxI", ALC5623_PWR_MANAG_ADD3, 5, 0, NULL, 0),
SND_SOC_DAPM_PGA("Right AuxI", ALC5623_PWR_MANAG_ADD3, 4, 0, NULL, 0),
SND_SOC_DAPM_PGA("MIC1 PGA", ALC5623_PWR_MANAG_ADD3, 3, 0, NULL, 0),
SND_SOC_DAPM_PGA("MIC2 PGA", ALC5623_PWR_MANAG_ADD3, 2, 0, NULL, 0),
SND_SOC_DAPM_PGA("MIC1 Pre Amp", ALC5623_PWR_MANAG_ADD3, 1, 0, NULL, 0),
SND_SOC_DAPM_PGA("MIC2 Pre Amp", ALC5623_PWR_MANAG_ADD3, 0, 0, NULL, 0),
SND_SOC_DAPM_MICBIAS("Mic Bias1", ALC5623_PWR_MANAG_ADD1, 11, 0),

SND_SOC_DAPM_OUTPUT("AUXOUTL"),
SND_SOC_DAPM_OUTPUT("AUXOUTR"),
SND_SOC_DAPM_OUTPUT("HPL"),
SND_SOC_DAPM_OUTPUT("HPR"),
SND_SOC_DAPM_OUTPUT("SPKOUT"),
SND_SOC_DAPM_OUTPUT("SPKOUTN"),
SND_SOC_DAPM_INPUT("LINEINL"),
SND_SOC_DAPM_INPUT("LINEINR"),
SND_SOC_DAPM_INPUT("AUXINL"),
SND_SOC_DAPM_INPUT("AUXINR"),
SND_SOC_DAPM_INPUT("MIC1"),
SND_SOC_DAPM_INPUT("MIC2"),
SND_SOC_DAPM_VMID("Vmid"),
};

static const char *alc5623_amp_names[] = {"AB Amp", "D Amp"};
static const struct soc_enum alc5623_amp_enum =
      SOC_ENUM_SINGLE(ALC5623_OUTPUT_MIXER_CTRL, 13, 2, alc5623_amp_names);
static const struct snd_kcontrol_new alc5623_amp_mux_controls =
      SOC_DAPM_ENUM("Route", alc5623_amp_enum);

static const struct snd_soc_dapm_widget alc5623_dapm_amp_widgets[] = {
SND_SOC_DAPM_PGA_E("D Amp", ALC5623_PWR_MANAG_ADD2, 14, 0, NULL, 0,
      amp_mixer_event, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
SND_SOC_DAPM_PGA("AB Amp", ALC5623_PWR_MANAG_ADD2, 15, 0, NULL, 0),
SND_SOC_DAPM_MUX("AB-D Amp Mux", SND_SOC_NOPM, 0, 0,
      &alc5623_amp_mux_controls),
};

static const struct snd_soc_dapm_route intercon[] = {
      /* virtual mixer - mixes left & right channels */
      {"I2S Mix", NULL,                   "Left DAC"},
      {"I2S Mix", NULL,                   "Right DAC"},
      {"Line Mix", NULL,                        "Right LineIn"},
      {"Line Mix", NULL,                        "Left LineIn"},
      {"AuxI Mix", NULL,                        "Left AuxI"},
      {"AuxI Mix", NULL,                        "Right AuxI"},
      {"AUXOUTL", NULL,                   "Left AuxOut"},
      {"AUXOUTR", NULL,                   "Right AuxOut"},

      /* HP mixer */
      {"HPL Mix", "ADC2HP_L Playback Switch",         "Left Capture Mix"},
      {"HPL Mix", NULL,                   "HP Mix"},
      {"HPR Mix", "ADC2HP_R Playback Switch",         "Right Capture Mix"},
      {"HPR Mix", NULL,                   "HP Mix"},
      {"HP Mix", "LI2HP Playback Switch",       "Line Mix"},
      {"HP Mix", "AUXI2HP Playback Switch",           "AuxI Mix"},
      {"HP Mix", "MIC12HP Playback Switch",           "MIC1 PGA"},
      {"HP Mix", "MIC22HP Playback Switch",           "MIC2 PGA"},
      {"HP Mix", "DAC2HP Playback Switch",            "I2S Mix"},

      /* speaker mixer */
      {"Speaker Mix", "LI2SPK Playback Switch", "Line Mix"},
      {"Speaker Mix", "AUXI2SPK Playback Switch",     "AuxI Mix"},
      {"Speaker Mix", "MIC12SPK Playback Switch",     "MIC1 PGA"},
      {"Speaker Mix", "MIC22SPK Playback Switch",     "MIC2 PGA"},
      {"Speaker Mix", "DAC2SPK Playback Switch",      "I2S Mix"},

      /* mono mixer */
      {"Mono Mix", "ADC2MONO_L Playback Switch",      "Left Capture Mix"},
      {"Mono Mix", "ADC2MONO_R Playback Switch",      "Right Capture Mix"},
      {"Mono Mix", "LI2MONO Playback Switch",         "Line Mix"},
      {"Mono Mix", "AUXI2MONO Playback Switch", "AuxI Mix"},
      {"Mono Mix", "MIC12MONO Playback Switch", "MIC1 PGA"},
      {"Mono Mix", "MIC22MONO Playback Switch", "MIC2 PGA"},
      {"Mono Mix", "DAC2MONO Playback Switch",  "I2S Mix"},

      /* Left record mixer */
      {"Left Capture Mix", "LineInL Capture Switch",  "LINEINL"},
      {"Left Capture Mix", "Left AuxI Capture Switch", "AUXINL"},
      {"Left Capture Mix", "Mic1 Capture Switch",     "MIC1 Pre Amp"},
      {"Left Capture Mix", "Mic2 Capture Switch",     "MIC2 Pre Amp"},
      {"Left Capture Mix", "HPMixerL Capture Switch", "HPL Mix"},
      {"Left Capture Mix", "SPKMixer Capture Switch", "Speaker Mix"},
      {"Left Capture Mix", "MonoMixer Capture Switch", "Mono Mix"},

      /*Right record mixer */
      {"Right Capture Mix", "LineInR Capture Switch", "LINEINR"},
      {"Right Capture Mix", "Right AuxI Capture Switch",    "AUXINR"},
      {"Right Capture Mix", "Mic1 Capture Switch",    "MIC1 Pre Amp"},
      {"Right Capture Mix", "Mic2 Capture Switch",    "MIC2 Pre Amp"},
      {"Right Capture Mix", "HPMixerR Capture Switch", "HPR Mix"},
      {"Right Capture Mix", "SPKMixer Capture Switch", "Speaker Mix"},
      {"Right Capture Mix", "MonoMixer Capture Switch", "Mono Mix"},

      /* headphone left mux */
      {"Left Headphone Mux", "HP Left Mix",           "HPL Mix"},
      {"Left Headphone Mux", "Vmid",                  "Vmid"},

      /* headphone right mux */
      {"Right Headphone Mux", "HP Right Mix",         "HPR Mix"},
      {"Right Headphone Mux", "Vmid",                 "Vmid"},

      /* speaker out mux */
      {"SpeakerOut Mux", "Vmid",                "Vmid"},
      {"SpeakerOut Mux", "HPOut Mix",                 "HPOut Mix"},
      {"SpeakerOut Mux", "Speaker Mix",         "Speaker Mix"},
      {"SpeakerOut Mux", "Mono Mix",                  "Mono Mix"},

      /* Mono/Aux Out mux */
      {"AuxOut Mux", "Vmid",                    "Vmid"},
      {"AuxOut Mux", "HPOut Mix",               "HPOut Mix"},
      {"AuxOut Mux", "Speaker Mix",             "Speaker Mix"},
      {"AuxOut Mux", "Mono Mix",                "Mono Mix"},

      /* output pga */
      {"HPL", NULL,                             "Left Headphone"},
      {"Left Headphone", NULL,                  "Left Headphone Mux"},
      {"HPR", NULL,                             "Right Headphone"},
      {"Right Headphone", NULL,                 "Right Headphone Mux"},
      {"Left AuxOut", NULL,                     "AuxOut Mux"},
      {"Right AuxOut", NULL,                    "AuxOut Mux"},

      /* input pga */
      {"Left LineIn", NULL,                     "LINEINL"},
      {"Right LineIn", NULL,                    "LINEINR"},
      {"Left AuxI", NULL,                       "AUXINL"},
      {"Right AuxI", NULL,                      "AUXINR"},
      {"MIC1 Pre Amp", NULL,                    "MIC1"},
      {"MIC2 Pre Amp", NULL,                    "MIC2"},
      {"MIC1 PGA", NULL,                        "MIC1 Pre Amp"},
      {"MIC2 PGA", NULL,                        "MIC2 Pre Amp"},

      /* left ADC */
      {"Left ADC", NULL,                        "Left Capture Mix"},

      /* right ADC */
      {"Right ADC", NULL,                       "Right Capture Mix"},

      {"SpeakerOut N Mux", "RN/-R",             "SpeakerOut"},
      {"SpeakerOut N Mux", "RP/+R",             "SpeakerOut"},
      {"SpeakerOut N Mux", "LN/-R",             "SpeakerOut"},
      {"SpeakerOut N Mux", "Vmid",              "Vmid"},

      {"SPKOUT", NULL,                    "SpeakerOut"},
      {"SPKOUTN", NULL,                   "SpeakerOut N Mux"},
};

static const struct snd_soc_dapm_route intercon_spk[] = {
      {"SpeakerOut", NULL,                      "SpeakerOut Mux"},
};

static const struct snd_soc_dapm_route intercon_amp_spk[] = {
      {"AB Amp", NULL,                    "SpeakerOut Mux"},
      {"D Amp", NULL,                           "SpeakerOut Mux"},
      {"AB-D Amp Mux", "AB Amp",                "AB Amp"},
      {"AB-D Amp Mux", "D Amp",                 "D Amp"},
      {"SpeakerOut", NULL,                      "AB-D Amp Mux"},
};

/* PLL divisors */
00477 struct _pll_div {
      u32 pll_in;
      u32 pll_out;
      u16 regvalue;
};

/* Note : pll code from original alc5623 driver. Not sure of how good it is */
/* usefull only for master mode */
static const struct _pll_div codec_master_pll_div[] = {

      {  2048000,  8192000,   0x0ea0},
      {  3686400,  8192000,   0x4e27},
      { 12000000,  8192000,   0x456b},
      { 13000000,  8192000,   0x495f},
      { 13100000,  8192000,   0x0320},
      {  2048000,  11289600,  0xf637},
      {  3686400,  11289600,  0x2f22},
      { 12000000,  11289600,  0x3e2f},
      { 13000000,  11289600,  0x4d5b},
      { 13100000,  11289600,  0x363b},
      {  2048000,  16384000,  0x1ea0},
      {  3686400,  16384000,  0x9e27},
      { 12000000,  16384000,  0x452b},
      { 13000000,  16384000,  0x542f},
      { 13100000,  16384000,  0x03a0},
      {  2048000,  16934400,  0xe625},
      {  3686400,  16934400,  0x9126},
      { 12000000,  16934400,  0x4d2c},
      { 13000000,  16934400,  0x742f},
      { 13100000,  16934400,  0x3c27},
      {  2048000,  22579200,  0x2aa0},
      {  3686400,  22579200,  0x2f20},
      { 12000000,  22579200,  0x7e2f},
      { 13000000,  22579200,  0x742f},
      { 13100000,  22579200,  0x3c27},
      {  2048000,  24576000,  0x2ea0},
      {  3686400,  24576000,  0xee27},
      { 12000000,  24576000,  0x2915},
      { 13000000,  24576000,  0x772e},
      { 13100000,  24576000,  0x0d20},
};

static const struct _pll_div codec_slave_pll_div[] = {

      {  1024000,  16384000,  0x3ea0},
      {  1411200,  22579200,  0x3ea0},
      {  1536000,  24576000,  0x3ea0},
      {  2048000,  16384000,  0x1ea0},
      {  2822400,  22579200,  0x1ea0},
      {  3072000,  24576000,  0x1ea0},

};

static int alc5623_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id,
            int source, unsigned int freq_in, unsigned int freq_out)
{
      int i;
      struct snd_soc_codec *codec = codec_dai->codec;
      int gbl_clk = 0, pll_div = 0;
      u16 reg;

      if (pll_id < ALC5623_PLL_FR_MCLK || pll_id > ALC5623_PLL_FR_BCK)
            return -ENODEV;

      /* Disable PLL power */
      snd_soc_update_bits(codec, ALC5623_PWR_MANAG_ADD2,
                        ALC5623_PWR_ADD2_PLL,
                        0);

      /* pll is not used in slave mode */
      reg = snd_soc_read(codec, ALC5623_DAI_CONTROL);
      if (reg & ALC5623_DAI_SDP_SLAVE_MODE)
            return 0;

      if (!freq_in || !freq_out)
            return 0;

      switch (pll_id) {
      case ALC5623_PLL_FR_MCLK:
            for (i = 0; i < ARRAY_SIZE(codec_master_pll_div); i++) {
                  if (codec_master_pll_div[i].pll_in == freq_in
                     && codec_master_pll_div[i].pll_out == freq_out) {
                        /* PLL source from MCLK */
                        pll_div  = codec_master_pll_div[i].regvalue;
                        break;
                  }
            }
            break;
      case ALC5623_PLL_FR_BCK:
            for (i = 0; i < ARRAY_SIZE(codec_slave_pll_div); i++) {
                  if (codec_slave_pll_div[i].pll_in == freq_in
                     && codec_slave_pll_div[i].pll_out == freq_out) {
                        /* PLL source from Bitclk */
                        gbl_clk = ALC5623_GBL_CLK_PLL_SOUR_SEL_BITCLK;
                        pll_div = codec_slave_pll_div[i].regvalue;
                        break;
                  }
            }
            break;
      default:
            return -EINVAL;
      }

      if (!pll_div)
            return -EINVAL;

      snd_soc_write(codec, ALC5623_GLOBAL_CLK_CTRL_REG, gbl_clk);
      snd_soc_write(codec, ALC5623_PLL_CTRL, pll_div);
      snd_soc_update_bits(codec, ALC5623_PWR_MANAG_ADD2,
                        ALC5623_PWR_ADD2_PLL,
                        ALC5623_PWR_ADD2_PLL);
      gbl_clk |= ALC5623_GBL_CLK_SYS_SOUR_SEL_PLL;
      snd_soc_write(codec, ALC5623_GLOBAL_CLK_CTRL_REG, gbl_clk);

      return 0;
}

00594 struct _coeff_div {
      u16 fs;
      u16 regvalue;
};

/* codec hifi mclk (after PLL) clock divider coefficients */
/* values inspired from column BCLK=32Fs of Appendix A table */
static const struct _coeff_div coeff_div[] = {
      {256*8, 0x3a69},
      {384*8, 0x3c6b},
      {256*4, 0x2a69},
      {384*4, 0x2c6b},
      {256*2, 0x1a69},
      {384*2, 0x1c6b},
      {256*1, 0x0a69},
      {384*1, 0x0c6b},
};

static int get_coeff(struct snd_soc_codec *codec, int rate)
{
      struct alc5623_priv *alc5623 = snd_soc_codec_get_drvdata(codec);
      int i;

      for (i = 0; i < ARRAY_SIZE(coeff_div); i++) {
            if (coeff_div[i].fs * rate == alc5623->sysclk)
                  return i;
      }
      return -EINVAL;
}

/*
 * Clock after PLL and dividers
 */
static int alc5623_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 alc5623_priv *alc5623 = snd_soc_codec_get_drvdata(codec);

      switch (freq) {
      case  8192000:
      case 11289600:
      case 12288000:
      case 16384000:
      case 16934400:
      case 18432000:
      case 22579200:
      case 24576000:
            alc5623->sysclk = freq;
            return 0;
      }
      return -EINVAL;
}

static int alc5623_set_dai_fmt(struct snd_soc_dai *codec_dai,
            unsigned int fmt)
{
      struct snd_soc_codec *codec = codec_dai->codec;
      u16 iface = 0;

      /* set master/slave audio interface */
      switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
      case SND_SOC_DAIFMT_CBM_CFM:
            iface = ALC5623_DAI_SDP_MASTER_MODE;
            break;
      case SND_SOC_DAIFMT_CBS_CFS:
            iface = ALC5623_DAI_SDP_SLAVE_MODE;
            break;
      default:
            return -EINVAL;
      }

      /* interface format */
      switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
      case SND_SOC_DAIFMT_I2S:
            iface |= ALC5623_DAI_I2S_DF_I2S;
            break;
      case SND_SOC_DAIFMT_RIGHT_J:
            iface |= ALC5623_DAI_I2S_DF_RIGHT;
            break;
      case SND_SOC_DAIFMT_LEFT_J:
            iface |= ALC5623_DAI_I2S_DF_LEFT;
            break;
      case SND_SOC_DAIFMT_DSP_A:
            iface |= ALC5623_DAI_I2S_DF_PCM;
            break;
      case SND_SOC_DAIFMT_DSP_B:
            iface |= ALC5623_DAI_I2S_DF_PCM | ALC5623_DAI_I2S_PCM_MODE;
            break;
      default:
            return -EINVAL;
      }

      /* clock inversion */
      switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
      case SND_SOC_DAIFMT_NB_NF:
            break;
      case SND_SOC_DAIFMT_IB_IF:
            iface |= ALC5623_DAI_MAIN_I2S_BCLK_POL_CTRL;
            break;
      case SND_SOC_DAIFMT_IB_NF:
            iface |= ALC5623_DAI_MAIN_I2S_BCLK_POL_CTRL;
            break;
      case SND_SOC_DAIFMT_NB_IF:
            break;
      default:
            return -EINVAL;
      }

      return snd_soc_write(codec, ALC5623_DAI_CONTROL, iface);
}

static int alc5623_pcm_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 alc5623_priv *alc5623 = snd_soc_codec_get_drvdata(codec);
      int coeff, rate;
      u16 iface;

      iface = snd_soc_read(codec, ALC5623_DAI_CONTROL);
      iface &= ~ALC5623_DAI_I2S_DL_MASK;

      /* bit size */
      switch (params_format(params)) {
      case SNDRV_PCM_FORMAT_S16_LE:
            iface |= ALC5623_DAI_I2S_DL_16;
            break;
      case SNDRV_PCM_FORMAT_S20_3LE:
            iface |= ALC5623_DAI_I2S_DL_20;
            break;
      case SNDRV_PCM_FORMAT_S24_LE:
            iface |= ALC5623_DAI_I2S_DL_24;
            break;
      case SNDRV_PCM_FORMAT_S32_LE:
            iface |= ALC5623_DAI_I2S_DL_32;
            break;
      default:
            return -EINVAL;
      }

      /* set iface & srate */
      snd_soc_write(codec, ALC5623_DAI_CONTROL, iface);
      rate = params_rate(params);
      coeff = get_coeff(codec, rate);
      if (coeff < 0)
            return -EINVAL;

      coeff = coeff_div[coeff].regvalue;
      dev_dbg(codec->dev, "%s: sysclk=%d,rate=%d,coeff=0x%04x\n",
            __func__, alc5623->sysclk, rate, coeff);
      snd_soc_write(codec, ALC5623_STEREO_AD_DA_CLK_CTRL, coeff);

      return 0;
}

static int alc5623_mute(struct snd_soc_dai *dai, int mute)
{
      struct snd_soc_codec *codec = dai->codec;
      u16 hp_mute = ALC5623_MISC_M_DAC_L_INPUT | ALC5623_MISC_M_DAC_R_INPUT;
      u16 mute_reg = snd_soc_read(codec, ALC5623_MISC_CTRL) & ~hp_mute;

      if (mute)
            mute_reg |= hp_mute;

      return snd_soc_write(codec, ALC5623_MISC_CTRL, mute_reg);
}

#define ALC5623_ADD2_POWER_EN (ALC5623_PWR_ADD2_VREF \
      | ALC5623_PWR_ADD2_DAC_REF_CIR)

#define ALC5623_ADD3_POWER_EN (ALC5623_PWR_ADD3_MAIN_BIAS \
      | ALC5623_PWR_ADD3_MIC1_BOOST_AD)

#define ALC5623_ADD1_POWER_EN \
      (ALC5623_PWR_ADD1_SHORT_CURR_DET_EN | ALC5623_PWR_ADD1_SOFTGEN_EN \
      | ALC5623_PWR_ADD1_DEPOP_BUF_HP | ALC5623_PWR_ADD1_HP_OUT_AMP \
      | ALC5623_PWR_ADD1_HP_OUT_ENH_AMP)

#define ALC5623_ADD1_POWER_EN_5622 \
      (ALC5623_PWR_ADD1_SHORT_CURR_DET_EN \
      | ALC5623_PWR_ADD1_HP_OUT_AMP)

static void enable_power_depop(struct snd_soc_codec *codec)
{
      struct alc5623_priv *alc5623 = snd_soc_codec_get_drvdata(codec);

      snd_soc_update_bits(codec, ALC5623_PWR_MANAG_ADD1,
                        ALC5623_PWR_ADD1_SOFTGEN_EN,
                        ALC5623_PWR_ADD1_SOFTGEN_EN);

      snd_soc_write(codec, ALC5623_PWR_MANAG_ADD3, ALC5623_ADD3_POWER_EN);

      snd_soc_update_bits(codec, ALC5623_MISC_CTRL,
                        ALC5623_MISC_HP_DEPOP_MODE2_EN,
                        ALC5623_MISC_HP_DEPOP_MODE2_EN);

      msleep(500);

      snd_soc_write(codec, ALC5623_PWR_MANAG_ADD2, ALC5623_ADD2_POWER_EN);

      /* avoid writing '1' into 5622 reserved bits */
      if (alc5623->id == 0x22)
            snd_soc_write(codec, ALC5623_PWR_MANAG_ADD1,
                  ALC5623_ADD1_POWER_EN_5622);
      else
            snd_soc_write(codec, ALC5623_PWR_MANAG_ADD1,
                  ALC5623_ADD1_POWER_EN);

      /* disable HP Depop2 */
      snd_soc_update_bits(codec, ALC5623_MISC_CTRL,
                        ALC5623_MISC_HP_DEPOP_MODE2_EN,
                        0);

}

static int alc5623_set_bias_level(struct snd_soc_codec *codec,
                              enum snd_soc_bias_level level)
{
      switch (level) {
      case SND_SOC_BIAS_ON:
            enable_power_depop(codec);
            break;
      case SND_SOC_BIAS_PREPARE:
            break;
      case SND_SOC_BIAS_STANDBY:
            /* everything off except vref/vmid, */
            snd_soc_write(codec, ALC5623_PWR_MANAG_ADD2,
                        ALC5623_PWR_ADD2_VREF);
            snd_soc_write(codec, ALC5623_PWR_MANAG_ADD3,
                        ALC5623_PWR_ADD3_MAIN_BIAS);
            break;
      case SND_SOC_BIAS_OFF:
            /* everything off, dac mute, inactive */
            snd_soc_write(codec, ALC5623_PWR_MANAG_ADD2, 0);
            snd_soc_write(codec, ALC5623_PWR_MANAG_ADD3, 0);
            snd_soc_write(codec, ALC5623_PWR_MANAG_ADD1, 0);
            break;
      }
      codec->dapm.bias_level = level;
      return 0;
}

#define ALC5623_FORMATS (SNDRV_PCM_FMTBIT_S16_LE \
                  | SNDRV_PCM_FMTBIT_S24_LE \
                  | SNDRV_PCM_FMTBIT_S32_LE)

static struct snd_soc_dai_ops alc5623_dai_ops = {
            .hw_params = alc5623_pcm_hw_params,
            .digital_mute = alc5623_mute,
            .set_fmt = alc5623_set_dai_fmt,
            .set_sysclk = alc5623_set_dai_sysclk,
            .set_pll = alc5623_set_dai_pll,
};

static struct snd_soc_dai_driver alc5623_dai = {
      .name = "alc5623-hifi",
      .playback = {
            .stream_name = "Playback",
            .channels_min = 1,
            .channels_max = 2,
            .rate_min = 8000,
            .rate_max = 48000,
            .rates = SNDRV_PCM_RATE_8000_48000,
            .formats = ALC5623_FORMATS,},
      .capture = {
            .stream_name = "Capture",
            .channels_min = 1,
            .channels_max = 2,
            .rate_min = 8000,
            .rate_max = 48000,
            .rates = SNDRV_PCM_RATE_8000_48000,
            .formats = ALC5623_FORMATS,},

      .ops = &alc5623_dai_ops,
};

static int alc5623_suspend(struct snd_soc_codec *codec, pm_message_t mesg)
{
      alc5623_set_bias_level(codec, SND_SOC_BIAS_OFF);
      return 0;
}

static int alc5623_resume(struct snd_soc_codec *codec)
{
      int i, step = codec->driver->reg_cache_step;
      u16 *cache = codec->reg_cache;

      /* Sync reg_cache with the hardware */
      for (i = 2 ; i < codec->driver->reg_cache_size ; i += step)
            snd_soc_write(codec, i, cache[i]);

      alc5623_set_bias_level(codec, SND_SOC_BIAS_STANDBY);

      /* charge alc5623 caps */
      if (codec->dapm.suspend_bias_level == SND_SOC_BIAS_ON) {
            alc5623_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
            codec->dapm.bias_level = SND_SOC_BIAS_ON;
            alc5623_set_bias_level(codec, codec->dapm.bias_level);
      }

      return 0;
}

static int alc5623_probe(struct snd_soc_codec *codec)
{
      struct alc5623_priv *alc5623 = snd_soc_codec_get_drvdata(codec);
      struct snd_soc_dapm_context *dapm = &codec->dapm;
      int ret;

      ret = snd_soc_codec_set_cache_io(codec, 8, 16, alc5623->control_type);
      if (ret < 0) {
            dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
            return ret;
      }

      alc5623_reset(codec);
      alc5623_fill_cache(codec);

      /* power on device */
      alc5623_set_bias_level(codec, SND_SOC_BIAS_STANDBY);

      if (alc5623->add_ctrl) {
            snd_soc_write(codec, ALC5623_ADD_CTRL_REG,
                        alc5623->add_ctrl);
      }

      if (alc5623->jack_det_ctrl) {
            snd_soc_write(codec, ALC5623_JACK_DET_CTRL,
                        alc5623->jack_det_ctrl);
      }

      switch (alc5623->id) {
      case 0x21:
            snd_soc_add_controls(codec, rt5621_vol_snd_controls,
                  ARRAY_SIZE(rt5621_vol_snd_controls));
            break;
      case 0x22:
            snd_soc_add_controls(codec, rt5622_vol_snd_controls,
                  ARRAY_SIZE(rt5622_vol_snd_controls));
            break;
      case 0x23:
            snd_soc_add_controls(codec, alc5623_vol_snd_controls,
                  ARRAY_SIZE(alc5623_vol_snd_controls));
            break;
      default:
            return -EINVAL;
      }

      snd_soc_add_controls(codec, alc5623_snd_controls,
                  ARRAY_SIZE(alc5623_snd_controls));

      snd_soc_dapm_new_controls(dapm, alc5623_dapm_widgets,
                              ARRAY_SIZE(alc5623_dapm_widgets));

      /* set up audio path interconnects */
      snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon));

      switch (alc5623->id) {
      case 0x21:
      case 0x22:
            snd_soc_dapm_new_controls(dapm, alc5623_dapm_amp_widgets,
                              ARRAY_SIZE(alc5623_dapm_amp_widgets));
            snd_soc_dapm_add_routes(dapm, intercon_amp_spk,
                              ARRAY_SIZE(intercon_amp_spk));
            break;
      case 0x23:
            snd_soc_dapm_add_routes(dapm, intercon_spk,
                              ARRAY_SIZE(intercon_spk));
            break;
      default:
            return -EINVAL;
      }

      return ret;
}

/* power down chip */
static int alc5623_remove(struct snd_soc_codec *codec)
{
      alc5623_set_bias_level(codec, SND_SOC_BIAS_OFF);
      return 0;
}

static struct snd_soc_codec_driver soc_codec_device_alc5623 = {
      .probe = alc5623_probe,
      .remove = alc5623_remove,
      .suspend = alc5623_suspend,
      .resume = alc5623_resume,
      .set_bias_level = alc5623_set_bias_level,
      .reg_cache_size = ALC5623_VENDOR_ID2+2,
      .reg_word_size = sizeof(u16),
      .reg_cache_step = 2,
};

/*
 * ALC5623 2 wire address is determined by A1 pin
 * state during powerup.
 *    low  = 0x1a
 *    high = 0x1b
 */
static int alc5623_i2c_probe(struct i2c_client *client,
                        const struct i2c_device_id *id)
{
      struct alc5623_platform_data *pdata;
      struct alc5623_priv *alc5623;
      int ret, vid1, vid2;

      vid1 = i2c_smbus_read_word_data(client, ALC5623_VENDOR_ID1);
      if (vid1 < 0) {
            dev_err(&client->dev, "failed to read I2C\n");
            return -EIO;
      }
      vid1 = ((vid1 & 0xff) << 8) | (vid1 >> 8);

      vid2 = i2c_smbus_read_byte_data(client, ALC5623_VENDOR_ID2);
      if (vid2 < 0) {
            dev_err(&client->dev, "failed to read I2C\n");
            return -EIO;
      }

      if ((vid1 != 0x10ec) || (vid2 != id->driver_data)) {
            dev_err(&client->dev, "unknown or wrong codec\n");
            dev_err(&client->dev, "Expected %x:%lx, got %x:%x\n",
                        0x10ec, id->driver_data,
                        vid1, vid2);
            return -ENODEV;
      }

      dev_dbg(&client->dev, "Found codec id : alc56%02x\n", vid2);

      alc5623 = kzalloc(sizeof(struct alc5623_priv), GFP_KERNEL);
      if (alc5623 == NULL)
            return -ENOMEM;

      pdata = client->dev.platform_data;
      if (pdata) {
            alc5623->add_ctrl = pdata->add_ctrl;
            alc5623->jack_det_ctrl = pdata->jack_det_ctrl;
      }

      alc5623->id = vid2;
      switch (alc5623->id) {
      case 0x21:
            alc5623_dai.name = "alc5621-hifi";
            break;
      case 0x22:
            alc5623_dai.name = "alc5622-hifi";
            break;
      case 0x23:
            alc5623_dai.name = "alc5623-hifi";
            break;
      default:
            kfree(alc5623);
            return -EINVAL;
      }

      i2c_set_clientdata(client, alc5623);
      alc5623->control_data = client;
      alc5623->control_type = SND_SOC_I2C;
      mutex_init(&alc5623->mutex);

      ret =  snd_soc_register_codec(&client->dev,
            &soc_codec_device_alc5623, &alc5623_dai, 1);
      if (ret != 0) {
            dev_err(&client->dev, "Failed to register codec: %d\n", ret);
            kfree(alc5623);
      }

      return ret;
}

static int alc5623_i2c_remove(struct i2c_client *client)
{
      struct alc5623_priv *alc5623 = i2c_get_clientdata(client);

      snd_soc_unregister_codec(&client->dev);
      kfree(alc5623);
      return 0;
}

static const struct i2c_device_id alc5623_i2c_table[] = {
      {"alc5621", 0x21},
      {"alc5622", 0x22},
      {"alc5623", 0x23},
      {}
};
MODULE_DEVICE_TABLE(i2c, alc5623_i2c_table);

/*  i2c codec control layer */
static struct i2c_driver alc5623_i2c_driver = {
      .driver = {
            .name = "alc562x-codec",
            .owner = THIS_MODULE,
      },
      .probe = alc5623_i2c_probe,
      .remove =  __devexit_p(alc5623_i2c_remove),
      .id_table = alc5623_i2c_table,
};

static int __init alc5623_modinit(void)
{
      int ret;

      ret = i2c_add_driver(&alc5623_i2c_driver);
      if (ret != 0) {
            printk(KERN_ERR "%s: can't add i2c driver", __func__);
            return ret;
      }

      return ret;
}
module_init(alc5623_modinit);

static void __exit alc5623_modexit(void)
{
      i2c_del_driver(&alc5623_i2c_driver);
}
module_exit(alc5623_modexit);

MODULE_DESCRIPTION("ASoC alc5621/2/3 driver");
MODULE_AUTHOR("Arnaud Patard <arnaud.patard@rtp-net.org>");
MODULE_LICENSE("GPL");

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