linux/sound/soc/codecs/alc5623.c
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   1/*
   2 * alc5623.c  --  alc562[123] ALSA Soc Audio driver
   3 *
   4 * Copyright 2008 Realtek Microelectronics
   5 * Author: flove <flove@realtek.com> Ethan <eku@marvell.com>
   6 *
   7 * Copyright 2010 Arnaud Patard <arnaud.patard@rtp-net.org>
   8 *
   9 *
  10 * Based on WM8753.c
  11 *
  12 * This program is free software; you can redistribute it and/or modify
  13 * it under the terms of the GNU General Public License version 2 as
  14 * published by the Free Software Foundation.
  15 *
  16 */
  17
  18#include <linux/module.h>
  19#include <linux/kernel.h>
  20#include <linux/init.h>
  21#include <linux/delay.h>
  22#include <linux/pm.h>
  23#include <linux/i2c.h>
  24#include <linux/slab.h>
  25#include <sound/core.h>
  26#include <sound/pcm.h>
  27#include <sound/pcm_params.h>
  28#include <sound/tlv.h>
  29#include <sound/soc.h>
  30#include <sound/initval.h>
  31#include <sound/alc5623.h>
  32
  33#include "alc5623.h"
  34
  35static int caps_charge = 2000;
  36module_param(caps_charge, int, 0);
  37MODULE_PARM_DESC(caps_charge, "ALC5623 cap charge time (msecs)");
  38
  39/* codec private data */
  40struct alc5623_priv {
  41        enum snd_soc_control_type control_type;
  42        u8 id;
  43        unsigned int sysclk;
  44        u16 reg_cache[ALC5623_VENDOR_ID2+2];
  45        unsigned int add_ctrl;
  46        unsigned int jack_det_ctrl;
  47};
  48
  49static void alc5623_fill_cache(struct snd_soc_codec *codec)
  50{
  51        int i, step = codec->driver->reg_cache_step;
  52        u16 *cache = codec->reg_cache;
  53
  54        /* not really efficient ... */
  55        codec->cache_bypass = 1;
  56        for (i = 0 ; i < codec->driver->reg_cache_size ; i += step)
  57                cache[i] = snd_soc_read(codec, i);
  58        codec->cache_bypass = 0;
  59}
  60
  61static inline int alc5623_reset(struct snd_soc_codec *codec)
  62{
  63        return snd_soc_write(codec, ALC5623_RESET, 0);
  64}
  65
  66static int amp_mixer_event(struct snd_soc_dapm_widget *w,
  67        struct snd_kcontrol *kcontrol, int event)
  68{
  69        /* to power-on/off class-d amp generators/speaker */
  70        /* need to write to 'index-46h' register :        */
  71        /* so write index num (here 0x46) to reg 0x6a     */
  72        /* and then 0xffff/0 to reg 0x6c                  */
  73        snd_soc_write(w->codec, ALC5623_HID_CTRL_INDEX, 0x46);
  74
  75        switch (event) {
  76        case SND_SOC_DAPM_PRE_PMU:
  77                snd_soc_write(w->codec, ALC5623_HID_CTRL_DATA, 0xFFFF);
  78                break;
  79        case SND_SOC_DAPM_POST_PMD:
  80                snd_soc_write(w->codec, ALC5623_HID_CTRL_DATA, 0);
  81                break;
  82        }
  83
  84        return 0;
  85}
  86
  87/*
  88 * ALC5623 Controls
  89 */
  90
  91static const DECLARE_TLV_DB_SCALE(vol_tlv, -3450, 150, 0);
  92static const DECLARE_TLV_DB_SCALE(hp_tlv, -4650, 150, 0);
  93static const DECLARE_TLV_DB_SCALE(adc_rec_tlv, -1650, 150, 0);
  94static const unsigned int boost_tlv[] = {
  95        TLV_DB_RANGE_HEAD(3),
  96        0, 0, TLV_DB_SCALE_ITEM(0, 0, 0),
  97        1, 1, TLV_DB_SCALE_ITEM(2000, 0, 0),
  98        2, 2, TLV_DB_SCALE_ITEM(3000, 0, 0),
  99};
 100static const DECLARE_TLV_DB_SCALE(dig_tlv, 0, 600, 0);
 101
 102static const struct snd_kcontrol_new alc5621_vol_snd_controls[] = {
 103        SOC_DOUBLE_TLV("Speaker Playback Volume",
 104                        ALC5623_SPK_OUT_VOL, 8, 0, 31, 1, hp_tlv),
 105        SOC_DOUBLE("Speaker Playback Switch",
 106                        ALC5623_SPK_OUT_VOL, 15, 7, 1, 1),
 107        SOC_DOUBLE_TLV("Headphone Playback Volume",
 108                        ALC5623_HP_OUT_VOL, 8, 0, 31, 1, hp_tlv),
 109        SOC_DOUBLE("Headphone Playback Switch",
 110                        ALC5623_HP_OUT_VOL, 15, 7, 1, 1),
 111};
 112
 113static const struct snd_kcontrol_new alc5622_vol_snd_controls[] = {
 114        SOC_DOUBLE_TLV("Speaker Playback Volume",
 115                        ALC5623_SPK_OUT_VOL, 8, 0, 31, 1, hp_tlv),
 116        SOC_DOUBLE("Speaker Playback Switch",
 117                        ALC5623_SPK_OUT_VOL, 15, 7, 1, 1),
 118        SOC_DOUBLE_TLV("Line Playback Volume",
 119                        ALC5623_HP_OUT_VOL, 8, 0, 31, 1, hp_tlv),
 120        SOC_DOUBLE("Line Playback Switch",
 121                        ALC5623_HP_OUT_VOL, 15, 7, 1, 1),
 122};
 123
 124static const struct snd_kcontrol_new alc5623_vol_snd_controls[] = {
 125        SOC_DOUBLE_TLV("Line Playback Volume",
 126                        ALC5623_SPK_OUT_VOL, 8, 0, 31, 1, hp_tlv),
 127        SOC_DOUBLE("Line Playback Switch",
 128                        ALC5623_SPK_OUT_VOL, 15, 7, 1, 1),
 129        SOC_DOUBLE_TLV("Headphone Playback Volume",
 130                        ALC5623_HP_OUT_VOL, 8, 0, 31, 1, hp_tlv),
 131        SOC_DOUBLE("Headphone Playback Switch",
 132                        ALC5623_HP_OUT_VOL, 15, 7, 1, 1),
 133};
 134
 135static const struct snd_kcontrol_new alc5623_snd_controls[] = {
 136        SOC_DOUBLE_TLV("Auxout Playback Volume",
 137                        ALC5623_MONO_AUX_OUT_VOL, 8, 0, 31, 1, hp_tlv),
 138        SOC_DOUBLE("Auxout Playback Switch",
 139                        ALC5623_MONO_AUX_OUT_VOL, 15, 7, 1, 1),
 140        SOC_DOUBLE_TLV("PCM Playback Volume",
 141                        ALC5623_STEREO_DAC_VOL, 8, 0, 31, 1, vol_tlv),
 142        SOC_DOUBLE_TLV("AuxI Capture Volume",
 143                        ALC5623_AUXIN_VOL, 8, 0, 31, 1, vol_tlv),
 144        SOC_DOUBLE_TLV("LineIn Capture Volume",
 145                        ALC5623_LINE_IN_VOL, 8, 0, 31, 1, vol_tlv),
 146        SOC_SINGLE_TLV("Mic1 Capture Volume",
 147                        ALC5623_MIC_VOL, 8, 31, 1, vol_tlv),
 148        SOC_SINGLE_TLV("Mic2 Capture Volume",
 149                        ALC5623_MIC_VOL, 0, 31, 1, vol_tlv),
 150        SOC_DOUBLE_TLV("Rec Capture Volume",
 151                        ALC5623_ADC_REC_GAIN, 7, 0, 31, 0, adc_rec_tlv),
 152        SOC_SINGLE_TLV("Mic 1 Boost Volume",
 153                        ALC5623_MIC_CTRL, 10, 2, 0, boost_tlv),
 154        SOC_SINGLE_TLV("Mic 2 Boost Volume",
 155                        ALC5623_MIC_CTRL, 8, 2, 0, boost_tlv),
 156        SOC_SINGLE_TLV("Digital Boost Volume",
 157                        ALC5623_ADD_CTRL_REG, 4, 3, 0, dig_tlv),
 158};
 159
 160/*
 161 * DAPM Controls
 162 */
 163static const struct snd_kcontrol_new alc5623_hp_mixer_controls[] = {
 164SOC_DAPM_SINGLE("LI2HP Playback Switch", ALC5623_LINE_IN_VOL, 15, 1, 1),
 165SOC_DAPM_SINGLE("AUXI2HP Playback Switch", ALC5623_AUXIN_VOL, 15, 1, 1),
 166SOC_DAPM_SINGLE("MIC12HP Playback Switch", ALC5623_MIC_ROUTING_CTRL, 15, 1, 1),
 167SOC_DAPM_SINGLE("MIC22HP Playback Switch", ALC5623_MIC_ROUTING_CTRL, 7, 1, 1),
 168SOC_DAPM_SINGLE("DAC2HP Playback Switch", ALC5623_STEREO_DAC_VOL, 15, 1, 1),
 169};
 170
 171static const struct snd_kcontrol_new alc5623_hpl_mixer_controls[] = {
 172SOC_DAPM_SINGLE("ADC2HP_L Playback Switch", ALC5623_ADC_REC_GAIN, 15, 1, 1),
 173};
 174
 175static const struct snd_kcontrol_new alc5623_hpr_mixer_controls[] = {
 176SOC_DAPM_SINGLE("ADC2HP_R Playback Switch", ALC5623_ADC_REC_GAIN, 14, 1, 1),
 177};
 178
 179static const struct snd_kcontrol_new alc5623_mono_mixer_controls[] = {
 180SOC_DAPM_SINGLE("ADC2MONO_L Playback Switch", ALC5623_ADC_REC_GAIN, 13, 1, 1),
 181SOC_DAPM_SINGLE("ADC2MONO_R Playback Switch", ALC5623_ADC_REC_GAIN, 12, 1, 1),
 182SOC_DAPM_SINGLE("LI2MONO Playback Switch", ALC5623_LINE_IN_VOL, 13, 1, 1),
 183SOC_DAPM_SINGLE("AUXI2MONO Playback Switch", ALC5623_AUXIN_VOL, 13, 1, 1),
 184SOC_DAPM_SINGLE("MIC12MONO Playback Switch", ALC5623_MIC_ROUTING_CTRL, 13, 1, 1),
 185SOC_DAPM_SINGLE("MIC22MONO Playback Switch", ALC5623_MIC_ROUTING_CTRL, 5, 1, 1),
 186SOC_DAPM_SINGLE("DAC2MONO Playback Switch", ALC5623_STEREO_DAC_VOL, 13, 1, 1),
 187};
 188
 189static const struct snd_kcontrol_new alc5623_speaker_mixer_controls[] = {
 190SOC_DAPM_SINGLE("LI2SPK Playback Switch", ALC5623_LINE_IN_VOL, 14, 1, 1),
 191SOC_DAPM_SINGLE("AUXI2SPK Playback Switch", ALC5623_AUXIN_VOL, 14, 1, 1),
 192SOC_DAPM_SINGLE("MIC12SPK Playback Switch", ALC5623_MIC_ROUTING_CTRL, 14, 1, 1),
 193SOC_DAPM_SINGLE("MIC22SPK Playback Switch", ALC5623_MIC_ROUTING_CTRL, 6, 1, 1),
 194SOC_DAPM_SINGLE("DAC2SPK Playback Switch", ALC5623_STEREO_DAC_VOL, 14, 1, 1),
 195};
 196
 197/* Left Record Mixer */
 198static const struct snd_kcontrol_new alc5623_captureL_mixer_controls[] = {
 199SOC_DAPM_SINGLE("Mic1 Capture Switch", ALC5623_ADC_REC_MIXER, 14, 1, 1),
 200SOC_DAPM_SINGLE("Mic2 Capture Switch", ALC5623_ADC_REC_MIXER, 13, 1, 1),
 201SOC_DAPM_SINGLE("LineInL Capture Switch", ALC5623_ADC_REC_MIXER, 12, 1, 1),
 202SOC_DAPM_SINGLE("Left AuxI Capture Switch", ALC5623_ADC_REC_MIXER, 11, 1, 1),
 203SOC_DAPM_SINGLE("HPMixerL Capture Switch", ALC5623_ADC_REC_MIXER, 10, 1, 1),
 204SOC_DAPM_SINGLE("SPKMixer Capture Switch", ALC5623_ADC_REC_MIXER, 9, 1, 1),
 205SOC_DAPM_SINGLE("MonoMixer Capture Switch", ALC5623_ADC_REC_MIXER, 8, 1, 1),
 206};
 207
 208/* Right Record Mixer */
 209static const struct snd_kcontrol_new alc5623_captureR_mixer_controls[] = {
 210SOC_DAPM_SINGLE("Mic1 Capture Switch", ALC5623_ADC_REC_MIXER, 6, 1, 1),
 211SOC_DAPM_SINGLE("Mic2 Capture Switch", ALC5623_ADC_REC_MIXER, 5, 1, 1),
 212SOC_DAPM_SINGLE("LineInR Capture Switch", ALC5623_ADC_REC_MIXER, 4, 1, 1),
 213SOC_DAPM_SINGLE("Right AuxI Capture Switch", ALC5623_ADC_REC_MIXER, 3, 1, 1),
 214SOC_DAPM_SINGLE("HPMixerR Capture Switch", ALC5623_ADC_REC_MIXER, 2, 1, 1),
 215SOC_DAPM_SINGLE("SPKMixer Capture Switch", ALC5623_ADC_REC_MIXER, 1, 1, 1),
 216SOC_DAPM_SINGLE("MonoMixer Capture Switch", ALC5623_ADC_REC_MIXER, 0, 1, 1),
 217};
 218
 219static const char *alc5623_spk_n_sour_sel[] = {
 220                "RN/-R", "RP/+R", "LN/-R", "Vmid" };
 221static const char *alc5623_hpl_out_input_sel[] = {
 222                "Vmid", "HP Left Mix"};
 223static const char *alc5623_hpr_out_input_sel[] = {
 224                "Vmid", "HP Right Mix"};
 225static const char *alc5623_spkout_input_sel[] = {
 226                "Vmid", "HPOut Mix", "Speaker Mix", "Mono Mix"};
 227static const char *alc5623_aux_out_input_sel[] = {
 228                "Vmid", "HPOut Mix", "Speaker Mix", "Mono Mix"};
 229
 230/* auxout output mux */
 231static const struct soc_enum alc5623_aux_out_input_enum =
 232SOC_ENUM_SINGLE(ALC5623_OUTPUT_MIXER_CTRL, 6, 4, alc5623_aux_out_input_sel);
 233static const struct snd_kcontrol_new alc5623_auxout_mux_controls =
 234SOC_DAPM_ENUM("Route", alc5623_aux_out_input_enum);
 235
 236/* speaker output mux */
 237static const struct soc_enum alc5623_spkout_input_enum =
 238SOC_ENUM_SINGLE(ALC5623_OUTPUT_MIXER_CTRL, 10, 4, alc5623_spkout_input_sel);
 239static const struct snd_kcontrol_new alc5623_spkout_mux_controls =
 240SOC_DAPM_ENUM("Route", alc5623_spkout_input_enum);
 241
 242/* headphone left output mux */
 243static const struct soc_enum alc5623_hpl_out_input_enum =
 244SOC_ENUM_SINGLE(ALC5623_OUTPUT_MIXER_CTRL, 9, 2, alc5623_hpl_out_input_sel);
 245static const struct snd_kcontrol_new alc5623_hpl_out_mux_controls =
 246SOC_DAPM_ENUM("Route", alc5623_hpl_out_input_enum);
 247
 248/* headphone right output mux */
 249static const struct soc_enum alc5623_hpr_out_input_enum =
 250SOC_ENUM_SINGLE(ALC5623_OUTPUT_MIXER_CTRL, 8, 2, alc5623_hpr_out_input_sel);
 251static const struct snd_kcontrol_new alc5623_hpr_out_mux_controls =
 252SOC_DAPM_ENUM("Route", alc5623_hpr_out_input_enum);
 253
 254/* speaker output N select */
 255static const struct soc_enum alc5623_spk_n_sour_enum =
 256SOC_ENUM_SINGLE(ALC5623_OUTPUT_MIXER_CTRL, 14, 4, alc5623_spk_n_sour_sel);
 257static const struct snd_kcontrol_new alc5623_spkoutn_mux_controls =
 258SOC_DAPM_ENUM("Route", alc5623_spk_n_sour_enum);
 259
 260static const struct snd_soc_dapm_widget alc5623_dapm_widgets[] = {
 261/* Muxes */
 262SND_SOC_DAPM_MUX("AuxOut Mux", SND_SOC_NOPM, 0, 0,
 263        &alc5623_auxout_mux_controls),
 264SND_SOC_DAPM_MUX("SpeakerOut Mux", SND_SOC_NOPM, 0, 0,
 265        &alc5623_spkout_mux_controls),
 266SND_SOC_DAPM_MUX("Left Headphone Mux", SND_SOC_NOPM, 0, 0,
 267        &alc5623_hpl_out_mux_controls),
 268SND_SOC_DAPM_MUX("Right Headphone Mux", SND_SOC_NOPM, 0, 0,
 269        &alc5623_hpr_out_mux_controls),
 270SND_SOC_DAPM_MUX("SpeakerOut N Mux", SND_SOC_NOPM, 0, 0,
 271        &alc5623_spkoutn_mux_controls),
 272
 273/* output mixers */
 274SND_SOC_DAPM_MIXER("HP Mix", SND_SOC_NOPM, 0, 0,
 275        &alc5623_hp_mixer_controls[0],
 276        ARRAY_SIZE(alc5623_hp_mixer_controls)),
 277SND_SOC_DAPM_MIXER("HPR Mix", ALC5623_PWR_MANAG_ADD2, 4, 0,
 278        &alc5623_hpr_mixer_controls[0],
 279        ARRAY_SIZE(alc5623_hpr_mixer_controls)),
 280SND_SOC_DAPM_MIXER("HPL Mix", ALC5623_PWR_MANAG_ADD2, 5, 0,
 281        &alc5623_hpl_mixer_controls[0],
 282        ARRAY_SIZE(alc5623_hpl_mixer_controls)),
 283SND_SOC_DAPM_MIXER("HPOut Mix", SND_SOC_NOPM, 0, 0, NULL, 0),
 284SND_SOC_DAPM_MIXER("Mono Mix", ALC5623_PWR_MANAG_ADD2, 2, 0,
 285        &alc5623_mono_mixer_controls[0],
 286        ARRAY_SIZE(alc5623_mono_mixer_controls)),
 287SND_SOC_DAPM_MIXER("Speaker Mix", ALC5623_PWR_MANAG_ADD2, 3, 0,
 288        &alc5623_speaker_mixer_controls[0],
 289        ARRAY_SIZE(alc5623_speaker_mixer_controls)),
 290
 291/* input mixers */
 292SND_SOC_DAPM_MIXER("Left Capture Mix", ALC5623_PWR_MANAG_ADD2, 1, 0,
 293        &alc5623_captureL_mixer_controls[0],
 294        ARRAY_SIZE(alc5623_captureL_mixer_controls)),
 295SND_SOC_DAPM_MIXER("Right Capture Mix", ALC5623_PWR_MANAG_ADD2, 0, 0,
 296        &alc5623_captureR_mixer_controls[0],
 297        ARRAY_SIZE(alc5623_captureR_mixer_controls)),
 298
 299SND_SOC_DAPM_DAC("Left DAC", "Left HiFi Playback",
 300        ALC5623_PWR_MANAG_ADD2, 9, 0),
 301SND_SOC_DAPM_DAC("Right DAC", "Right HiFi Playback",
 302        ALC5623_PWR_MANAG_ADD2, 8, 0),
 303SND_SOC_DAPM_MIXER("I2S Mix", ALC5623_PWR_MANAG_ADD1, 15, 0, NULL, 0),
 304SND_SOC_DAPM_MIXER("AuxI Mix", SND_SOC_NOPM, 0, 0, NULL, 0),
 305SND_SOC_DAPM_MIXER("Line Mix", SND_SOC_NOPM, 0, 0, NULL, 0),
 306SND_SOC_DAPM_ADC("Left ADC", "Left HiFi Capture",
 307        ALC5623_PWR_MANAG_ADD2, 7, 0),
 308SND_SOC_DAPM_ADC("Right ADC", "Right HiFi Capture",
 309        ALC5623_PWR_MANAG_ADD2, 6, 0),
 310SND_SOC_DAPM_PGA("Left Headphone", ALC5623_PWR_MANAG_ADD3, 10, 0, NULL, 0),
 311SND_SOC_DAPM_PGA("Right Headphone", ALC5623_PWR_MANAG_ADD3, 9, 0, NULL, 0),
 312SND_SOC_DAPM_PGA("SpeakerOut", ALC5623_PWR_MANAG_ADD3, 12, 0, NULL, 0),
 313SND_SOC_DAPM_PGA("Left AuxOut", ALC5623_PWR_MANAG_ADD3, 14, 0, NULL, 0),
 314SND_SOC_DAPM_PGA("Right AuxOut", ALC5623_PWR_MANAG_ADD3, 13, 0, NULL, 0),
 315SND_SOC_DAPM_PGA("Left LineIn", ALC5623_PWR_MANAG_ADD3, 7, 0, NULL, 0),
 316SND_SOC_DAPM_PGA("Right LineIn", ALC5623_PWR_MANAG_ADD3, 6, 0, NULL, 0),
 317SND_SOC_DAPM_PGA("Left AuxI", ALC5623_PWR_MANAG_ADD3, 5, 0, NULL, 0),
 318SND_SOC_DAPM_PGA("Right AuxI", ALC5623_PWR_MANAG_ADD3, 4, 0, NULL, 0),
 319SND_SOC_DAPM_PGA("MIC1 PGA", ALC5623_PWR_MANAG_ADD3, 3, 0, NULL, 0),
 320SND_SOC_DAPM_PGA("MIC2 PGA", ALC5623_PWR_MANAG_ADD3, 2, 0, NULL, 0),
 321SND_SOC_DAPM_PGA("MIC1 Pre Amp", ALC5623_PWR_MANAG_ADD3, 1, 0, NULL, 0),
 322SND_SOC_DAPM_PGA("MIC2 Pre Amp", ALC5623_PWR_MANAG_ADD3, 0, 0, NULL, 0),
 323SND_SOC_DAPM_MICBIAS("Mic Bias1", ALC5623_PWR_MANAG_ADD1, 11, 0),
 324
 325SND_SOC_DAPM_OUTPUT("AUXOUTL"),
 326SND_SOC_DAPM_OUTPUT("AUXOUTR"),
 327SND_SOC_DAPM_OUTPUT("HPL"),
 328SND_SOC_DAPM_OUTPUT("HPR"),
 329SND_SOC_DAPM_OUTPUT("SPKOUT"),
 330SND_SOC_DAPM_OUTPUT("SPKOUTN"),
 331SND_SOC_DAPM_INPUT("LINEINL"),
 332SND_SOC_DAPM_INPUT("LINEINR"),
 333SND_SOC_DAPM_INPUT("AUXINL"),
 334SND_SOC_DAPM_INPUT("AUXINR"),
 335SND_SOC_DAPM_INPUT("MIC1"),
 336SND_SOC_DAPM_INPUT("MIC2"),
 337SND_SOC_DAPM_VMID("Vmid"),
 338};
 339
 340static const char *alc5623_amp_names[] = {"AB Amp", "D Amp"};
 341static const struct soc_enum alc5623_amp_enum =
 342        SOC_ENUM_SINGLE(ALC5623_OUTPUT_MIXER_CTRL, 13, 2, alc5623_amp_names);
 343static const struct snd_kcontrol_new alc5623_amp_mux_controls =
 344        SOC_DAPM_ENUM("Route", alc5623_amp_enum);
 345
 346static const struct snd_soc_dapm_widget alc5623_dapm_amp_widgets[] = {
 347SND_SOC_DAPM_PGA_E("D Amp", ALC5623_PWR_MANAG_ADD2, 14, 0, NULL, 0,
 348        amp_mixer_event, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
 349SND_SOC_DAPM_PGA("AB Amp", ALC5623_PWR_MANAG_ADD2, 15, 0, NULL, 0),
 350SND_SOC_DAPM_MUX("AB-D Amp Mux", SND_SOC_NOPM, 0, 0,
 351        &alc5623_amp_mux_controls),
 352};
 353
 354static const struct snd_soc_dapm_route intercon[] = {
 355        /* virtual mixer - mixes left & right channels */
 356        {"I2S Mix", NULL,                               "Left DAC"},
 357        {"I2S Mix", NULL,                               "Right DAC"},
 358        {"Line Mix", NULL,                              "Right LineIn"},
 359        {"Line Mix", NULL,                              "Left LineIn"},
 360        {"AuxI Mix", NULL,                              "Left AuxI"},
 361        {"AuxI Mix", NULL,                              "Right AuxI"},
 362        {"AUXOUTL", NULL,                               "Left AuxOut"},
 363        {"AUXOUTR", NULL,                               "Right AuxOut"},
 364
 365        /* HP mixer */
 366        {"HPL Mix", "ADC2HP_L Playback Switch",         "Left Capture Mix"},
 367        {"HPL Mix", NULL,                               "HP Mix"},
 368        {"HPR Mix", "ADC2HP_R Playback Switch",         "Right Capture Mix"},
 369        {"HPR Mix", NULL,                               "HP Mix"},
 370        {"HP Mix", "LI2HP Playback Switch",             "Line Mix"},
 371        {"HP Mix", "AUXI2HP Playback Switch",           "AuxI Mix"},
 372        {"HP Mix", "MIC12HP Playback Switch",           "MIC1 PGA"},
 373        {"HP Mix", "MIC22HP Playback Switch",           "MIC2 PGA"},
 374        {"HP Mix", "DAC2HP Playback Switch",            "I2S Mix"},
 375
 376        /* speaker mixer */
 377        {"Speaker Mix", "LI2SPK Playback Switch",       "Line Mix"},
 378        {"Speaker Mix", "AUXI2SPK Playback Switch",     "AuxI Mix"},
 379        {"Speaker Mix", "MIC12SPK Playback Switch",     "MIC1 PGA"},
 380        {"Speaker Mix", "MIC22SPK Playback Switch",     "MIC2 PGA"},
 381        {"Speaker Mix", "DAC2SPK Playback Switch",      "I2S Mix"},
 382
 383        /* mono mixer */
 384        {"Mono Mix", "ADC2MONO_L Playback Switch",      "Left Capture Mix"},
 385        {"Mono Mix", "ADC2MONO_R Playback Switch",      "Right Capture Mix"},
 386        {"Mono Mix", "LI2MONO Playback Switch",         "Line Mix"},
 387        {"Mono Mix", "AUXI2MONO Playback Switch",       "AuxI Mix"},
 388        {"Mono Mix", "MIC12MONO Playback Switch",       "MIC1 PGA"},
 389        {"Mono Mix", "MIC22MONO Playback Switch",       "MIC2 PGA"},
 390        {"Mono Mix", "DAC2MONO Playback Switch",        "I2S Mix"},
 391
 392        /* Left record mixer */
 393        {"Left Capture Mix", "LineInL Capture Switch",  "LINEINL"},
 394        {"Left Capture Mix", "Left AuxI Capture Switch", "AUXINL"},
 395        {"Left Capture Mix", "Mic1 Capture Switch",     "MIC1 Pre Amp"},
 396        {"Left Capture Mix", "Mic2 Capture Switch",     "MIC2 Pre Amp"},
 397        {"Left Capture Mix", "HPMixerL Capture Switch", "HPL Mix"},
 398        {"Left Capture Mix", "SPKMixer Capture Switch", "Speaker Mix"},
 399        {"Left Capture Mix", "MonoMixer Capture Switch", "Mono Mix"},
 400
 401        /*Right record mixer */
 402        {"Right Capture Mix", "LineInR Capture Switch", "LINEINR"},
 403        {"Right Capture Mix", "Right AuxI Capture Switch",      "AUXINR"},
 404        {"Right Capture Mix", "Mic1 Capture Switch",    "MIC1 Pre Amp"},
 405        {"Right Capture Mix", "Mic2 Capture Switch",    "MIC2 Pre Amp"},
 406        {"Right Capture Mix", "HPMixerR Capture Switch", "HPR Mix"},
 407        {"Right Capture Mix", "SPKMixer Capture Switch", "Speaker Mix"},
 408        {"Right Capture Mix", "MonoMixer Capture Switch", "Mono Mix"},
 409
 410        /* headphone left mux */
 411        {"Left Headphone Mux", "HP Left Mix",           "HPL Mix"},
 412        {"Left Headphone Mux", "Vmid",                  "Vmid"},
 413
 414        /* headphone right mux */
 415        {"Right Headphone Mux", "HP Right Mix",         "HPR Mix"},
 416        {"Right Headphone Mux", "Vmid",                 "Vmid"},
 417
 418        /* speaker out mux */
 419        {"SpeakerOut Mux", "Vmid",                      "Vmid"},
 420        {"SpeakerOut Mux", "HPOut Mix",                 "HPOut Mix"},
 421        {"SpeakerOut Mux", "Speaker Mix",               "Speaker Mix"},
 422        {"SpeakerOut Mux", "Mono Mix",                  "Mono Mix"},
 423
 424        /* Mono/Aux Out mux */
 425        {"AuxOut Mux", "Vmid",                          "Vmid"},
 426        {"AuxOut Mux", "HPOut Mix",                     "HPOut Mix"},
 427        {"AuxOut Mux", "Speaker Mix",                   "Speaker Mix"},
 428        {"AuxOut Mux", "Mono Mix",                      "Mono Mix"},
 429
 430        /* output pga */
 431        {"HPL", NULL,                                   "Left Headphone"},
 432        {"Left Headphone", NULL,                        "Left Headphone Mux"},
 433        {"HPR", NULL,                                   "Right Headphone"},
 434        {"Right Headphone", NULL,                       "Right Headphone Mux"},
 435        {"Left AuxOut", NULL,                           "AuxOut Mux"},
 436        {"Right AuxOut", NULL,                          "AuxOut Mux"},
 437
 438        /* input pga */
 439        {"Left LineIn", NULL,                           "LINEINL"},
 440        {"Right LineIn", NULL,                          "LINEINR"},
 441        {"Left AuxI", NULL,                             "AUXINL"},
 442        {"Right AuxI", NULL,                            "AUXINR"},
 443        {"MIC1 Pre Amp", NULL,                          "MIC1"},
 444        {"MIC2 Pre Amp", NULL,                          "MIC2"},
 445        {"MIC1 PGA", NULL,                              "MIC1 Pre Amp"},
 446        {"MIC2 PGA", NULL,                              "MIC2 Pre Amp"},
 447
 448        /* left ADC */
 449        {"Left ADC", NULL,                              "Left Capture Mix"},
 450
 451        /* right ADC */
 452        {"Right ADC", NULL,                             "Right Capture Mix"},
 453
 454        {"SpeakerOut N Mux", "RN/-R",                   "SpeakerOut"},
 455        {"SpeakerOut N Mux", "RP/+R",                   "SpeakerOut"},
 456        {"SpeakerOut N Mux", "LN/-R",                   "SpeakerOut"},
 457        {"SpeakerOut N Mux", "Vmid",                    "Vmid"},
 458
 459        {"SPKOUT", NULL,                                "SpeakerOut"},
 460        {"SPKOUTN", NULL,                               "SpeakerOut N Mux"},
 461};
 462
 463static const struct snd_soc_dapm_route intercon_spk[] = {
 464        {"SpeakerOut", NULL,                            "SpeakerOut Mux"},
 465};
 466
 467static const struct snd_soc_dapm_route intercon_amp_spk[] = {
 468        {"AB Amp", NULL,                                "SpeakerOut Mux"},
 469        {"D Amp", NULL,                                 "SpeakerOut Mux"},
 470        {"AB-D Amp Mux", "AB Amp",                      "AB Amp"},
 471        {"AB-D Amp Mux", "D Amp",                       "D Amp"},
 472        {"SpeakerOut", NULL,                            "AB-D Amp Mux"},
 473};
 474
 475/* PLL divisors */
 476struct _pll_div {
 477        u32 pll_in;
 478        u32 pll_out;
 479        u16 regvalue;
 480};
 481
 482/* Note : pll code from original alc5623 driver. Not sure of how good it is */
 483/* useful only for master mode */
 484static const struct _pll_div codec_master_pll_div[] = {
 485
 486        {  2048000,  8192000,   0x0ea0},
 487        {  3686400,  8192000,   0x4e27},
 488        { 12000000,  8192000,   0x456b},
 489        { 13000000,  8192000,   0x495f},
 490        { 13100000,  8192000,   0x0320},
 491        {  2048000,  11289600,  0xf637},
 492        {  3686400,  11289600,  0x2f22},
 493        { 12000000,  11289600,  0x3e2f},
 494        { 13000000,  11289600,  0x4d5b},
 495        { 13100000,  11289600,  0x363b},
 496        {  2048000,  16384000,  0x1ea0},
 497        {  3686400,  16384000,  0x9e27},
 498        { 12000000,  16384000,  0x452b},
 499        { 13000000,  16384000,  0x542f},
 500        { 13100000,  16384000,  0x03a0},
 501        {  2048000,  16934400,  0xe625},
 502        {  3686400,  16934400,  0x9126},
 503        { 12000000,  16934400,  0x4d2c},
 504        { 13000000,  16934400,  0x742f},
 505        { 13100000,  16934400,  0x3c27},
 506        {  2048000,  22579200,  0x2aa0},
 507        {  3686400,  22579200,  0x2f20},
 508        { 12000000,  22579200,  0x7e2f},
 509        { 13000000,  22579200,  0x742f},
 510        { 13100000,  22579200,  0x3c27},
 511        {  2048000,  24576000,  0x2ea0},
 512        {  3686400,  24576000,  0xee27},
 513        { 12000000,  24576000,  0x2915},
 514        { 13000000,  24576000,  0x772e},
 515        { 13100000,  24576000,  0x0d20},
 516};
 517
 518static const struct _pll_div codec_slave_pll_div[] = {
 519
 520        {  1024000,  16384000,  0x3ea0},
 521        {  1411200,  22579200,  0x3ea0},
 522        {  1536000,  24576000,  0x3ea0},
 523        {  2048000,  16384000,  0x1ea0},
 524        {  2822400,  22579200,  0x1ea0},
 525        {  3072000,  24576000,  0x1ea0},
 526
 527};
 528
 529static int alc5623_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id,
 530                int source, unsigned int freq_in, unsigned int freq_out)
 531{
 532        int i;
 533        struct snd_soc_codec *codec = codec_dai->codec;
 534        int gbl_clk = 0, pll_div = 0;
 535        u16 reg;
 536
 537        if (pll_id < ALC5623_PLL_FR_MCLK || pll_id > ALC5623_PLL_FR_BCK)
 538                return -ENODEV;
 539
 540        /* Disable PLL power */
 541        snd_soc_update_bits(codec, ALC5623_PWR_MANAG_ADD2,
 542                                ALC5623_PWR_ADD2_PLL,
 543                                0);
 544
 545        /* pll is not used in slave mode */
 546        reg = snd_soc_read(codec, ALC5623_DAI_CONTROL);
 547        if (reg & ALC5623_DAI_SDP_SLAVE_MODE)
 548                return 0;
 549
 550        if (!freq_in || !freq_out)
 551                return 0;
 552
 553        switch (pll_id) {
 554        case ALC5623_PLL_FR_MCLK:
 555                for (i = 0; i < ARRAY_SIZE(codec_master_pll_div); i++) {
 556                        if (codec_master_pll_div[i].pll_in == freq_in
 557                           && codec_master_pll_div[i].pll_out == freq_out) {
 558                                /* PLL source from MCLK */
 559                                pll_div  = codec_master_pll_div[i].regvalue;
 560                                break;
 561                        }
 562                }
 563                break;
 564        case ALC5623_PLL_FR_BCK:
 565                for (i = 0; i < ARRAY_SIZE(codec_slave_pll_div); i++) {
 566                        if (codec_slave_pll_div[i].pll_in == freq_in
 567                           && codec_slave_pll_div[i].pll_out == freq_out) {
 568                                /* PLL source from Bitclk */
 569                                gbl_clk = ALC5623_GBL_CLK_PLL_SOUR_SEL_BITCLK;
 570                                pll_div = codec_slave_pll_div[i].regvalue;
 571                                break;
 572                        }
 573                }
 574                break;
 575        default:
 576                return -EINVAL;
 577        }
 578
 579        if (!pll_div)
 580                return -EINVAL;
 581
 582        snd_soc_write(codec, ALC5623_GLOBAL_CLK_CTRL_REG, gbl_clk);
 583        snd_soc_write(codec, ALC5623_PLL_CTRL, pll_div);
 584        snd_soc_update_bits(codec, ALC5623_PWR_MANAG_ADD2,
 585                                ALC5623_PWR_ADD2_PLL,
 586                                ALC5623_PWR_ADD2_PLL);
 587        gbl_clk |= ALC5623_GBL_CLK_SYS_SOUR_SEL_PLL;
 588        snd_soc_write(codec, ALC5623_GLOBAL_CLK_CTRL_REG, gbl_clk);
 589
 590        return 0;
 591}
 592
 593struct _coeff_div {
 594        u16 fs;
 595        u16 regvalue;
 596};
 597
 598/* codec hifi mclk (after PLL) clock divider coefficients */
 599/* values inspired from column BCLK=32Fs of Appendix A table */
 600static const struct _coeff_div coeff_div[] = {
 601        {256*8, 0x3a69},
 602        {384*8, 0x3c6b},
 603        {256*4, 0x2a69},
 604        {384*4, 0x2c6b},
 605        {256*2, 0x1a69},
 606        {384*2, 0x1c6b},
 607        {256*1, 0x0a69},
 608        {384*1, 0x0c6b},
 609};
 610
 611static int get_coeff(struct snd_soc_codec *codec, int rate)
 612{
 613        struct alc5623_priv *alc5623 = snd_soc_codec_get_drvdata(codec);
 614        int i;
 615
 616        for (i = 0; i < ARRAY_SIZE(coeff_div); i++) {
 617                if (coeff_div[i].fs * rate == alc5623->sysclk)
 618                        return i;
 619        }
 620        return -EINVAL;
 621}
 622
 623/*
 624 * Clock after PLL and dividers
 625 */
 626static int alc5623_set_dai_sysclk(struct snd_soc_dai *codec_dai,
 627                int clk_id, unsigned int freq, int dir)
 628{
 629        struct snd_soc_codec *codec = codec_dai->codec;
 630        struct alc5623_priv *alc5623 = snd_soc_codec_get_drvdata(codec);
 631
 632        switch (freq) {
 633        case  8192000:
 634        case 11289600:
 635        case 12288000:
 636        case 16384000:
 637        case 16934400:
 638        case 18432000:
 639        case 22579200:
 640        case 24576000:
 641                alc5623->sysclk = freq;
 642                return 0;
 643        }
 644        return -EINVAL;
 645}
 646
 647static int alc5623_set_dai_fmt(struct snd_soc_dai *codec_dai,
 648                unsigned int fmt)
 649{
 650        struct snd_soc_codec *codec = codec_dai->codec;
 651        u16 iface = 0;
 652
 653        /* set master/slave audio interface */
 654        switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
 655        case SND_SOC_DAIFMT_CBM_CFM:
 656                iface = ALC5623_DAI_SDP_MASTER_MODE;
 657                break;
 658        case SND_SOC_DAIFMT_CBS_CFS:
 659                iface = ALC5623_DAI_SDP_SLAVE_MODE;
 660                break;
 661        default:
 662                return -EINVAL;
 663        }
 664
 665        /* interface format */
 666        switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
 667        case SND_SOC_DAIFMT_I2S:
 668                iface |= ALC5623_DAI_I2S_DF_I2S;
 669                break;
 670        case SND_SOC_DAIFMT_RIGHT_J:
 671                iface |= ALC5623_DAI_I2S_DF_RIGHT;
 672                break;
 673        case SND_SOC_DAIFMT_LEFT_J:
 674                iface |= ALC5623_DAI_I2S_DF_LEFT;
 675                break;
 676        case SND_SOC_DAIFMT_DSP_A:
 677                iface |= ALC5623_DAI_I2S_DF_PCM;
 678                break;
 679        case SND_SOC_DAIFMT_DSP_B:
 680                iface |= ALC5623_DAI_I2S_DF_PCM | ALC5623_DAI_I2S_PCM_MODE;
 681                break;
 682        default:
 683                return -EINVAL;
 684        }
 685
 686        /* clock inversion */
 687        switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
 688        case SND_SOC_DAIFMT_NB_NF:
 689                break;
 690        case SND_SOC_DAIFMT_IB_IF:
 691                iface |= ALC5623_DAI_MAIN_I2S_BCLK_POL_CTRL;
 692                break;
 693        case SND_SOC_DAIFMT_IB_NF:
 694                iface |= ALC5623_DAI_MAIN_I2S_BCLK_POL_CTRL;
 695                break;
 696        case SND_SOC_DAIFMT_NB_IF:
 697                break;
 698        default:
 699                return -EINVAL;
 700        }
 701
 702        return snd_soc_write(codec, ALC5623_DAI_CONTROL, iface);
 703}
 704
 705static int alc5623_pcm_hw_params(struct snd_pcm_substream *substream,
 706                struct snd_pcm_hw_params *params, struct snd_soc_dai *dai)
 707{
 708        struct snd_soc_pcm_runtime *rtd = substream->private_data;
 709        struct snd_soc_codec *codec = rtd->codec;
 710        struct alc5623_priv *alc5623 = snd_soc_codec_get_drvdata(codec);
 711        int coeff, rate;
 712        u16 iface;
 713
 714        iface = snd_soc_read(codec, ALC5623_DAI_CONTROL);
 715        iface &= ~ALC5623_DAI_I2S_DL_MASK;
 716
 717        /* bit size */
 718        switch (params_format(params)) {
 719        case SNDRV_PCM_FORMAT_S16_LE:
 720                iface |= ALC5623_DAI_I2S_DL_16;
 721                break;
 722        case SNDRV_PCM_FORMAT_S20_3LE:
 723                iface |= ALC5623_DAI_I2S_DL_20;
 724                break;
 725        case SNDRV_PCM_FORMAT_S24_LE:
 726                iface |= ALC5623_DAI_I2S_DL_24;
 727                break;
 728        case SNDRV_PCM_FORMAT_S32_LE:
 729                iface |= ALC5623_DAI_I2S_DL_32;
 730                break;
 731        default:
 732                return -EINVAL;
 733        }
 734
 735        /* set iface & srate */
 736        snd_soc_write(codec, ALC5623_DAI_CONTROL, iface);
 737        rate = params_rate(params);
 738        coeff = get_coeff(codec, rate);
 739        if (coeff < 0)
 740                return -EINVAL;
 741
 742        coeff = coeff_div[coeff].regvalue;
 743        dev_dbg(codec->dev, "%s: sysclk=%d,rate=%d,coeff=0x%04x\n",
 744                __func__, alc5623->sysclk, rate, coeff);
 745        snd_soc_write(codec, ALC5623_STEREO_AD_DA_CLK_CTRL, coeff);
 746
 747        return 0;
 748}
 749
 750static int alc5623_mute(struct snd_soc_dai *dai, int mute)
 751{
 752        struct snd_soc_codec *codec = dai->codec;
 753        u16 hp_mute = ALC5623_MISC_M_DAC_L_INPUT | ALC5623_MISC_M_DAC_R_INPUT;
 754        u16 mute_reg = snd_soc_read(codec, ALC5623_MISC_CTRL) & ~hp_mute;
 755
 756        if (mute)
 757                mute_reg |= hp_mute;
 758
 759        return snd_soc_write(codec, ALC5623_MISC_CTRL, mute_reg);
 760}
 761
 762#define ALC5623_ADD2_POWER_EN (ALC5623_PWR_ADD2_VREF \
 763        | ALC5623_PWR_ADD2_DAC_REF_CIR)
 764
 765#define ALC5623_ADD3_POWER_EN (ALC5623_PWR_ADD3_MAIN_BIAS \
 766        | ALC5623_PWR_ADD3_MIC1_BOOST_AD)
 767
 768#define ALC5623_ADD1_POWER_EN \
 769        (ALC5623_PWR_ADD1_SHORT_CURR_DET_EN | ALC5623_PWR_ADD1_SOFTGEN_EN \
 770        | ALC5623_PWR_ADD1_DEPOP_BUF_HP | ALC5623_PWR_ADD1_HP_OUT_AMP \
 771        | ALC5623_PWR_ADD1_HP_OUT_ENH_AMP)
 772
 773#define ALC5623_ADD1_POWER_EN_5622 \
 774        (ALC5623_PWR_ADD1_SHORT_CURR_DET_EN \
 775        | ALC5623_PWR_ADD1_HP_OUT_AMP)
 776
 777static void enable_power_depop(struct snd_soc_codec *codec)
 778{
 779        struct alc5623_priv *alc5623 = snd_soc_codec_get_drvdata(codec);
 780
 781        snd_soc_update_bits(codec, ALC5623_PWR_MANAG_ADD1,
 782                                ALC5623_PWR_ADD1_SOFTGEN_EN,
 783                                ALC5623_PWR_ADD1_SOFTGEN_EN);
 784
 785        snd_soc_write(codec, ALC5623_PWR_MANAG_ADD3, ALC5623_ADD3_POWER_EN);
 786
 787        snd_soc_update_bits(codec, ALC5623_MISC_CTRL,
 788                                ALC5623_MISC_HP_DEPOP_MODE2_EN,
 789                                ALC5623_MISC_HP_DEPOP_MODE2_EN);
 790
 791        msleep(500);
 792
 793        snd_soc_write(codec, ALC5623_PWR_MANAG_ADD2, ALC5623_ADD2_POWER_EN);
 794
 795        /* avoid writing '1' into 5622 reserved bits */
 796        if (alc5623->id == 0x22)
 797                snd_soc_write(codec, ALC5623_PWR_MANAG_ADD1,
 798                        ALC5623_ADD1_POWER_EN_5622);
 799        else
 800                snd_soc_write(codec, ALC5623_PWR_MANAG_ADD1,
 801                        ALC5623_ADD1_POWER_EN);
 802
 803        /* disable HP Depop2 */
 804        snd_soc_update_bits(codec, ALC5623_MISC_CTRL,
 805                                ALC5623_MISC_HP_DEPOP_MODE2_EN,
 806                                0);
 807
 808}
 809
 810static int alc5623_set_bias_level(struct snd_soc_codec *codec,
 811                                      enum snd_soc_bias_level level)
 812{
 813        switch (level) {
 814        case SND_SOC_BIAS_ON:
 815                enable_power_depop(codec);
 816                break;
 817        case SND_SOC_BIAS_PREPARE:
 818                break;
 819        case SND_SOC_BIAS_STANDBY:
 820                /* everything off except vref/vmid, */
 821                snd_soc_write(codec, ALC5623_PWR_MANAG_ADD2,
 822                                ALC5623_PWR_ADD2_VREF);
 823                snd_soc_write(codec, ALC5623_PWR_MANAG_ADD3,
 824                                ALC5623_PWR_ADD3_MAIN_BIAS);
 825                break;
 826        case SND_SOC_BIAS_OFF:
 827                /* everything off, dac mute, inactive */
 828                snd_soc_write(codec, ALC5623_PWR_MANAG_ADD2, 0);
 829                snd_soc_write(codec, ALC5623_PWR_MANAG_ADD3, 0);
 830                snd_soc_write(codec, ALC5623_PWR_MANAG_ADD1, 0);
 831                break;
 832        }
 833        codec->dapm.bias_level = level;
 834        return 0;
 835}
 836
 837#define ALC5623_FORMATS (SNDRV_PCM_FMTBIT_S16_LE \
 838                        | SNDRV_PCM_FMTBIT_S24_LE \
 839                        | SNDRV_PCM_FMTBIT_S32_LE)
 840
 841static const struct snd_soc_dai_ops alc5623_dai_ops = {
 842                .hw_params = alc5623_pcm_hw_params,
 843                .digital_mute = alc5623_mute,
 844                .set_fmt = alc5623_set_dai_fmt,
 845                .set_sysclk = alc5623_set_dai_sysclk,
 846                .set_pll = alc5623_set_dai_pll,
 847};
 848
 849static struct snd_soc_dai_driver alc5623_dai = {
 850        .name = "alc5623-hifi",
 851        .playback = {
 852                .stream_name = "Playback",
 853                .channels_min = 1,
 854                .channels_max = 2,
 855                .rate_min =     8000,
 856                .rate_max =     48000,
 857                .rates = SNDRV_PCM_RATE_8000_48000,
 858                .formats = ALC5623_FORMATS,},
 859        .capture = {
 860                .stream_name = "Capture",
 861                .channels_min = 1,
 862                .channels_max = 2,
 863                .rate_min =     8000,
 864                .rate_max =     48000,
 865                .rates = SNDRV_PCM_RATE_8000_48000,
 866                .formats = ALC5623_FORMATS,},
 867
 868        .ops = &alc5623_dai_ops,
 869};
 870
 871static int alc5623_suspend(struct snd_soc_codec *codec)
 872{
 873        alc5623_set_bias_level(codec, SND_SOC_BIAS_OFF);
 874        return 0;
 875}
 876
 877static int alc5623_resume(struct snd_soc_codec *codec)
 878{
 879        int i, step = codec->driver->reg_cache_step;
 880        u16 *cache = codec->reg_cache;
 881
 882        /* Sync reg_cache with the hardware */
 883        for (i = 2 ; i < codec->driver->reg_cache_size ; i += step)
 884                snd_soc_write(codec, i, cache[i]);
 885
 886        alc5623_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 887
 888        /* charge alc5623 caps */
 889        if (codec->dapm.suspend_bias_level == SND_SOC_BIAS_ON) {
 890                alc5623_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 891                codec->dapm.bias_level = SND_SOC_BIAS_ON;
 892                alc5623_set_bias_level(codec, codec->dapm.bias_level);
 893        }
 894
 895        return 0;
 896}
 897
 898static int alc5623_probe(struct snd_soc_codec *codec)
 899{
 900        struct alc5623_priv *alc5623 = snd_soc_codec_get_drvdata(codec);
 901        struct snd_soc_dapm_context *dapm = &codec->dapm;
 902        int ret;
 903
 904        ret = snd_soc_codec_set_cache_io(codec, 8, 16, alc5623->control_type);
 905        if (ret < 0) {
 906                dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
 907                return ret;
 908        }
 909
 910        alc5623_reset(codec);
 911        alc5623_fill_cache(codec);
 912
 913        /* power on device */
 914        alc5623_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 915
 916        if (alc5623->add_ctrl) {
 917                snd_soc_write(codec, ALC5623_ADD_CTRL_REG,
 918                                alc5623->add_ctrl);
 919        }
 920
 921        if (alc5623->jack_det_ctrl) {
 922                snd_soc_write(codec, ALC5623_JACK_DET_CTRL,
 923                                alc5623->jack_det_ctrl);
 924        }
 925
 926        switch (alc5623->id) {
 927        case 0x21:
 928                snd_soc_add_codec_controls(codec, alc5621_vol_snd_controls,
 929                        ARRAY_SIZE(alc5621_vol_snd_controls));
 930                break;
 931        case 0x22:
 932                snd_soc_add_codec_controls(codec, alc5622_vol_snd_controls,
 933                        ARRAY_SIZE(alc5622_vol_snd_controls));
 934                break;
 935        case 0x23:
 936                snd_soc_add_codec_controls(codec, alc5623_vol_snd_controls,
 937                        ARRAY_SIZE(alc5623_vol_snd_controls));
 938                break;
 939        default:
 940                return -EINVAL;
 941        }
 942
 943        snd_soc_add_codec_controls(codec, alc5623_snd_controls,
 944                        ARRAY_SIZE(alc5623_snd_controls));
 945
 946        snd_soc_dapm_new_controls(dapm, alc5623_dapm_widgets,
 947                                        ARRAY_SIZE(alc5623_dapm_widgets));
 948
 949        /* set up audio path interconnects */
 950        snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon));
 951
 952        switch (alc5623->id) {
 953        case 0x21:
 954        case 0x22:
 955                snd_soc_dapm_new_controls(dapm, alc5623_dapm_amp_widgets,
 956                                        ARRAY_SIZE(alc5623_dapm_amp_widgets));
 957                snd_soc_dapm_add_routes(dapm, intercon_amp_spk,
 958                                        ARRAY_SIZE(intercon_amp_spk));
 959                break;
 960        case 0x23:
 961                snd_soc_dapm_add_routes(dapm, intercon_spk,
 962                                        ARRAY_SIZE(intercon_spk));
 963                break;
 964        default:
 965                return -EINVAL;
 966        }
 967
 968        return ret;
 969}
 970
 971/* power down chip */
 972static int alc5623_remove(struct snd_soc_codec *codec)
 973{
 974        alc5623_set_bias_level(codec, SND_SOC_BIAS_OFF);
 975        return 0;
 976}
 977
 978static struct snd_soc_codec_driver soc_codec_device_alc5623 = {
 979        .probe = alc5623_probe,
 980        .remove = alc5623_remove,
 981        .suspend = alc5623_suspend,
 982        .resume = alc5623_resume,
 983        .set_bias_level = alc5623_set_bias_level,
 984        .reg_cache_size = ALC5623_VENDOR_ID2+2,
 985        .reg_word_size = sizeof(u16),
 986        .reg_cache_step = 2,
 987};
 988
 989/*
 990 * ALC5623 2 wire address is determined by A1 pin
 991 * state during powerup.
 992 *    low  = 0x1a
 993 *    high = 0x1b
 994 */
 995static __devinit int alc5623_i2c_probe(struct i2c_client *client,
 996                                const struct i2c_device_id *id)
 997{
 998        struct alc5623_platform_data *pdata;
 999        struct alc5623_priv *alc5623;
1000        int ret, vid1, vid2;
1001
1002        vid1 = i2c_smbus_read_word_data(client, ALC5623_VENDOR_ID1);
1003        if (vid1 < 0) {
1004                dev_err(&client->dev, "failed to read I2C\n");
1005                return -EIO;
1006        }
1007        vid1 = ((vid1 & 0xff) << 8) | (vid1 >> 8);
1008
1009        vid2 = i2c_smbus_read_byte_data(client, ALC5623_VENDOR_ID2);
1010        if (vid2 < 0) {
1011                dev_err(&client->dev, "failed to read I2C\n");
1012                return -EIO;
1013        }
1014
1015        if ((vid1 != 0x10ec) || (vid2 != id->driver_data)) {
1016                dev_err(&client->dev, "unknown or wrong codec\n");
1017                dev_err(&client->dev, "Expected %x:%lx, got %x:%x\n",
1018                                0x10ec, id->driver_data,
1019                                vid1, vid2);
1020                return -ENODEV;
1021        }
1022
1023        dev_dbg(&client->dev, "Found codec id : alc56%02x\n", vid2);
1024
1025        alc5623 = devm_kzalloc(&client->dev, sizeof(struct alc5623_priv),
1026                               GFP_KERNEL);
1027        if (alc5623 == NULL)
1028                return -ENOMEM;
1029
1030        pdata = client->dev.platform_data;
1031        if (pdata) {
1032                alc5623->add_ctrl = pdata->add_ctrl;
1033                alc5623->jack_det_ctrl = pdata->jack_det_ctrl;
1034        }
1035
1036        alc5623->id = vid2;
1037        switch (alc5623->id) {
1038        case 0x21:
1039                alc5623_dai.name = "alc5621-hifi";
1040                break;
1041        case 0x22:
1042                alc5623_dai.name = "alc5622-hifi";
1043                break;
1044        case 0x23:
1045                alc5623_dai.name = "alc5623-hifi";
1046                break;
1047        default:
1048                return -EINVAL;
1049        }
1050
1051        i2c_set_clientdata(client, alc5623);
1052        alc5623->control_type = SND_SOC_I2C;
1053
1054        ret =  snd_soc_register_codec(&client->dev,
1055                &soc_codec_device_alc5623, &alc5623_dai, 1);
1056        if (ret != 0)
1057                dev_err(&client->dev, "Failed to register codec: %d\n", ret);
1058
1059        return ret;
1060}
1061
1062static __devexit int alc5623_i2c_remove(struct i2c_client *client)
1063{
1064        snd_soc_unregister_codec(&client->dev);
1065        return 0;
1066}
1067
1068static const struct i2c_device_id alc5623_i2c_table[] = {
1069        {"alc5621", 0x21},
1070        {"alc5622", 0x22},
1071        {"alc5623", 0x23},
1072        {}
1073};
1074MODULE_DEVICE_TABLE(i2c, alc5623_i2c_table);
1075
1076/*  i2c codec control layer */
1077static struct i2c_driver alc5623_i2c_driver = {
1078        .driver = {
1079                .name = "alc562x-codec",
1080                .owner = THIS_MODULE,
1081        },
1082        .probe = alc5623_i2c_probe,
1083        .remove =  __devexit_p(alc5623_i2c_remove),
1084        .id_table = alc5623_i2c_table,
1085};
1086
1087static int __init alc5623_modinit(void)
1088{
1089        int ret;
1090
1091        ret = i2c_add_driver(&alc5623_i2c_driver);
1092        if (ret != 0) {
1093                printk(KERN_ERR "%s: can't add i2c driver", __func__);
1094                return ret;
1095        }
1096
1097        return ret;
1098}
1099module_init(alc5623_modinit);
1100
1101static void __exit alc5623_modexit(void)
1102{
1103        i2c_del_driver(&alc5623_i2c_driver);
1104}
1105module_exit(alc5623_modexit);
1106
1107MODULE_DESCRIPTION("ASoC alc5621/2/3 driver");
1108MODULE_AUTHOR("Arnaud Patard <arnaud.patard@rtp-net.org>");
1109MODULE_LICENSE("GPL");
1110