ASoC: add support for alc562[123] codecs
Arnaud Patard (Rtp) [Thu, 21 Oct 2010 17:40:02 +0000 (19:40 +0200)]
This patch is adding support for alc562[123] codecs. It's based
on the source code available in HP source code and other places.

Signed-off-by: Arnaud Patard <arnaud.patard@rtp-net.org>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

include/sound/alc5623.h [new file with mode: 0644]
sound/soc/codecs/Kconfig
sound/soc/codecs/Makefile
sound/soc/codecs/alc5623.c [new file with mode: 0644]
sound/soc/codecs/alc5623.h [new file with mode: 0644]

diff --git a/include/sound/alc5623.h b/include/sound/alc5623.h
new file mode 100644 (file)
index 0000000..422c97d
--- /dev/null
@@ -0,0 +1,15 @@
+#ifndef _INCLUDE_SOUND_ALC5623_H
+#define _INCLUDE_SOUND_ALC5623_H
+struct alc5623_platform_data {
+       /* configure :                              */
+       /* Lineout/Speaker Amps Vmid ratio control  */
+       /* enable/disable adc/dac high pass filters */
+       unsigned int add_ctrl;
+       /* configure :                              */
+       /* output to enable when jack is low        */
+       /* output to enable when jack is high       */
+       /* jack detect (gpio/nc/jack detect [12]    */
+       unsigned int jack_det_ctrl;
+};
+#endif
+
index 94a9d06..658cbe0 100644 (file)
@@ -22,6 +22,7 @@ config SND_SOC_ALL_CODECS
        select SND_SOC_AK4535 if I2C
        select SND_SOC_AK4642 if I2C
        select SND_SOC_AK4671 if I2C
+       select SND_SOC_ALC562 if I2C
        select SND_SOC_CQ0093VC if MFD_DAVINCI_VOICECODEC
        select SND_SOC_CS42L51 if I2C
        select SND_SOC_CS4270 if I2C
@@ -129,6 +130,9 @@ config SND_SOC_AK4642
 config SND_SOC_AK4671
        tristate
 
+config SND_SOC_ALC5623
+       tristate
+
 config SND_SOC_CQ0093VC
        tristate
 
@@ -317,3 +321,4 @@ config SND_SOC_WM2000
 
 config SND_SOC_WM9090
        tristate
+
index f67a2d6..0dcaed3 100644 (file)
@@ -17,6 +17,7 @@ snd-soc-da7210-objs := da7210.o
 snd-soc-l3-objs := l3.o
 snd-soc-max98088-objs := max98088.o
 snd-soc-pcm3008-objs := pcm3008.o
+snd-soc-alc5623-objs := alc5623.o
 snd-soc-spdif-objs := spdif_transciever.o
 snd-soc-ssm2602-objs := ssm2602.o
 snd-soc-stac9766-objs := stac9766.o
@@ -92,6 +93,7 @@ obj-$(CONFIG_SND_SOC_L3)      += snd-soc-l3.o
 obj-$(CONFIG_SND_SOC_JZ4740_CODEC)     += snd-soc-jz4740-codec.o
 obj-$(CONFIG_SND_SOC_MAX98088) += snd-soc-max98088.o
 obj-$(CONFIG_SND_SOC_PCM3008)  += snd-soc-pcm3008.o
+obj-$(CONFIG_SND_SOC_ALC5623)    += snd-soc-alc5623.o
 obj-$(CONFIG_SND_SOC_SPDIF)    += snd-soc-spdif.o
 obj-$(CONFIG_SND_SOC_SSM2602)  += snd-soc-ssm2602.o
 obj-$(CONFIG_SND_SOC_STAC9766) += snd-soc-stac9766.o
diff --git a/sound/soc/codecs/alc5623.c b/sound/soc/codecs/alc5623.c
new file mode 100644 (file)
index 0000000..fac6174
--- /dev/null
@@ -0,0 +1,1118 @@
+/*
+ * 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/soc-dapm.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 */
+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 */
+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;
+}
+
+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->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->suspend_bias_level == SND_SOC_BIAS_ON) {
+               alc5623_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+               codec->bias_level = SND_SOC_BIAS_ON;
+               alc5623_set_bias_level(codec, codec->bias_level);
+       }
+
+       return 0;
+}
+
+static int alc5623_probe(struct snd_soc_codec *codec)
+{
+       struct alc5623_priv *alc5623 = snd_soc_codec_get_drvdata(codec);
+       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) {
+       default:
+       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;
+       }
+
+       snd_soc_add_controls(codec, alc5623_snd_controls,
+                       ARRAY_SIZE(alc5623_snd_controls));
+
+       snd_soc_dapm_new_controls(codec, alc5623_dapm_widgets,
+                                       ARRAY_SIZE(alc5623_dapm_widgets));
+
+       /* set up audio path interconnects */
+       snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon));
+
+       switch (alc5623->id) {
+       default:
+       case 0x21:
+       case 0x22:
+               snd_soc_dapm_new_controls(codec, alc5623_dapm_amp_widgets,
+                                       ARRAY_SIZE(alc5623_dapm_amp_widgets));
+               snd_soc_dapm_add_routes(codec, intercon_amp_spk,
+                                               ARRAY_SIZE(intercon_amp_spk));
+               break;
+       case 0x23:
+               snd_soc_dapm_add_routes(codec, intercon_spk,
+                                               ARRAY_SIZE(intercon_spk));
+               break;
+       }
+
+       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) {
+               ret = -ENOMEM;
+               goto err;
+       }
+
+       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;
+       default:
+       case 0x23:
+               alc5623_dai.name = "alc5623-hifi";
+               break;
+       }
+
+       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);
+               goto err;
+       }
+
+       return 0;
+
+err:
+       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");
diff --git a/sound/soc/codecs/alc5623.h b/sound/soc/codecs/alc5623.h
new file mode 100644 (file)
index 0000000..f3d6826
--- /dev/null
@@ -0,0 +1,161 @@
+/*
+ * alc5623.h  --  alc562[123] ALSA Soc Audio driver
+ *
+ * Copyright 2008 Realtek Microelectronics
+ * Copyright 2010 Arnaud Patard <arnaud.patard@rtp-net.org>
+ *
+ * Author: flove <flove@realtek.com>
+ * Arnaud Patard <arnaud.patard@rtp-net.org>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ *
+ */
+
+#ifndef _ALC5623_H
+#define _ALC5623_H
+
+#define ALC5623_RESET                          0x00
+/*                             5621 5622 5623  */
+/* speaker output vol             2    2       */
+/* line output vol                      4    2  */
+/* HP output vol                  4    0    4  */
+#define ALC5623_SPK_OUT_VOL                    0x02
+#define ALC5623_HP_OUT_VOL                     0x04
+#define ALC5623_MONO_AUX_OUT_VOL               0x06
+#define ALC5623_AUXIN_VOL                      0x08
+#define ALC5623_LINE_IN_VOL                    0x0A
+#define ALC5623_STEREO_DAC_VOL                 0x0C
+#define ALC5623_MIC_VOL                                0x0E
+#define ALC5623_MIC_ROUTING_CTRL               0x10
+#define ALC5623_ADC_REC_GAIN                   0x12
+#define ALC5623_ADC_REC_MIXER                  0x14
+#define ALC5623_SOFT_VOL_CTRL_TIME             0x16
+/* ALC5623_OUTPUT_MIXER_CTRL :                 */
+/* same remark as for reg 2 line vs speaker    */
+#define ALC5623_OUTPUT_MIXER_CTRL              0x1C
+#define ALC5623_MIC_CTRL                       0x22
+
+#define        ALC5623_DAI_CONTROL                     0x34
+#define ALC5623_DAI_SDP_MASTER_MODE            (0 << 15)
+#define ALC5623_DAI_SDP_SLAVE_MODE             (1 << 15)
+#define ALC5623_DAI_I2S_PCM_MODE               (1 << 14)
+#define ALC5623_DAI_MAIN_I2S_BCLK_POL_CTRL     (1 <<  7)
+#define ALC5623_DAI_ADC_DATA_L_R_SWAP          (1 <<  5)
+#define ALC5623_DAI_DAC_DATA_L_R_SWAP          (1 <<  4)
+#define ALC5623_DAI_I2S_DL_MASK                        (3 <<  2)
+#define ALC5623_DAI_I2S_DL_32                  (3 <<  2)
+#define        ALC5623_DAI_I2S_DL_24                   (2 <<  2)
+#define ALC5623_DAI_I2S_DL_20                  (1 <<  2)
+#define ALC5623_DAI_I2S_DL_16                  (0 <<  2)
+#define ALC5623_DAI_I2S_DF_PCM                 (3 <<  0)
+#define        ALC5623_DAI_I2S_DF_LEFT                 (2 <<  0)
+#define ALC5623_DAI_I2S_DF_RIGHT               (1 <<  0)
+#define ALC5623_DAI_I2S_DF_I2S                 (0 <<  0)
+
+#define ALC5623_STEREO_AD_DA_CLK_CTRL          0x36
+#define        ALC5623_COMPANDING_CTRL                 0x38
+
+#define        ALC5623_PWR_MANAG_ADD1                  0x3A
+#define ALC5623_PWR_ADD1_MAIN_I2S_EN           (1 << 15)
+#define ALC5623_PWR_ADD1_ZC_DET_PD_EN          (1 << 14)
+#define ALC5623_PWR_ADD1_MIC1_BIAS_EN          (1 << 11)
+#define ALC5623_PWR_ADD1_SHORT_CURR_DET_EN     (1 << 10)
+#define ALC5623_PWR_ADD1_SOFTGEN_EN            (1 <<  8) /* rsvd on 5622 */
+#define        ALC5623_PWR_ADD1_DEPOP_BUF_HP           (1 <<  6) /* rsvd on 5622 */
+#define        ALC5623_PWR_ADD1_HP_OUT_AMP             (1 <<  5)
+#define        ALC5623_PWR_ADD1_HP_OUT_ENH_AMP         (1 <<  4) /* rsvd on 5622 */
+#define ALC5623_PWR_ADD1_DEPOP_BUF_AUX         (1 <<  2)
+#define ALC5623_PWR_ADD1_AUX_OUT_AMP           (1 <<  1)
+#define ALC5623_PWR_ADD1_AUX_OUT_ENH_AMP       (1 <<  0) /* rsvd on 5622 */
+
+#define ALC5623_PWR_MANAG_ADD2                 0x3C
+#define ALC5623_PWR_ADD2_LINEOUT               (1 << 15) /* rt5623 */
+#define ALC5623_PWR_ADD2_CLASS_AB              (1 << 15) /* rt5621 */
+#define ALC5623_PWR_ADD2_CLASS_D               (1 << 14) /* rt5621 */
+#define ALC5623_PWR_ADD2_VREF                  (1 << 13)
+#define ALC5623_PWR_ADD2_PLL                   (1 << 12)
+#define ALC5623_PWR_ADD2_DAC_REF_CIR           (1 << 10)
+#define ALC5623_PWR_ADD2_L_DAC_CLK             (1 <<  9)
+#define ALC5623_PWR_ADD2_R_DAC_CLK             (1 <<  8)
+#define ALC5623_PWR_ADD2_L_ADC_CLK_GAIN                (1 <<  7)
+#define ALC5623_PWR_ADD2_R_ADC_CLK_GAIN                (1 <<  6)
+#define ALC5623_PWR_ADD2_L_HP_MIXER            (1 <<  5)
+#define ALC5623_PWR_ADD2_R_HP_MIXER            (1 <<  4)
+#define ALC5623_PWR_ADD2_SPK_MIXER             (1 <<  3)
+#define ALC5623_PWR_ADD2_MONO_MIXER            (1 <<  2)
+#define ALC5623_PWR_ADD2_L_ADC_REC_MIXER       (1 <<  1)
+#define ALC5623_PWR_ADD2_R_ADC_REC_MIXER       (1 <<  0)
+
+#define ALC5623_PWR_MANAG_ADD3                 0x3E
+#define ALC5623_PWR_ADD3_MAIN_BIAS             (1 << 15)
+#define ALC5623_PWR_ADD3_AUXOUT_L_VOL_AMP      (1 << 14)
+#define ALC5623_PWR_ADD3_AUXOUT_R_VOL_AMP      (1 << 13)
+#define ALC5623_PWR_ADD3_SPK_OUT               (1 << 12)
+#define ALC5623_PWR_ADD3_HP_L_OUT_VOL          (1 << 10)
+#define ALC5623_PWR_ADD3_HP_R_OUT_VOL          (1 <<  9)
+#define ALC5623_PWR_ADD3_LINEIN_L_VOL          (1 <<  7)
+#define ALC5623_PWR_ADD3_LINEIN_R_VOL          (1 <<  6)
+#define ALC5623_PWR_ADD3_AUXIN_L_VOL           (1 <<  5)
+#define ALC5623_PWR_ADD3_AUXIN_R_VOL           (1 <<  4)
+#define ALC5623_PWR_ADD3_MIC1_FUN_CTRL         (1 <<  3)
+#define ALC5623_PWR_ADD3_MIC2_FUN_CTRL         (1 <<  2)
+#define ALC5623_PWR_ADD3_MIC1_BOOST_AD         (1 <<  1)
+#define ALC5623_PWR_ADD3_MIC2_BOOST_AD         (1 <<  0)
+
+#define ALC5623_ADD_CTRL_REG                   0x40
+
+#define        ALC5623_GLOBAL_CLK_CTRL_REG             0x42
+#define ALC5623_GBL_CLK_SYS_SOUR_SEL_PLL       (1 << 15)
+#define ALC5623_GBL_CLK_SYS_SOUR_SEL_MCLK      (0 << 15)
+#define ALC5623_GBL_CLK_PLL_SOUR_SEL_BITCLK    (1 << 14)
+#define ALC5623_GBL_CLK_PLL_SOUR_SEL_MCLK      (0 << 14)
+#define ALC5623_GBL_CLK_PLL_DIV_RATIO_DIV8     (3 <<  1)
+#define ALC5623_GBL_CLK_PLL_DIV_RATIO_DIV4     (2 <<  1)
+#define ALC5623_GBL_CLK_PLL_DIV_RATIO_DIV2     (1 <<  1)
+#define ALC5623_GBL_CLK_PLL_DIV_RATIO_DIV1     (0 <<  1)
+#define ALC5623_GBL_CLK_PLL_PRE_DIV2           (1 <<  0)
+#define ALC5623_GBL_CLK_PLL_PRE_DIV1           (0 <<  0)
+
+#define ALC5623_PLL_CTRL                       0x44
+#define ALC5623_PLL_CTRL_N_VAL(n)              (((n)&0xff) << 8)
+#define ALC5623_PLL_CTRL_K_VAL(k)              (((k)&0x7)  << 4)
+#define ALC5623_PLL_CTRL_M_VAL(m)              ((m)&0xf)
+
+#define ALC5623_GPIO_OUTPUT_PIN_CTRL           0x4A
+#define ALC5623_GPIO_PIN_CONFIG                        0x4C
+#define ALC5623_GPIO_PIN_POLARITY              0x4E
+#define ALC5623_GPIO_PIN_STICKY                        0x50
+#define ALC5623_GPIO_PIN_WAKEUP                        0x52
+#define ALC5623_GPIO_PIN_STATUS                        0x54
+#define ALC5623_GPIO_PIN_SHARING               0x56
+#define        ALC5623_OVER_CURR_STATUS                0x58
+#define ALC5623_JACK_DET_CTRL                  0x5A
+
+#define ALC5623_MISC_CTRL                      0x5E
+#define ALC5623_MISC_DISABLE_FAST_VREG         (1 << 15)
+#define ALC5623_MISC_SPK_CLASS_AB_OC_PD                (1 << 13) /* 5621 */
+#define ALC5623_MISC_SPK_CLASS_AB_OC_DET       (1 << 12) /* 5621 */
+#define ALC5623_MISC_HP_DEPOP_MODE3_EN         (1 << 10)
+#define ALC5623_MISC_HP_DEPOP_MODE2_EN         (1 <<  9)
+#define ALC5623_MISC_HP_DEPOP_MODE1_EN         (1 <<  8)
+#define ALC5623_MISC_AUXOUT_DEPOP_MODE3_EN     (1 <<  6)
+#define ALC5623_MISC_AUXOUT_DEPOP_MODE2_EN     (1 <<  5)
+#define ALC5623_MISC_AUXOUT_DEPOP_MODE1_EN     (1 <<  4)
+#define ALC5623_MISC_M_DAC_L_INPUT             (1 <<  3)
+#define ALC5623_MISC_M_DAC_R_INPUT             (1 <<  2)
+#define ALC5623_MISC_IRQOUT_INV_CTRL           (1 <<  0)
+
+#define        ALC5623_PSEDUEO_SPATIAL_CTRL            0x60
+#define ALC5623_EQ_CTRL                                0x62
+#define ALC5623_EQ_MODE_ENABLE                 0x66
+#define ALC5623_AVC_CTRL                       0x68
+#define ALC5623_HID_CTRL_INDEX                 0x6A
+#define ALC5623_HID_CTRL_DATA                  0x6C
+#define ALC5623_VENDOR_ID1                     0x7C
+#define ALC5623_VENDOR_ID2                     0x7E
+
+#define ALC5623_PLL_FR_MCLK                    0
+#define ALC5623_PLL_FR_BCK                     1
+#endif