ASoC: Add support for CS42L73 codec

This patch adds support for the Cirrus Logic CS42L73 low power stereo
codec.

Signed-off-by: Brian Austin <brian.austin@cirrus.com>
Signed-off-by: Georgi Vlaev <joe@nucleusys.com>
Acked-by: Vinod Koul <vinod.koul@linux.intel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

4 files changed, 1690 insertions, 0 deletions

diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index 684cc1570689..686f45a07f34 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -29,6 +29,7 @@ config SND_SOC_ALL_CODECS
 	select SND_SOC_ALC5632 if I2C
 	select SND_SOC_CQ0093VC if MFD_DAVINCI_VOICECODEC
 	select SND_SOC_CS42L51 if I2C
+	select SND_SOC_CS42L73 if I2C
 	select SND_SOC_CS4270 if I2C
 	select SND_SOC_CS4271 if SND_SOC_I2C_AND_SPI
 	select SND_SOC_CX20442
@@ -178,6 +179,9 @@ config SND_SOC_CQ0093VC
 config SND_SOC_CS42L51
 	tristate
 
+config SND_SOC_CS42L73
+	tristate
+
 # Cirrus Logic CS4270 Codec
 config SND_SOC_CS4270
 	tristate
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index af64905f36ca..62b01e4e7983 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -15,6 +15,7 @@ snd-soc-ak4642-objs := ak4642.o
 snd-soc-ak4671-objs := ak4671.o
 snd-soc-cq93vc-objs := cq93vc.o
 snd-soc-cs42l51-objs := cs42l51.o
+snd-soc-cs42l73-objs := cs42l73.o
 snd-soc-cs4270-objs := cs4270.o
 snd-soc-cs4271-objs := cs4271.o
 snd-soc-cx20442-objs := cx20442.o
@@ -117,6 +118,7 @@ obj-$(CONFIG_SND_SOC_ALC5623)    += snd-soc-alc5623.o
 obj-$(CONFIG_SND_SOC_ALC5632)	+= snd-soc-alc5632.o
 obj-$(CONFIG_SND_SOC_CQ0093VC) += snd-soc-cq93vc.o
 obj-$(CONFIG_SND_SOC_CS42L51)	+= snd-soc-cs42l51.o
+obj-$(CONFIG_SND_SOC_CS42L73)	+= snd-soc-cs42l73.o
 obj-$(CONFIG_SND_SOC_CS4270)	+= snd-soc-cs4270.o
 obj-$(CONFIG_SND_SOC_CS4271)	+= snd-soc-cs4271.o
 obj-$(CONFIG_SND_SOC_CX20442)	+= snd-soc-cx20442.o
diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c
new file mode 100644
index 000000000000..6fe259aceb7b
--- /dev/null
+++ b/sound/soc/codecs/cs42l73.c
@@ -0,0 +1,1457 @@
+/*
+ * cs42l73.c  --  CS42L73 ALSA Soc Audio driver
+ *
+ * Copyright 2011 Cirrus Logic, Inc.
+ *
+ * Authors: Georgi Vlaev, Nucleus Systems Ltd, <joe@nucleusys.com>
+ *	    Brian Austin, Cirrus Logic Inc, <brian.austin@cirrus.com>
+ *
+ * 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/moduleparam.h>
+#include <linux/kernel.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/i2c.h>
+#include <linux/regmap.h>
+#include <linux/slab.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/initval.h>
+#include <sound/tlv.h>
+#include "cs42l73.h"
+
+struct sp_config {
+	u8 spc, mmcc, spfs;
+	u32 srate;
+};
+struct  cs42l73_private {
+	struct sp_config config[3];
+	struct regmap *regmap;
+	u32 sysclk;
+	u8 mclksel;
+	u32 mclk;
+};
+
+struct reg_default cs42l73_reg_defaults[] = {
+	{ 1, 0x42 },	/* r01	- Device ID A&B */
+	{ 2, 0xA7 },	/* r02	- Device ID C&D */
+	{ 3, 0x30 },	/* r03	- Device ID E */
+	{ 6, 0xF1 },	/* r06	- Power Ctl 1 */
+	{ 7, 0xDF },	/* r07	- Power Ctl 2 */
+	{ 8, 0x3F },	/* r08	- Power Ctl 3 */
+	{ 9, 0x50 },	/* r09	- Charge Pump Freq */
+	{ 10, 0x53 },	/* r0A	- Output Load MicBias Short Detect */
+	{ 11, 0x00 },	/* r0B	- DMIC Master Clock Ctl */
+	{ 12, 0x00 },	/* r0C	- Aux PCM Ctl */
+	{ 13, 0x15 },	/* r0D	- Aux PCM Master Clock Ctl */
+	{ 14, 0x00 },	/* r0E	- Audio PCM Ctl */
+	{ 15, 0x15 },	/* r0F	- Audio PCM Master Clock Ctl */
+	{ 16, 0x00 },	/* r10	- Voice PCM Ctl */
+	{ 17, 0x15 },	/* r11	- Voice PCM Master Clock Ctl */
+	{ 18, 0x00 },	/* r12	- Voice/Aux Sample Rate */
+	{ 19, 0x06 },	/* r13	- Misc I/O Path Ctl */
+	{ 20, 0x00 },	/* r14	- ADC Input Path Ctl */
+	{ 21, 0x00 },	/* r15	- MICA Preamp, PGA Volume */
+	{ 22, 0x00 },	/* r16	- MICB Preamp, PGA Volume */
+	{ 23, 0x00 },	/* r17	- Input Path A Digital Volume */
+	{ 24, 0x00 },	/* r18	- Input Path B Digital Volume */
+	{ 25, 0x00 },	/* r19	- Playback Digital Ctl */
+	{ 26, 0x00 },	/* r1A	- HP/LO Left Digital Volume */
+	{ 27, 0x00 },	/* r1B	- HP/LO Right Digital Volume */
+	{ 28, 0x00 },	/* r1C	- Speakerphone Digital Volume */
+	{ 29, 0x00 },	/* r1D	- Ear/SPKLO Digital Volume */
+	{ 30, 0x00 },	/* r1E	- HP Left Analog Volume */
+	{ 31, 0x00 },	/* r1F	- HP Right Analog Volume */
+	{ 32, 0x00 },	/* r20	- LO Left Analog Volume */
+	{ 33, 0x00 },	/* r21	- LO Right Analog Volume */
+	{ 34, 0x00 },	/* r22	- Stereo Input Path Advisory Volume */
+	{ 35, 0x00 },	/* r23	- Aux PCM Input Advisory Volume */
+	{ 36, 0x00 },	/* r24	- Audio PCM Input Advisory Volume */
+	{ 37, 0x00 },	/* r25	- Voice PCM Input Advisory Volume */
+	{ 38, 0x00 },	/* r26	- Limiter Attack Rate HP/LO */
+	{ 39, 0x7F },	/* r27	- Limter Ctl, Release Rate HP/LO */
+	{ 40, 0x00 },	/* r28	- Limter Threshold HP/LO */
+	{ 41, 0x00 },	/* r29	- Limiter Attack Rate Speakerphone */
+	{ 42, 0x3F },	/* r2A	- Limter Ctl, Release Rate Speakerphone */
+	{ 43, 0x00 },	/* r2B	- Limter Threshold Speakerphone */
+	{ 44, 0x00 },	/* r2C	- Limiter Attack Rate Ear/SPKLO */
+	{ 45, 0x3F },	/* r2D	- Limter Ctl, Release Rate Ear/SPKLO */
+	{ 46, 0x00 },	/* r2E	- Limter Threshold Ear/SPKLO */
+	{ 47, 0x00 },	/* r2F	- ALC Enable, Attack Rate Left/Right */
+	{ 48, 0x3F },	/* r30	- ALC Release Rate Left/Right */
+	{ 49, 0x00 },	/* r31	- ALC Threshold Left/Right */
+	{ 50, 0x00 },	/* r32	- Noise Gate Ctl Left/Right */
+	{ 51, 0x00 },	/* r33	- ALC/NG Misc Ctl */
+	{ 52, 0x18 },	/* r34	- Mixer Ctl */
+	{ 53, 0x3F },	/* r35	- HP/LO Left Mixer Input Path Volume */
+	{ 54, 0x3F },	/* r36	- HP/LO Right Mixer Input Path Volume */
+	{ 55, 0x3F },	/* r37	- HP/LO Left Mixer Aux PCM Volume */
+	{ 56, 0x3F },	/* r38	- HP/LO Right Mixer Aux PCM Volume */
+	{ 57, 0x3F },	/* r39	- HP/LO Left Mixer Audio PCM Volume */
+	{ 58, 0x3F },	/* r3A	- HP/LO Right Mixer Audio PCM Volume */
+	{ 59, 0x3F },	/* r3B	- HP/LO Left Mixer Voice PCM Mono Volume */
+	{ 60, 0x3F },	/* r3C	- HP/LO Right Mixer Voice PCM Mono Volume */
+	{ 61, 0x3F },	/* r3D	- Aux PCM Left Mixer Input Path Volume */
+	{ 62, 0x3F },	/* r3E	- Aux PCM Right Mixer Input Path Volume */
+	{ 63, 0x3F },	/* r3F	- Aux PCM Left Mixer Volume */
+	{ 64, 0x3F },	/* r40	- Aux PCM Left Mixer Volume */
+	{ 65, 0x3F },	/* r41	- Aux PCM Left Mixer Audio PCM L Volume */
+	{ 66, 0x3F },	/* r42	- Aux PCM Right Mixer Audio PCM R Volume */
+	{ 67, 0x3F },	/* r43	- Aux PCM Left Mixer Voice PCM Volume */
+	{ 68, 0x3F },	/* r44	- Aux PCM Right Mixer Voice PCM Volume */
+	{ 69, 0x3F },	/* r45	- Audio PCM Left Input Path Volume */
+	{ 70, 0x3F },	/* r46	- Audio PCM Right Input Path Volume */
+	{ 71, 0x3F },	/* r47	- Audio PCM Left Mixer Aux PCM L Volume */
+	{ 72, 0x3F },	/* r48	- Audio PCM Right Mixer Aux PCM R Volume */
+	{ 73, 0x3F },	/* r49	- Audio PCM Left Mixer Volume */
+	{ 74, 0x3F },	/* r4A	- Audio PCM Right Mixer Volume */
+	{ 75, 0x3F },	/* r4B	- Audio PCM Left Mixer Voice PCM Volume */
+	{ 76, 0x3F },	/* r4C	- Audio PCM Right Mixer Voice PCM Volume */
+	{ 77, 0x3F },	/* r4D	- Voice PCM Left Input Path Volume */
+	{ 78, 0x3F },	/* r4E	- Voice PCM Right Input Path Volume */
+	{ 79, 0x3F },	/* r4F	- Voice PCM Left Mixer Aux PCM L Volume */
+	{ 80, 0x3F },	/* r50	- Voice PCM Right Mixer Aux PCM R Volume */
+	{ 81, 0x3F },	/* r51	- Voice PCM Left Mixer Audio PCM L Volume */
+	{ 82, 0x3F },	/* r52	- Voice PCM Right Mixer Audio PCM R Volume */
+	{ 83, 0x3F },	/* r53	- Voice PCM Left Mixer Voice PCM Volume */
+	{ 84, 0x3F },	/* r54	- Voice PCM Right Mixer Voice PCM Volume */
+	{ 85, 0xAA },	/* r55	- Mono Mixer Ctl */
+	{ 86, 0x3F },	/* r56	- SPK Mono Mixer Input Path Volume */
+	{ 87, 0x3F },	/* r57	- SPK Mono Mixer Aux PCM Mono/L/R Volume */
+	{ 88, 0x3F },	/* r58	- SPK Mono Mixer Audio PCM Mono/L/R Volume */
+	{ 89, 0x3F },	/* r59	- SPK Mono Mixer Voice PCM Mono Volume */
+	{ 90, 0x3F },	/* r5A	- SPKLO Mono Mixer Input Path Mono Volume */
+	{ 91, 0x3F },	/* r5B	- SPKLO Mono Mixer Aux Mono/L/R Volume */
+	{ 92, 0x3F },	/* r5C	- SPKLO Mono Mixer Audio Mono/L/R Volume */
+	{ 93, 0x3F },	/* r5D	- SPKLO Mono Mixer Voice Mono Volume */
+	{ 94, 0x00 },	/* r5E	- Interrupt Mask 1 */
+	{ 95, 0x00 },	/* r5F	- Interrupt Mask 2 */
+};
+
+static bool cs42l73_volatile_register(struct device *dev, unsigned int reg)
+{
+	switch (reg) {
+	case CS42L73_IS1:
+	case CS42L73_IS2:
+		return true;
+	default:
+		return false;
+	}
+}
+
+static bool cs42l73_readable_register(struct device *dev, unsigned int reg)
+{
+	switch (reg) {
+	case CS42L73_DEVID_AB:
+	case CS42L73_DEVID_CD:
+	case CS42L73_DEVID_E:
+	case CS42L73_REVID:
+	case CS42L73_PWRCTL1:
+	case CS42L73_PWRCTL2:
+	case CS42L73_PWRCTL3:
+	case CS42L73_CPFCHC:
+	case CS42L73_OLMBMSDC:
+	case CS42L73_DMMCC:
+	case CS42L73_XSPC:
+	case CS42L73_XSPMMCC:
+	case CS42L73_ASPC:
+	case CS42L73_ASPMMCC:
+	case CS42L73_VSPC:
+	case CS42L73_VSPMMCC:
+	case CS42L73_VXSPFS:
+	case CS42L73_MIOPC:
+	case CS42L73_ADCIPC:
+	case CS42L73_MICAPREPGAAVOL:
+	case CS42L73_MICBPREPGABVOL:
+	case CS42L73_IPADVOL:
+	case CS42L73_IPBDVOL:
+	case CS42L73_PBDC:
+	case CS42L73_HLADVOL:
+	case CS42L73_HLBDVOL:
+	case CS42L73_SPKDVOL:
+	case CS42L73_ESLDVOL:
+	case CS42L73_HPAAVOL:
+	case CS42L73_HPBAVOL:
+	case CS42L73_LOAAVOL:
+	case CS42L73_LOBAVOL:
+	case CS42L73_STRINV:
+	case CS42L73_XSPINV:
+	case CS42L73_ASPINV:
+	case CS42L73_VSPINV:
+	case CS42L73_LIMARATEHL:
+	case CS42L73_LIMRRATEHL:
+	case CS42L73_LMAXHL:
+	case CS42L73_LIMARATESPK:
+	case CS42L73_LIMRRATESPK:
+	case CS42L73_LMAXSPK:
+	case CS42L73_LIMARATEESL:
+	case CS42L73_LIMRRATEESL:
+	case CS42L73_LMAXESL:
+	case CS42L73_ALCARATE:
+	case CS42L73_ALCRRATE:
+	case CS42L73_ALCMINMAX:
+	case CS42L73_NGCAB:
+	case CS42L73_ALCNGMC:
+	case CS42L73_MIXERCTL:
+	case CS42L73_HLAIPAA:
+	case CS42L73_HLBIPBA:
+	case CS42L73_HLAXSPAA:
+	case CS42L73_HLBXSPBA:
+	case CS42L73_HLAASPAA:
+	case CS42L73_HLBASPBA:
+	case CS42L73_HLAVSPMA:
+	case CS42L73_HLBVSPMA:
+	case CS42L73_XSPAIPAA:
+	case CS42L73_XSPBIPBA:
+	case CS42L73_XSPAXSPAA:
+	case CS42L73_XSPBXSPBA:
+	case CS42L73_XSPAASPAA:
+	case CS42L73_XSPAASPBA:
+	case CS42L73_XSPAVSPMA:
+	case CS42L73_XSPBVSPMA:
+	case CS42L73_ASPAIPAA:
+	case CS42L73_ASPBIPBA:
+	case CS42L73_ASPAXSPAA:
+	case CS42L73_ASPBXSPBA:
+	case CS42L73_ASPAASPAA:
+	case CS42L73_ASPBASPBA:
+	case CS42L73_ASPAVSPMA:
+	case CS42L73_ASPBVSPMA:
+	case CS42L73_VSPAIPAA:
+	case CS42L73_VSPBIPBA:
+	case CS42L73_VSPAXSPAA:
+	case CS42L73_VSPBXSPBA:
+	case CS42L73_VSPAASPAA:
+	case CS42L73_VSPBASPBA:
+	case CS42L73_VSPAVSPMA:
+	case CS42L73_VSPBVSPMA:
+	case CS42L73_MMIXCTL:
+	case CS42L73_SPKMIPMA:
+	case CS42L73_SPKMXSPA:
+	case CS42L73_SPKMASPA:
+	case CS42L73_SPKMVSPMA:
+	case CS42L73_ESLMIPMA:
+	case CS42L73_ESLMXSPA:
+	case CS42L73_ESLMASPA:
+	case CS42L73_ESLMVSPMA:
+	case CS42L73_IM1:
+	case CS42L73_IM2:
+		return true;
+	default:
+		return false;
+	}
+}
+
+static const unsigned int hpaloa_tlv[] = {
+	TLV_DB_RANGE_HEAD(2),
+	0, 13, TLV_DB_SCALE_ITEM(-7600, 200, 0),
+	14, 75, TLV_DB_SCALE_ITEM(-4900, 100, 0),
+};
+
+static DECLARE_TLV_DB_SCALE(adc_boost_tlv, 0, 2500, 0);
+
+static DECLARE_TLV_DB_SCALE(hl_tlv, -10200, 50, 0);
+
+static DECLARE_TLV_DB_SCALE(ipd_tlv, -9600, 100, 0);
+
+static DECLARE_TLV_DB_SCALE(micpga_tlv, -600, 50, 0);
+
+static const unsigned int limiter_tlv[] = {
+	TLV_DB_RANGE_HEAD(2),
+	0, 2, TLV_DB_SCALE_ITEM(-3000, 600, 0),
+	3, 7, TLV_DB_SCALE_ITEM(-1200, 300, 0),
+};
+
+static const DECLARE_TLV_DB_SCALE(attn_tlv, -6300, 100, 1);
+
+static const char * const cs42l73_pgaa_text[] = { "Line A", "Mic 1" };
+static const char * const cs42l73_pgab_text[] = { "Line B", "Mic 2" };
+
+static const struct soc_enum pgaa_enum =
+	SOC_ENUM_SINGLE(CS42L73_ADCIPC, 3,
+		ARRAY_SIZE(cs42l73_pgaa_text), cs42l73_pgaa_text);
+
+static const struct soc_enum pgab_enum =
+	SOC_ENUM_SINGLE(CS42L73_ADCIPC, 7,
+		ARRAY_SIZE(cs42l73_pgab_text), cs42l73_pgab_text);
+
+static const struct snd_kcontrol_new pgaa_mux =
+	SOC_DAPM_ENUM("Left Analog Input Capture Mux", pgaa_enum);
+
+static const struct snd_kcontrol_new pgab_mux =
+	SOC_DAPM_ENUM("Right Analog Input Capture Mux", pgab_enum);
+
+static const struct snd_kcontrol_new input_left_mixer[] = {
+	SOC_DAPM_SINGLE("ADC Left Input", CS42L73_PWRCTL1,
+			5, 1, 1),
+	SOC_DAPM_SINGLE("DMIC Left Input", CS42L73_PWRCTL1,
+			4, 1, 1),
+};
+
+static const struct snd_kcontrol_new input_right_mixer[] = {
+	SOC_DAPM_SINGLE("ADC Right Input", CS42L73_PWRCTL1,
+			7, 1, 1),
+	SOC_DAPM_SINGLE("DMIC Right Input", CS42L73_PWRCTL1,
+			6, 1, 1),
+};
+
+static const char * const cs42l73_ng_delay_text[] = {
+	"50ms", "100ms", "150ms", "200ms" };
+
+static const struct soc_enum ng_delay_enum =
+	SOC_ENUM_SINGLE(CS42L73_NGCAB, 0,
+		ARRAY_SIZE(cs42l73_ng_delay_text), cs42l73_ng_delay_text);
+
+static const char * const charge_pump_freq_text[] = {
+	"0", "1", "2", "3", "4",
+	"5", "6", "7", "8", "9",
+	"10", "11", "12", "13", "14", "15" };
+
+static const struct soc_enum charge_pump_enum =
+	SOC_ENUM_SINGLE(CS42L73_CPFCHC, 4,
+		ARRAY_SIZE(charge_pump_freq_text), charge_pump_freq_text);
+
+static const char * const cs42l73_mono_mix_texts[] = {
+	"Left", "Right", "Mono Mix"};
+
+static const unsigned int cs42l73_mono_mix_values[] = { 0, 1, 2 };
+
+static const struct soc_enum spk_asp_enum =
+	SOC_VALUE_ENUM_SINGLE(CS42L73_MMIXCTL, 6, 1,
+			      ARRAY_SIZE(cs42l73_mono_mix_texts),
+			      cs42l73_mono_mix_texts,
+			      cs42l73_mono_mix_values);
+
+static const struct snd_kcontrol_new spk_asp_mixer =
+	SOC_DAPM_ENUM("Route", spk_asp_enum);
+
+static const struct soc_enum spk_xsp_enum =
+	SOC_VALUE_ENUM_SINGLE(CS42L73_MMIXCTL, 4, 3,
+			      ARRAY_SIZE(cs42l73_mono_mix_texts),
+			      cs42l73_mono_mix_texts,
+			      cs42l73_mono_mix_values);
+
+static const struct snd_kcontrol_new spk_xsp_mixer =
+	SOC_DAPM_ENUM("Route", spk_xsp_enum);
+
+static const struct soc_enum esl_asp_enum =
+	SOC_VALUE_ENUM_SINGLE(CS42L73_MMIXCTL, 2, 5,
+			      ARRAY_SIZE(cs42l73_mono_mix_texts),
+			      cs42l73_mono_mix_texts,
+			      cs42l73_mono_mix_values);
+
+static const struct snd_kcontrol_new esl_asp_mixer =
+	SOC_DAPM_ENUM("Route", esl_asp_enum);
+
+static const struct soc_enum esl_xsp_enum =
+	SOC_VALUE_ENUM_SINGLE(CS42L73_MMIXCTL, 0, 7,
+			      ARRAY_SIZE(cs42l73_mono_mix_texts),
+			      cs42l73_mono_mix_texts,
+			      cs42l73_mono_mix_values);
+
+static const struct snd_kcontrol_new esl_xsp_mixer =
+	SOC_DAPM_ENUM("Route", esl_xsp_enum);
+
+static const char * const cs42l73_ip_swap_text[] = {
+	"Stereo", "Mono A", "Mono B", "Swap A-B"};
+
+static const struct soc_enum ip_swap_enum =
+	SOC_ENUM_SINGLE(CS42L73_MIOPC, 6,
+		ARRAY_SIZE(cs42l73_ip_swap_text), cs42l73_ip_swap_text);
+
+static const char * const cs42l73_spo_mixer_text[] = {"Mono", "Stereo"};
+
+static const struct soc_enum vsp_output_mux_enum =
+	SOC_ENUM_SINGLE(CS42L73_MIXERCTL, 5,
+		ARRAY_SIZE(cs42l73_spo_mixer_text), cs42l73_spo_mixer_text);
+
+static const struct soc_enum xsp_output_mux_enum =
+	SOC_ENUM_SINGLE(CS42L73_MIXERCTL, 4,
+		ARRAY_SIZE(cs42l73_spo_mixer_text), cs42l73_spo_mixer_text);
+
+static const struct snd_kcontrol_new vsp_output_mux =
+	SOC_DAPM_ENUM("Route", vsp_output_mux_enum);
+
+static const struct snd_kcontrol_new xsp_output_mux =
+	SOC_DAPM_ENUM("Route", xsp_output_mux_enum);
+
+static const struct snd_kcontrol_new hp_amp_ctl =
+	SOC_DAPM_SINGLE("Switch", CS42L73_PWRCTL3, 0, 1, 1);
+
+static const struct snd_kcontrol_new lo_amp_ctl =
+	SOC_DAPM_SINGLE("Switch", CS42L73_PWRCTL3, 1, 1, 1);
+
+static const struct snd_kcontrol_new spk_amp_ctl =
+	SOC_DAPM_SINGLE("Switch", CS42L73_PWRCTL3, 2, 1, 1);
+
+static const struct snd_kcontrol_new spklo_amp_ctl =
+	SOC_DAPM_SINGLE("Switch", CS42L73_PWRCTL3, 4, 1, 1);
+
+static const struct snd_kcontrol_new ear_amp_ctl =
+	SOC_DAPM_SINGLE("Switch", CS42L73_PWRCTL3, 3, 1, 1);
+
+static const struct snd_kcontrol_new cs42l73_snd_controls[] = {
+	SOC_DOUBLE_R_SX_TLV("Headphone Analog Playback Volume",
+			CS42L73_HPAAVOL, CS42L73_HPBAVOL, 7,
+			0xffffffC1, 0x0C, hpaloa_tlv),
+
+	SOC_DOUBLE_R_SX_TLV("LineOut Analog Playback Volume", CS42L73_LOAAVOL,
+			CS42L73_LOBAVOL, 7, 0xffffffC1, 0x0C, hpaloa_tlv),
+
+	SOC_DOUBLE_R_SX_TLV("Input PGA Analog Volume", CS42L73_MICAPREPGAAVOL,
+			CS42L73_MICBPREPGABVOL, 5, 0xffffff35,
+			0x34, micpga_tlv),
+
+	SOC_DOUBLE_R("MIC Preamp Switch", CS42L73_MICAPREPGAAVOL,
+			CS42L73_MICBPREPGABVOL, 6, 1, 1),
+
+	SOC_DOUBLE_R_SX_TLV("Input Path Digital Volume", CS42L73_IPADVOL,
+			CS42L73_IPBDVOL, 7, 0xffffffA0, 0xA0, ipd_tlv),
+
+	SOC_DOUBLE_R_SX_TLV("HL Digital Playback Volume",
+			CS42L73_HLADVOL, CS42L73_HLBDVOL, 7, 0xffffffE5,
+			0xE4, hl_tlv),
+
+	SOC_SINGLE_TLV("ADC A Boost Volume",
+			CS42L73_ADCIPC, 2, 0x01, 1, adc_boost_tlv),
+
+	SOC_SINGLE_TLV("ADC B Boost Volume",
+			CS42L73_ADCIPC, 6, 0x01, 1, adc_boost_tlv),
+
+	SOC_SINGLE_TLV("Speakerphone Digital Playback Volume",
+			CS42L73_SPKDVOL, 0, 0xE4, 1, hl_tlv),
+
+	SOC_SINGLE_TLV("Ear Speaker Digital Playback Volume",
+			CS42L73_ESLDVOL, 0, 0xE4, 1, hl_tlv),
+
+	SOC_DOUBLE_R("Headphone Analog Playback Switch", CS42L73_HPAAVOL,
+			CS42L73_HPBAVOL, 7, 1, 1),
+
+	SOC_DOUBLE_R("LineOut Analog Playback Switch", CS42L73_LOAAVOL,
+			CS42L73_LOBAVOL, 7, 1, 1),
+	SOC_DOUBLE("Input Path Digital Switch", CS42L73_ADCIPC, 0, 4, 1, 1),
+	SOC_DOUBLE("HL Digital Playback Switch", CS42L73_PBDC, 0,
+			1, 1, 1),
+	SOC_SINGLE("Speakerphone Digital Playback Switch", CS42L73_PBDC, 2, 1,
+			1),
+	SOC_SINGLE("Ear Speaker Digital Playback Switch", CS42L73_PBDC, 3, 1,
+			1),
+
+	SOC_SINGLE("PGA Soft-Ramp Switch", CS42L73_MIOPC, 3, 1, 0),
+	SOC_SINGLE("Analog Zero Cross Switch", CS42L73_MIOPC, 2, 1, 0),
+	SOC_SINGLE("Digital Soft-Ramp Switch", CS42L73_MIOPC, 1, 1, 0),
+	SOC_SINGLE("Analog Output Soft-Ramp Switch", CS42L73_MIOPC, 0, 1, 0),
+
+	SOC_DOUBLE("ADC Signal Polarity Switch", CS42L73_ADCIPC, 1, 5, 1,
+			0),
+
+	SOC_SINGLE("HL Limiter Attack Rate", CS42L73_LIMARATEHL, 0, 0x3F,
+			0),
+	SOC_SINGLE("HL Limiter Release Rate", CS42L73_LIMRRATEHL, 0,
+			0x3F, 0),
+
+
+	SOC_SINGLE("HL Limiter Switch", CS42L73_LIMRRATEHL, 7, 1, 0),
+	SOC_SINGLE("HL Limiter All Channels Switch", CS42L73_LIMRRATEHL, 6, 1,
+			0),
+
+	SOC_SINGLE_TLV("HL Limiter Max Threshold Volume", CS42L73_LMAXHL, 5, 7,
+			1, limiter_tlv),
+
+	SOC_SINGLE_TLV("HL Limiter Cushion Volume", CS42L73_LMAXHL, 2, 7, 1,
+			limiter_tlv),
+
+	SOC_SINGLE("SPK Limiter Attack Rate Volume", CS42L73_LIMARATESPK, 0,
+			0x3F, 0),
+	SOC_SINGLE("SPK Limiter Release Rate Volume", CS42L73_LIMRRATESPK, 0,
+			0x3F, 0),
+	SOC_SINGLE("SPK Limiter Switch", CS42L73_LIMRRATESPK, 7, 1, 0),
+	SOC_SINGLE("SPK Limiter All Channels Switch", CS42L73_LIMRRATESPK,
+			6, 1, 0),
+	SOC_SINGLE_TLV("SPK Limiter Max Threshold Volume", CS42L73_LMAXSPK, 5,
+			7, 1, limiter_tlv),
+
+	SOC_SINGLE_TLV("SPK Limiter Cushion Volume", CS42L73_LMAXSPK, 2, 7, 1,
+			limiter_tlv),
+
+	SOC_SINGLE("ESL Limiter Attack Rate Volume", CS42L73_LIMARATEESL, 0,
+			0x3F, 0),
+	SOC_SINGLE("ESL Limiter Release Rate Volume", CS42L73_LIMRRATEESL, 0,
+			0x3F, 0),
+	SOC_SINGLE("ESL Limiter Switch", CS42L73_LIMRRATEESL, 7, 1, 0),
+	SOC_SINGLE_TLV("ESL Limiter Max Threshold Volume", CS42L73_LMAXESL, 5,
+			7, 1, limiter_tlv),
+
+	SOC_SINGLE_TLV("ESL Limiter Cushion Volume", CS42L73_LMAXESL, 2, 7, 1,
+			limiter_tlv),
+
+	SOC_SINGLE("ALC Attack Rate Volume", CS42L73_ALCARATE, 0, 0x3F, 0),
+	SOC_SINGLE("ALC Release Rate Volume", CS42L73_ALCRRATE, 0, 0x3F, 0),
+	SOC_DOUBLE("ALC Switch", CS42L73_ALCARATE, 6, 7, 1, 0),
+	SOC_SINGLE_TLV("ALC Max Threshold Volume", CS42L73_ALCMINMAX, 5, 7, 0,
+			limiter_tlv),
+	SOC_SINGLE_TLV("ALC Min Threshold Volume", CS42L73_ALCMINMAX, 2, 7, 0,
+			limiter_tlv),
+
+	SOC_DOUBLE("NG Enable Switch", CS42L73_NGCAB, 6, 7, 1, 0),
+	SOC_SINGLE("NG Boost Switch", CS42L73_NGCAB, 5, 1, 0),
+	/*
+	    NG Threshold depends on NG_BOOTSAB, which selects
+	    between two threshold scales in decibels.
+	    Set linear values for now ..
+	*/
+	SOC_SINGLE("NG Threshold", CS42L73_NGCAB, 2, 7, 0),
+	SOC_ENUM("NG Delay", ng_delay_enum),
+
+	SOC_ENUM("Charge Pump Frequency", charge_pump_enum),
+
+	SOC_DOUBLE_R_TLV("XSP-IP Volume",
+			CS42L73_XSPAIPAA, CS42L73_XSPBIPBA, 0, 0x3F, 1,
+			attn_tlv),
+	SOC_DOUBLE_R_TLV("XSP-XSP Volume",
+			CS42L73_XSPAXSPAA, CS42L73_XSPBXSPBA, 0, 0x3F, 1,
+			attn_tlv),
+	SOC_DOUBLE_R_TLV("XSP-ASP Volume",
+			CS42L73_XSPAASPAA, CS42L73_XSPAASPBA, 0, 0x3F, 1,
+			attn_tlv),
+	SOC_DOUBLE_R_TLV("XSP-VSP Volume",
+			CS42L73_XSPAVSPMA, CS42L73_XSPBVSPMA, 0, 0x3F, 1,
+			attn_tlv),
+
+	SOC_DOUBLE_R_TLV("ASP-IP Volume",
+			CS42L73_ASPAIPAA, CS42L73_ASPBIPBA, 0, 0x3F, 1,
+			attn_tlv),
+	SOC_DOUBLE_R_TLV("ASP-XSP Volume",
+			CS42L73_ASPAXSPAA, CS42L73_ASPBXSPBA, 0, 0x3F, 1,
+			attn_tlv),
+	SOC_DOUBLE_R_TLV("ASP-ASP Volume",
+			CS42L73_ASPAASPAA, CS42L73_ASPBASPBA, 0, 0x3F, 1,
+			attn_tlv),
+	SOC_DOUBLE_R_TLV("ASP-VSP Volume",
+			CS42L73_ASPAVSPMA, CS42L73_ASPBVSPMA, 0, 0x3F, 1,
+			attn_tlv),
+
+	SOC_DOUBLE_R_TLV("VSP-IP Volume",
+			CS42L73_VSPAIPAA, CS42L73_VSPBIPBA, 0, 0x3F, 1,
+			attn_tlv),
+	SOC_DOUBLE_R_TLV("VSP-XSP Volume",
+			CS42L73_VSPAXSPAA, CS42L73_VSPBXSPBA, 0, 0x3F, 1,
+			attn_tlv),
+	SOC_DOUBLE_R_TLV("VSP-ASP Volume",
+			CS42L73_VSPAASPAA, CS42L73_VSPBASPBA, 0, 0x3F, 1,
+			attn_tlv),
+	SOC_DOUBLE_R_TLV("VSP-VSP Volume",
+			CS42L73_VSPAVSPMA, CS42L73_VSPBVSPMA, 0, 0x3F, 1,
+			attn_tlv),
+
+	SOC_DOUBLE_R_TLV("HL-IP Volume",
+			CS42L73_HLAIPAA, CS42L73_HLBIPBA, 0, 0x3F, 1,
+			attn_tlv),
+	SOC_DOUBLE_R_TLV("HL-XSP Volume",
+			CS42L73_HLAXSPAA, CS42L73_HLBXSPBA, 0, 0x3F, 1,
+			attn_tlv),
+	SOC_DOUBLE_R_TLV("HL-ASP Volume",
+			CS42L73_HLAASPAA, CS42L73_HLBASPBA, 0, 0x3F, 1,
+			attn_tlv),
+	SOC_DOUBLE_R_TLV("HL-VSP Volume",
+			CS42L73_HLAVSPMA, CS42L73_HLBVSPMA, 0, 0x3F, 1,
+			attn_tlv),
+
+	SOC_SINGLE_TLV("SPK-IP Mono Volume",
+			CS42L73_SPKMIPMA, 0, 0x3E, 1, attn_tlv),
+	SOC_SINGLE_TLV("SPK-XSP Mono Volume",
+			CS42L73_SPKMXSPA, 0, 0x3E, 1, attn_tlv),
+	SOC_SINGLE_TLV("SPK-ASP Mono Volume",
+			CS42L73_SPKMASPA, 0, 0x3E, 1, attn_tlv),
+	SOC_SINGLE_TLV("SPK-VSP Mono Volume",
+			CS42L73_SPKMVSPMA, 0, 0x3E, 1, attn_tlv),
+
+	SOC_SINGLE_TLV("ESL-IP Mono Volume",
+			CS42L73_ESLMIPMA, 0, 0x3E, 1, attn_tlv),
+	SOC_SINGLE_TLV("ESL-XSP Mono Volume",
+			CS42L73_ESLMXSPA, 0, 0x3E, 1, attn_tlv),
+	SOC_SINGLE_TLV("ESL-ASP Mono Volume",
+			CS42L73_ESLMASPA, 0, 0x3E, 1, attn_tlv),
+	SOC_SINGLE_TLV("ESL-VSP Mono Volume",
+			CS42L73_ESLMVSPMA, 0, 0x3E, 1, attn_tlv),
+
+	SOC_ENUM("IP Digital Swap/Mono Select", ip_swap_enum),
+
+	SOC_ENUM("VSPOUT Mono/Stereo Select", vsp_output_mux_enum),
+	SOC_ENUM("XSPOUT Mono/Stereo Select", xsp_output_mux_enum),
+};
+
+static const struct snd_soc_dapm_widget cs42l73_dapm_widgets[] = {
+	SND_SOC_DAPM_INPUT("LINEINA"),
+	SND_SOC_DAPM_INPUT("LINEINB"),
+	SND_SOC_DAPM_INPUT("MIC1"),
+	SND_SOC_DAPM_SUPPLY("MIC1 Bias", CS42L73_PWRCTL2, 6, 1, NULL, 0),
+	SND_SOC_DAPM_INPUT("MIC2"),
+	SND_SOC_DAPM_SUPPLY("MIC2 Bias", CS42L73_PWRCTL2, 7, 1, NULL, 0),
+
+	SND_SOC_DAPM_AIF_OUT("XSPOUTL", "XSP Capture",  0,
+			CS42L73_PWRCTL2, 1, 1),
+	SND_SOC_DAPM_AIF_OUT("XSPOUTR", "XSP Capture",  0,
+			CS42L73_PWRCTL2, 1, 1),
+	SND_SOC_DAPM_AIF_OUT("ASPOUTL", "ASP Capture",  0,
+			CS42L73_PWRCTL2, 3, 1),
+	SND_SOC_DAPM_AIF_OUT("ASPOUTR", "ASP Capture",  0,
+			CS42L73_PWRCTL2, 3, 1),
+	SND_SOC_DAPM_AIF_OUT("VSPOUTL", "VSP Capture",  0,
+			CS42L73_PWRCTL2, 4, 1),
+	SND_SOC_DAPM_AIF_OUT("VSPOUTR", "VSP Capture",  0,
+			CS42L73_PWRCTL2, 4, 1),
+
+	SND_SOC_DAPM_PGA("PGA Left", SND_SOC_NOPM, 0, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("PGA Right", SND_SOC_NOPM, 0, 0, NULL, 0),
+
+	SND_SOC_DAPM_MUX("PGA Left Mux", SND_SOC_NOPM, 0, 0, &pgaa_mux),
+	SND_SOC_DAPM_MUX("PGA Right Mux", SND_SOC_NOPM, 0, 0, &pgab_mux),
+
+	SND_SOC_DAPM_ADC("ADC Left", NULL, CS42L73_PWRCTL1, 7, 1),
+	SND_SOC_DAPM_ADC("ADC Right", NULL, CS42L73_PWRCTL1, 5, 1),
+	SND_SOC_DAPM_ADC("DMIC Left", NULL, CS42L73_PWRCTL1, 6, 1),
+	SND_SOC_DAPM_ADC("DMIC Right", NULL, CS42L73_PWRCTL1, 4, 1),
+
+	SND_SOC_DAPM_MIXER_NAMED_CTL("Input Left Capture", SND_SOC_NOPM,
+			 0, 0, input_left_mixer,
+			 ARRAY_SIZE(input_left_mixer)),
+
+	SND_SOC_DAPM_MIXER_NAMED_CTL("Input Right Capture", SND_SOC_NOPM,
+			0, 0, input_right_mixer,
+			ARRAY_SIZE(input_right_mixer)),
+
+	SND_SOC_DAPM_MIXER("ASPL Output Mixer", SND_SOC_NOPM, 0, 0, NULL, 0),
+	SND_SOC_DAPM_MIXER("ASPR Output Mixer", SND_SOC_NOPM, 0, 0, NULL, 0),
+	SND_SOC_DAPM_MIXER("XSPL Output Mixer", SND_SOC_NOPM, 0, 0, NULL, 0),
+	SND_SOC_DAPM_MIXER("XSPR Output Mixer", SND_SOC_NOPM, 0, 0, NULL, 0),
+	SND_SOC_DAPM_MIXER("VSPL Output Mixer", SND_SOC_NOPM, 0, 0, NULL, 0),
+	SND_SOC_DAPM_MIXER("VSPR Output Mixer", SND_SOC_NOPM, 0, 0, NULL, 0),
+
+	SND_SOC_DAPM_AIF_IN("XSPINL", "XSP Playback", 0,
+				CS42L73_PWRCTL2, 0, 1),
+	SND_SOC_DAPM_AIF_IN("XSPINR", "XSP Playback", 0,
+				CS42L73_PWRCTL2, 0, 1),
+	SND_SOC_DAPM_AIF_IN("XSPINM", "XSP Playback", 0,
+				CS42L73_PWRCTL2, 0, 1),
+
+	SND_SOC_DAPM_AIF_IN("ASPINL", "ASP Playback", 0,
+				CS42L73_PWRCTL2, 2, 1),
+	SND_SOC_DAPM_AIF_IN("ASPINR", "ASP Playback", 0,
+				CS42L73_PWRCTL2, 2, 1),
+	SND_SOC_DAPM_AIF_IN("ASPINM", "ASP Playback", 0,
+				CS42L73_PWRCTL2, 2, 1),
+
+	SND_SOC_DAPM_AIF_IN("VSPIN", "VSP Playback", 0,
+				CS42L73_PWRCTL2, 4, 1),
+
+	SND_SOC_DAPM_MIXER("HL Left Mixer", SND_SOC_NOPM, 0, 0, NULL, 0),
+	SND_SOC_DAPM_MIXER("HL Right Mixer", SND_SOC_NOPM, 0, 0, NULL, 0),
+	SND_SOC_DAPM_MIXER("SPK Mixer", SND_SOC_NOPM, 0, 0, NULL, 0),
+	SND_SOC_DAPM_MIXER("ESL Mixer", SND_SOC_NOPM, 0, 0, NULL, 0),
+
+	SND_SOC_DAPM_MUX("ESL-XSP Mux", SND_SOC_NOPM,
+			 0, 0, &esl_xsp_mixer),
+
+	SND_SOC_DAPM_MUX("ESL-ASP Mux", SND_SOC_NOPM,
+			 0, 0, &esl_asp_mixer),
+
+	SND_SOC_DAPM_MUX("SPK-ASP Mux", SND_SOC_NOPM,
+			 0, 0, &spk_asp_mixer),
+
+	SND_SOC_DAPM_MUX("SPK-XSP Mux", SND_SOC_NOPM,
+			 0, 0, &spk_xsp_mixer),
+
+	SND_SOC_DAPM_PGA("HL Left DAC", SND_SOC_NOPM, 0, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("HL Right DAC", SND_SOC_NOPM, 0, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("SPK DAC", SND_SOC_NOPM, 0, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("ESL DAC", SND_SOC_NOPM, 0, 0, NULL, 0),
+
+	SND_SOC_DAPM_SWITCH("HP Amp", CS42L73_PWRCTL3, 0, 1,
+			    &hp_amp_ctl),
+	SND_SOC_DAPM_SWITCH("LO Amp", CS42L73_PWRCTL3, 1, 1,
+			    &lo_amp_ctl),
+	SND_SOC_DAPM_SWITCH("SPK Amp", CS42L73_PWRCTL3, 2, 1,
+			    &spk_amp_ctl),
+	SND_SOC_DAPM_SWITCH("EAR Amp", CS42L73_PWRCTL3, 3, 1,
+			    &ear_amp_ctl),
+	SND_SOC_DAPM_SWITCH("SPKLO Amp", CS42L73_PWRCTL3, 4, 1,
+			    &spklo_amp_ctl),
+
+	SND_SOC_DAPM_OUTPUT("HPOUTA"),
+	SND_SOC_DAPM_OUTPUT("HPOUTB"),
+	SND_SOC_DAPM_OUTPUT("LINEOUTA"),
+	SND_SOC_DAPM_OUTPUT("LINEOUTB"),
+	SND_SOC_DAPM_OUTPUT("EAROUT"),
+	SND_SOC_DAPM_OUTPUT("SPKOUT"),
+	SND_SOC_DAPM_OUTPUT("SPKLINEOUT"),
+};
+
+static const struct snd_soc_dapm_route cs42l73_audio_map[] = {
+
+	/* SPKLO EARSPK Paths */
+	{"EAROUT", NULL, "EAR Amp"},
+	{"SPKLINEOUT", NULL, "SPKLO Amp"},
+
+	{"EAR Amp", "Switch", "ESL DAC"},
+	{"SPKLO Amp", "Switch", "ESL DAC"},
+
+	{"ESL DAC", "ESL-ASP Mono Volume", "ESL Mixer"},
+	{"ESL DAC", "ESL-XSP Mono Volume", "ESL Mixer"},
+	{"ESL DAC", "ESL-VSP Mono Volume", "VSPIN"},
+	/* Loopback */
+	{"ESL DAC", "ESL-IP Mono Volume", "Input Left Capture"},
+	{"ESL DAC", "ESL-IP Mono Volume", "Input Right Capture"},
+
+	{"ESL Mixer", NULL, "ESL-ASP Mux"},
+	{"ESL Mixer", NULL, "ESL-XSP Mux"},
+
+	{"ESL-ASP Mux", "Left", "ASPINL"},
+	{"ESL-ASP Mux", "Right", "ASPINR"},
+	{"ESL-ASP Mux", "Mono Mix", "ASPINM"},
+
+	{"ESL-XSP Mux", "Left", "XSPINL"},
+	{"ESL-XSP Mux", "Right", "XSPINR"},
+	{"ESL-XSP Mux", "Mono Mix", "XSPINM"},
+
+	/* Speakerphone Paths */
+	{"SPKOUT", NULL, "SPK Amp"},
+	{"SPK Amp", "Switch", "SPK DAC"},
+
+	{"SPK DAC", "SPK-ASP Mono Volume", "SPK Mixer"},
+	{"SPK DAC", "SPK-XSP Mono Volume", "SPK Mixer"},
+	{"SPK DAC", "SPK-VSP Mono Volume", "VSPIN"},
+	/* Loopback */
+	{"SPK DAC", "SPK-IP Mono Volume", "Input Left Capture"},
+	{"SPK DAC", "SPK-IP Mono Volume", "Input Right Capture"},
+
+	{"SPK Mixer", NULL, "SPK-ASP Mux"},
+	{"SPK Mixer", NULL, "SPK-XSP Mux"},
+
+	{"SPK-ASP Mux", "Left", "ASPINL"},
+	{"SPK-ASP Mux", "Mono Mix", "ASPINM"},
+	{"SPK-ASP Mux", "Right", "ASPINR"},
+
+	{"SPK-XSP Mux", "Left", "XSPINL"},
+	{"SPK-XSP Mux", "Mono Mix", "XSPINM"},
+	{"SPK-XSP Mux", "Right", "XSPINR"},
+
+	/* HP LineOUT Paths */
+	{"HPOUTA", NULL, "HP Amp"},
+	{"HPOUTB", NULL, "HP Amp"},
+	{"LINEOUTA", NULL, "LO Amp"},
+	{"LINEOUTB", NULL, "LO Amp"},
+
+	{"HP Amp", "Switch", "HL Left DAC"},
+	{"HP Amp", "Switch", "HL Right DAC"},
+	{"LO Amp", "Switch", "HL Left DAC"},
+	{"LO Amp", "Switch", "HL Right DAC"},
+
+	{"HL Left DAC", "HL-XSP Volume", "HL Left Mixer"},
+	{"HL Right DAC", "HL-XSP Volume", "HL Right Mixer"},
+	{"HL Left DAC", "HL-ASP Volume", "HL Left Mixer"},
+	{"HL Right DAC", "HL-ASP Volume", "HL Right Mixer"},
+	{"HL Left DAC", "HL-VSP Volume", "HL Left Mixer"},
+	{"HL Right DAC", "HL-VSP Volume", "HL Right Mixer"},
+	/* Loopback */
+	{"HL Left DAC", "HL-IP Volume", "HL Left Mixer"},
+	{"HL Right DAC", "HL-IP Volume", "HL Right Mixer"},
+	{"HL Left Mixer", NULL, "Input Left Capture"},
+	{"HL Right Mixer", NULL, "Input Right Capture"},
+
+	{"HL Left Mixer", NULL, "ASPINL"},
+	{"HL Right Mixer", NULL, "ASPINR"},
+	{"HL Left Mixer", NULL, "XSPINL"},
+	{"HL Right Mixer", NULL, "XSPINR"},
+	{"HL Left Mixer", NULL, "VSPIN"},
+	{"HL Right Mixer", NULL, "VSPIN"},
+
+	/* Capture Paths */
+	{"MIC1", NULL, "MIC1 Bias"},
+	{"PGA Left Mux", "Mic 1", "MIC1"},
+	{"MIC2", NULL, "MIC2 Bias"},
+	{"PGA Right Mux", "Mic 2", "MIC2"},
+
+	{"PGA Left Mux", "Line A", "LINEINA"},
+	{"PGA Right Mux", "Line B", "LINEINB"},
+
+	{"PGA Left", NULL, "PGA Left Mux"},
+	{"PGA Right", NULL, "PGA Right Mux"},
+
+	{"ADC Left", NULL, "PGA Left"},
+	{"ADC Right", NULL, "PGA Right"},
+
+	{"Input Left Capture", "ADC Left Input", "ADC Left"},
+	{"Input Right Capture", "ADC Right Input", "ADC Right"},
+	{"Input Left Capture", "DMIC Left Input", "DMIC Left"},
+	{"Input Right Capture", "DMIC Right Input", "DMIC Right"},
+
+	/* Audio Capture */
+	{"ASPL Output Mixer", NULL, "Input Left Capture"},
+	{"ASPR Output Mixer", NULL, "Input Right Capture"},
+
+	{"ASPOUTL", "ASP-IP Volume", "ASPL Output Mixer"},
+	{"ASPOUTR", "ASP-IP Volume", "ASPR Output Mixer"},
+
+	/* Auxillary Capture */
+	{"XSPL Output Mixer", NULL, "Input Left Capture"},
+	{"XSPR Output Mixer", NULL, "Input Right Capture"},
+
+	{"XSPOUTL", "XSP-IP Volume", "XSPL Output Mixer"},
+	{"XSPOUTR", "XSP-IP Volume", "XSPR Output Mixer"},
+
+	{"XSPOUTL", NULL, "XSPL Output Mixer"},
+	{"XSPOUTR", NULL, "XSPR Output Mixer"},
+
+	/* Voice Capture */
+	{"VSPL Output Mixer", NULL, "Input Left Capture"},
+	{"VSPR Output Mixer", NULL, "Input Left Capture"},
+
+	{"VSPOUTL", "VSP-IP Volume", "VSPL Output Mixer"},
+	{"VSPOUTR", "VSP-IP Volume", "VSPR Output Mixer"},
+
+	{"VSPOUTL", NULL, "VSPL Output Mixer"},
+	{"VSPOUTR", NULL, "VSPR Output Mixer"},
+};
+
+struct cs42l73_mclk_div {
+	u32 mclk;
+	u32 srate;
+	u8 mmcc;
+};
+
+static struct cs42l73_mclk_div cs42l73_mclk_coeffs[] = {
+	/* MCLK, Sample Rate, xMMCC[5:0] */
+	{5644800, 11025, 0x30},
+	{5644800, 22050, 0x20},
+	{5644800, 44100, 0x10},
+
+	{6000000,  8000, 0x39},
+	{6000000, 11025, 0x33},
+	{6000000, 12000, 0x31},
+	{6000000, 16000, 0x29},
+	{6000000, 22050, 0x23},
+	{6000000, 24000, 0x21},
+	{6000000, 32000, 0x19},
+	{6000000, 44100, 0x13},
+	{6000000, 48000, 0x11},
+
+	{6144000,  8000, 0x38},
+	{6144000, 12000, 0x30},
+	{6144000, 16000, 0x28},
+	{6144000, 24000, 0x20},
+	{6144000, 32000, 0x18},
+	{6144000, 48000, 0x10},
+
+	{6500000,  8000, 0x3C},
+	{6500000, 11025, 0x35},
+	{6500000, 12000, 0x34},
+	{6500000, 16000, 0x2C},
+	{6500000, 22050, 0x25},
+	{6500000, 24000, 0x24},
+	{6500000, 32000, 0x1C},
+	{6500000, 44100, 0x15},
+	{6500000, 48000, 0x14},
+
+	{6400000,  8000, 0x3E},
+	{6400000, 11025, 0x37},
+	{6400000, 12000, 0x36},
+	{6400000, 16000, 0x2E},
+	{6400000, 22050, 0x27},
+	{6400000, 24000, 0x26},
+	{6400000, 32000, 0x1E},
+	{6400000, 44100, 0x17},
+	{6400000, 48000, 0x16},
+};
+
+struct cs42l73_mclkx_div {
+	u32 mclkx;
+	u8 ratio;
+	u8 mclkdiv;
+};
+
+static struct cs42l73_mclkx_div cs42l73_mclkx_coeffs[] = {
+	{5644800,  1, 0},	/* 5644800 */
+	{6000000,  1, 0},	/* 6000000 */
+	{6144000,  1, 0},	/* 6144000 */
+	{11289600, 2, 2},	/* 5644800 */
+	{12288000, 2, 2},	/* 6144000 */
+	{12000000, 2, 2},	/* 6000000 */
+	{13000000, 2, 2},	/* 6500000 */
+	{19200000, 3, 3},	/* 6400000 */
+	{24000000, 4, 4},	/* 6000000 */
+	{26000000, 4, 4},	/* 6500000 */
+	{38400000, 6, 5}	/* 6400000 */
+};
+
+static int cs42l73_get_mclkx_coeff(int mclkx)
+{
+	int i;
+
+	for (i = 0; i < ARRAY_SIZE(cs42l73_mclkx_coeffs); i++) {
+		if (cs42l73_mclkx_coeffs[i].mclkx == mclkx)
+			return i;
+	}
+	return -EINVAL;
+}
+
+static int cs42l73_get_mclk_coeff(int mclk, int srate)
+{
+	int i;
+
+	for (i = 0; i < ARRAY_SIZE(cs42l73_mclk_coeffs); i++) {
+		if (cs42l73_mclk_coeffs[i].mclk == mclk &&
+		    cs42l73_mclk_coeffs[i].srate == srate)
+			return i;
+	}
+	return -EINVAL;
+
+}
+
+static int cs42l73_set_mclk(struct snd_soc_dai *dai, unsigned int freq)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	struct cs42l73_private *priv = snd_soc_codec_get_drvdata(codec);
+
+	int mclkx_coeff;
+	u32 mclk = 0;
+	u8 dmmcc = 0;
+
+	/* MCLKX -> MCLK */
+	mclkx_coeff = cs42l73_get_mclkx_coeff(freq);
+
+	mclk = cs42l73_mclkx_coeffs[mclkx_coeff].mclkx /
+		cs42l73_mclkx_coeffs[mclkx_coeff].ratio;
+
+	dev_dbg(codec->dev, "MCLK%u %u  <-> internal MCLK %u\n",
+		 priv->mclksel + 1, cs42l73_mclkx_coeffs[mclkx_coeff].mclkx,
+		 mclk);
+
+	dmmcc = (priv->mclksel << 4) |
+		(cs42l73_mclkx_coeffs[mclkx_coeff].mclkdiv << 1);
+
+	snd_soc_write(codec, CS42L73_DMMCC, dmmcc);
+
+	priv->sysclk = mclkx_coeff;
+	priv->mclk = mclk;
+
+	return 0;
+}
+
+static int cs42l73_set_sysclk(struct snd_soc_dai *dai,
+			      int clk_id, unsigned int freq, int dir)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	struct cs42l73_private *priv = snd_soc_codec_get_drvdata(codec);
+
+	switch (clk_id) {
+	case CS42L73_CLKID_MCLK1:
+		break;
+	case CS42L73_CLKID_MCLK2:
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	if ((cs42l73_set_mclk(dai, freq)) < 0) {
+		dev_err(codec->dev, "Unable to set MCLK for dai %s\n",
+			dai->name);
+		return -EINVAL;
+	}
+
+	priv->mclksel = clk_id;
+
+	return 0;
+}
+
+static int cs42l73_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	struct cs42l73_private *priv = snd_soc_codec_get_drvdata(codec);
+	u8 id = codec_dai->id;
+	u8 inv, format;
+	u8 spc, mmcc;
+
+	spc = snd_soc_read(codec, CS42L73_SPC(id));
+	mmcc = snd_soc_read(codec, CS42L73_MMCC(id));
+
+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBM_CFM:
+		mmcc |= MS_MASTER;
+		break;
+
+	case SND_SOC_DAIFMT_CBS_CFS:
+		mmcc &= ~MS_MASTER;
+		break;
+
+	default:
+		return -EINVAL;
+	}
+
+	format = (fmt & SND_SOC_DAIFMT_FORMAT_MASK);
+	inv = (fmt & SND_SOC_DAIFMT_INV_MASK);
+
+	switch (format) {
+	case SND_SOC_DAIFMT_I2S:
+		spc &= ~SPDIF_PCM;
+		break;
+	case SND_SOC_DAIFMT_DSP_A:
+	case SND_SOC_DAIFMT_DSP_B:
+		if (mmcc & MS_MASTER) {
+			dev_err(codec->dev,
+				"PCM format in slave mode only\n");
+			return -EINVAL;
+		}
+		if (id == CS42L73_ASP) {
+			dev_err(codec->dev,
+				"PCM format is not supported on ASP port\n");
+			return -EINVAL;
+		}
+		spc |= SPDIF_PCM;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	if (spc & SPDIF_PCM) {
+		spc &= (31 << 3);	/* Clear PCM mode, set MSB->LSB */
+		switch (format) {
+		case SND_SOC_DAIFMT_DSP_B:
+			if (inv == SND_SOC_DAIFMT_IB_IF)
+				spc |= (PCM_MODE0 << 4);
+			if (inv == SND_SOC_DAIFMT_IB_NF)
+				spc |= (PCM_MODE1 << 4);
+		break;
+		case SND_SOC_DAIFMT_DSP_A:
+			if (inv == SND_SOC_DAIFMT_IB_IF)
+				spc |= (PCM_MODE1 << 4);
+			break;
+		default:
+			return -EINVAL;
+		}
+	}
+
+	priv->config[id].spc = spc;
+	priv->config[id].mmcc = mmcc;
+
+	return 0;
+}
+
+static u32 cs42l73_asrc_rates[] = {
+	8000, 11025, 12000, 16000, 22050,
+	24000, 32000, 44100, 48000
+};
+
+static unsigned int cs42l73_get_xspfs_coeff(u32 rate)
+{
+	int i;
+	for (i = 0; i < ARRAY_SIZE(cs42l73_asrc_rates); i++) {
+		if (cs42l73_asrc_rates[i] == rate)
+			return i + 1;
+	}
+	return 0;		/* 0 = Don't know */
+}
+
+static void cs42l73_update_asrc(struct snd_soc_codec *codec, int id, int srate)
+{
+	u8 spfs = 0;
+
+	if (srate > 0)
+		spfs = cs42l73_get_xspfs_coeff(srate);
+
+	switch (id) {
+	case CS42L73_XSP:
+		snd_soc_update_bits(codec, CS42L73_VXSPFS, 0x0f, spfs);
+	break;
+	case CS42L73_ASP:
+		snd_soc_update_bits(codec, CS42L73_ASPC, 0x3c, spfs << 2);
+	break;
+	case CS42L73_VSP:
+		snd_soc_update_bits(codec, CS42L73_VXSPFS, 0xf0, spfs << 4);
+	break;
+	default:
+	break;
+	}
+}
+
+static int cs42l73_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 cs42l73_private *priv = snd_soc_codec_get_drvdata(codec);
+	int id = dai->id;
+	int mclk_coeff;
+	int srate = params_rate(params);
+
+	if (priv->config[id].mmcc & MS_MASTER) {
+		/* CS42L73 Master */
+		/* MCLK -> srate */
+		mclk_coeff =
+		    cs42l73_get_mclk_coeff(priv->mclk, srate);
+
+		if (mclk_coeff < 0)
+			return -EINVAL;
+
+		dev_dbg(codec->dev,
+			 "DAI[%d]: MCLK %u, srate %u, MMCC[5:0] = %x\n",
+			 id, priv->mclk, srate,
+			 cs42l73_mclk_coeffs[mclk_coeff].mmcc);
+
+		priv->config[id].mmcc &= 0xC0;
+		priv->config[id].mmcc |= cs42l73_mclk_coeffs[mclk_coeff].mmcc;
+		priv->config[id].spc &= 0xFC;
+		priv->config[id].spc &= MCK_SCLK_64FS;
+	} else {
+		/* CS42L73 Slave */
+		priv->config[id].spc &= 0xFC;
+		priv->config[id].spc |= MCK_SCLK_64FS;
+	}
+	/* Update ASRCs */
+	priv->config[id].srate = srate;
+
+	snd_soc_write(codec, CS42L73_SPC(id), priv->config[id].spc);
+	snd_soc_write(codec, CS42L73_MMCC(id), priv->config[id].mmcc);
+
+	cs42l73_update_asrc(codec, id, srate);
+
+	return 0;
+}
+
+static int cs42l73_set_bias_level(struct snd_soc_codec *codec,
+				  enum snd_soc_bias_level level)
+{
+	struct cs42l73_private *cs42l73 = snd_soc_codec_get_drvdata(codec);
+
+	switch (level) {
+	case SND_SOC_BIAS_ON:
+		snd_soc_update_bits(codec, CS42L73_DMMCC, MCLKDIS, 0);
+		snd_soc_update_bits(codec, CS42L73_PWRCTL1, PDN, 0);
+		break;
+
+	case SND_SOC_BIAS_PREPARE:
+		break;
+
+	case SND_SOC_BIAS_STANDBY:
+		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
+			regcache_cache_only(cs42l73->regmap, false);
+			regcache_sync(cs42l73->regmap);
+		}
+		snd_soc_update_bits(codec, CS42L73_PWRCTL1, PDN, 1);
+		break;
+
+	case SND_SOC_BIAS_OFF:
+		snd_soc_update_bits(codec, CS42L73_PWRCTL1, PDN, 1);
+		snd_soc_update_bits(codec, CS42L73_DMMCC, MCLKDIS, 1);
+		break;
+	}
+	codec->dapm.bias_level = level;
+	return 0;
+}
+
+static int cs42l73_set_tristate(struct snd_soc_dai *dai, int tristate)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	int id = dai->id;
+
+	return snd_soc_update_bits(codec, CS42L73_SPC(id),
+					0x7F, tristate << 7);
+}
+
+static struct snd_pcm_hw_constraint_list constraints_12_24 = {
+	.count  = ARRAY_SIZE(cs42l73_asrc_rates),
+	.list   = cs42l73_asrc_rates,
+};
+
+static int cs42l73_pcm_startup(struct snd_pcm_substream *substream,
+			       struct snd_soc_dai *dai)
+{
+	snd_pcm_hw_constraint_list(substream->runtime, 0,
+					SNDRV_PCM_HW_PARAM_RATE,
+					&constraints_12_24);
+	return 0;
+}
+
+/* SNDRV_PCM_RATE_KNOT -> 12000, 24000 Hz, limit with constraint list */
+#define CS42L73_RATES (SNDRV_PCM_RATE_8000_48000 | SNDRV_PCM_RATE_KNOT)
+
+
+#define CS42L73_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
+	SNDRV_PCM_FMTBIT_S24_LE)
+
+static const struct snd_soc_dai_ops cs42l73_ops = {
+	.startup = cs42l73_pcm_startup,
+	.hw_params = cs42l73_pcm_hw_params,
+	.set_fmt = cs42l73_set_dai_fmt,
+	.set_sysclk = cs42l73_set_sysclk,
+	.set_tristate = cs42l73_set_tristate,
+};
+
+static struct snd_soc_dai_driver cs42l73_dai[] = {
+	{
+		.name = "cs42l73-xsp",
+		.id = CS42L73_XSP,
+		.playback = {
+			.stream_name = "XSP Playback",
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = CS42L73_RATES,
+			.formats = CS42L73_FORMATS,
+		},
+		.capture = {
+			.stream_name = "XSP Capture",
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = CS42L73_RATES,
+			.formats = CS42L73_FORMATS,
+		},
+		.ops = &cs42l73_ops,
+		.symmetric_rates = 1,
+	 },
+	{
+		.name = "cs42l73-asp",
+		.id = CS42L73_ASP,
+		.playback = {
+			.stream_name = "ASP Playback",
+			.channels_min = 2,
+			.channels_max = 2,
+			.rates = CS42L73_RATES,
+			.formats = CS42L73_FORMATS,
+		},
+		.capture = {
+			.stream_name = "ASP Capture",
+			.channels_min = 2,
+			.channels_max = 2,
+			.rates = CS42L73_RATES,
+			.formats = CS42L73_FORMATS,
+		},
+		.ops = &cs42l73_ops,
+		.symmetric_rates = 1,
+	 },
+	{
+		.name = "cs42l73-vsp",
+		.id = CS42L73_VSP,
+		.playback = {
+			.stream_name = "VSP Playback",
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = CS42L73_RATES,
+			.formats = CS42L73_FORMATS,
+		},
+		.capture = {
+			.stream_name = "VSP Capture",
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = CS42L73_RATES,
+			.formats = CS42L73_FORMATS,
+		},
+		.ops = &cs42l73_ops,
+		.symmetric_rates = 1,
+	 }
+};
+
+static int cs42l73_suspend(struct snd_soc_codec *codec, pm_message_t state)
+{
+	cs42l73_set_bias_level(codec, SND_SOC_BIAS_OFF);
+
+	return 0;
+}
+
+static int cs42l73_resume(struct snd_soc_codec *codec)
+{
+
+	struct cs42l73_private *cs42l73 = snd_soc_codec_get_drvdata(codec);
+	regcache_sync(cs42l73->regmap);
+
+	cs42l73_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+	return 0;
+}
+
+static int cs42l73_probe(struct snd_soc_codec *codec)
+{
+	int ret;
+	struct cs42l73_private *cs42l73 = snd_soc_codec_get_drvdata(codec);
+
+	codec->control_data = cs42l73->regmap;
+
+	ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP);
+	if (ret < 0) {
+		dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
+		return ret;
+	}
+
+	regcache_cache_only(cs42l73->regmap, true);
+
+	cs42l73_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+	cs42l73->mclksel = CS42L73_CLKID_MCLK1;	/* MCLK1 as master clk */
+	cs42l73->mclk = 0;
+
+	return ret;
+}
+
+static int cs42l73_remove(struct snd_soc_codec *codec)
+{
+	cs42l73_set_bias_level(codec, SND_SOC_BIAS_OFF);
+	return 0;
+}
+
+static struct snd_soc_codec_driver soc_codec_dev_cs42l73 = {
+	.probe = cs42l73_probe,
+	.remove = cs42l73_remove,
+	.suspend = cs42l73_suspend,
+	.resume = cs42l73_resume,
+	.set_bias_level = cs42l73_set_bias_level,
+
+	.dapm_widgets = cs42l73_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(cs42l73_dapm_widgets),
+	.dapm_routes = cs42l73_audio_map,
+	.num_dapm_routes = ARRAY_SIZE(cs42l73_audio_map),
+
+	.controls = cs42l73_snd_controls,
+	.num_controls = ARRAY_SIZE(cs42l73_snd_controls),
+};
+
+static struct regmap_config cs42l73_regmap = {
+	.reg_bits = 8,
+	.val_bits = 8,
+
+	.max_register = CS42L73_MAX_REGISTER,
+	.reg_defaults = cs42l73_reg_defaults,
+	.num_reg_defaults = ARRAY_SIZE(cs42l73_reg_defaults),
+	.volatile_reg = cs42l73_volatile_register,
+	.readable_reg = cs42l73_readable_register,
+	.cache_type = REGCACHE_RBTREE,
+};
+
+static __devinit int cs42l73_i2c_probe(struct i2c_client *i2c_client,
+				       const struct i2c_device_id *id)
+{
+	struct cs42l73_private *cs42l73;
+	int ret;
+	unsigned int devid = 0;
+	unsigned int reg;
+
+	cs42l73 = kzalloc((sizeof *cs42l73), GFP_KERNEL);
+	if (!cs42l73) {
+		dev_err(&i2c_client->dev, "could not allocate codec\n");
+		return -ENOMEM;
+	}
+
+	i2c_set_clientdata(i2c_client, cs42l73);
+
+	cs42l73->regmap = regmap_init_i2c(i2c_client, &cs42l73_regmap);
+	if (IS_ERR(cs42l73->regmap)) {
+		ret = PTR_ERR(cs42l73->regmap);
+		dev_err(&i2c_client->dev, "regmap_init() failed: %d\n", ret);
+		goto err;
+	}
+	/* initialize codec */
+	ret = regmap_read(cs42l73->regmap, CS42L73_DEVID_AB, &reg);
+	devid = (reg & 0xFF) << 12;
+
+	ret = regmap_read(cs42l73->regmap, CS42L73_DEVID_CD, &reg);
+	devid |= (reg & 0xFF) << 4;
+
+	ret = regmap_read(cs42l73->regmap, CS42L73_DEVID_E, &reg);
+	devid |= (reg & 0xF0) >> 4;
+
+
+	if (devid != CS42L73_DEVID) {
+		dev_err(&i2c_client->dev,
+			"CS42L73 Device ID (%X). Expected %X\n",
+			devid, CS42L73_DEVID);
+		goto err_regmap;
+	}
+
+	ret = regmap_read(cs42l73->regmap, CS42L73_REVID, &reg);
+	if (ret < 0) {
+		dev_err(&i2c_client->dev, "Get Revision ID failed\n");
+		goto err_regmap;
+	}
+
+	dev_info(&i2c_client->dev,
+		 "Cirrus Logic CS42L73, Revision: %02X\n", ret & 0xFF);
+
+	regcache_cache_only(cs42l73->regmap, true);
+
+	ret =  snd_soc_register_codec(&i2c_client->dev,
+			&soc_codec_dev_cs42l73, cs42l73_dai,
+			ARRAY_SIZE(cs42l73_dai));
+	if (ret < 0)
+		goto err_regmap;
+	return 0;
+
+err_regmap:
+	regmap_exit(cs42l73->regmap);
+
+err:
+	kfree(cs42l73);
+
+	return ret;
+}
+
+static __devexit int cs42l73_i2c_remove(struct i2c_client *client)
+{
+	struct cs42l73_private *cs42l73 = i2c_get_clientdata(client);
+
+	snd_soc_unregister_codec(&client->dev);
+	regmap_exit(cs42l73->regmap);
+
+	kfree(cs42l73);
+	return 0;
+}
+
+static const struct i2c_device_id cs42l73_id[] = {
+	{"cs42l73", 0},
+	{}
+};
+
+MODULE_DEVICE_TABLE(i2c, cs42l73_id);
+
+static struct i2c_driver cs42l73_i2c_driver = {
+	.driver = {
+		   .name = "cs42l73",
+		   .owner = THIS_MODULE,
+		   },
+	.id_table = cs42l73_id,
+	.probe = cs42l73_i2c_probe,
+	.remove = __devexit_p(cs42l73_i2c_remove),
+
+};
+
+static int __init cs42l73_modinit(void)
+{
+	int ret;
+	ret = i2c_add_driver(&cs42l73_i2c_driver);
+	if (ret != 0) {
+		pr_err("Failed to register CS42L73 I2C driver: %d\n", ret);
+		return ret;
+	}
+	return 0;
+}
+
+module_init(cs42l73_modinit);
+
+static void __exit cs42l73_exit(void)
+{
+	i2c_del_driver(&cs42l73_i2c_driver);
+}
+
+module_exit(cs42l73_exit);
+
+MODULE_DESCRIPTION("ASoC CS42L73 driver");
+MODULE_AUTHOR("Georgi Vlaev, Nucleus Systems Ltd, <joe@nucleusys.com>");
+MODULE_AUTHOR("Brian Austin, Cirrus Logic Inc, <brian.austin@cirrus.com>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/cs42l73.h b/sound/soc/codecs/cs42l73.h
new file mode 100644
index 000000000000..7c3bf7fd2f99
--- /dev/null
+++ b/sound/soc/codecs/cs42l73.h
@@ -0,0 +1,227 @@
+/*
+ * ALSA SoC CS42L73 codec driver
+ *
+ * Copyright 2011 Cirrus Logic, Inc.
+ *
+ * Author: Georgi Vlaev <joe@nucleusys.com>
+ *	   Brian Austin <brian.austin@cirrus.com>
+ *
+ * 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.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
+ * 02110-1301 USA
+ *
+ */
+
+#ifndef __CS42L73_H__
+#define __CS42L73_H__
+
+/* I2C Registers */
+/* I2C Address: 1001010[R/W] - 10010100 = 0x94(Write); 10010101 = 0x95(Read) */
+#define CS42L73_CHIP_ID		0x4a
+#define CS42L73_DEVID_AB	0x01	/* Device ID A & B [RO]. */
+#define CS42L73_DEVID_CD	0x02    /* Device ID C & D [RO]. */
+#define CS42L73_DEVID_E		0x03    /* Device ID E [RO]. */
+#define CS42L73_REVID		0x05    /* Revision ID [RO]. */
+#define CS42L73_PWRCTL1		0x06    /* Power Control 1. */
+#define CS42L73_PWRCTL2		0x07    /* Power Control 2. */
+#define CS42L73_PWRCTL3		0x08    /* Power Control 3. */
+#define CS42L73_CPFCHC		0x09    /* Charge Pump Freq. Class H Ctl. */
+#define CS42L73_OLMBMSDC	0x0A    /* Output Load, MIC Bias, MIC2 SDT */
+#define CS42L73_DMMCC		0x0B    /* Digital MIC & Master Clock Ctl. */
+#define CS42L73_XSPC		0x0C    /* Auxiliary Serial Port (XSP) Ctl. */
+#define CS42L73_XSPMMCC		0x0D    /* XSP Master Mode Clocking Control. */
+#define CS42L73_ASPC		0x0E    /* Audio Serial Port (ASP) Control. */
+#define CS42L73_ASPMMCC		0x0F    /* ASP Master Mode Clocking Control. */
+#define CS42L73_VSPC		0x10    /* Voice Serial Port (VSP) Control. */
+#define CS42L73_VSPMMCC		0x11    /* VSP Master Mode Clocking Control. */
+#define CS42L73_VXSPFS		0x12    /* VSP & XSP Sample Rate. */
+#define CS42L73_MIOPC		0x13    /* Misc. Input & Output Path Control. */
+#define CS42L73_ADCIPC		0x14	/* ADC/IP Control. */
+#define CS42L73_MICAPREPGAAVOL	0x15	/* MIC 1 [A] PreAmp, PGAA Vol. */
+#define CS42L73_MICBPREPGABVOL	0x16	/* MIC 2 [B] PreAmp, PGAB Vol. */
+#define CS42L73_IPADVOL		0x17	/* Input Pat7h A Digital Volume. */
+#define CS42L73_IPBDVOL		0x18	/* Input Path B Digital Volume. */
+#define CS42L73_PBDC		0x19	/* Playback Digital Control. */
+#define CS42L73_HLADVOL		0x1A	/* HP/Line A Out Digital Vol. */
+#define CS42L73_HLBDVOL		0x1B	/* HP/Line B Out Digital Vol. */
+#define CS42L73_SPKDVOL		0x1C	/* Spkphone Out [A] Digital Vol. */
+#define CS42L73_ESLDVOL		0x1D	/* Ear/Spkphone LO [B] Digital */
+#define CS42L73_HPAAVOL		0x1E	/* HP A Analog Volume. */
+#define CS42L73_HPBAVOL		0x1F	/* HP B Analog Volume. */
+#define CS42L73_LOAAVOL		0x20	/* Line Out A Analog Volume. */
+#define CS42L73_LOBAVOL		0x21	/* Line Out B Analog Volume. */
+#define CS42L73_STRINV		0x22	/* Stereo Input Path Adv. Vol. */
+#define CS42L73_XSPINV		0x23	/* Auxiliary Port Input Advisory Vol. */
+#define CS42L73_ASPINV		0x24	/* Audio Port Input Advisory Vol. */
+#define CS42L73_VSPINV		0x25	/* Voice Port Input Advisory Vol. */
+#define CS42L73_LIMARATEHL	0x26	/* Lmtr Attack Rate HP/Line. */
+#define CS42L73_LIMRRATEHL	0x27	/* Lmtr Ctl, Rel.Rate HP/Line. */
+#define CS42L73_LMAXHL		0x28	/* Lmtr Thresholds HP/Line. */
+#define CS42L73_LIMARATESPK	0x29	/* Lmtr Attack Rate Spkphone [A]. */
+#define CS42L73_LIMRRATESPK	0x2A	/* Lmtr Ctl,Release Rate Spk. [A]. */
+#define CS42L73_LMAXSPK		0x2B	/* Lmtr Thresholds Spkphone [A]. */
+#define CS42L73_LIMARATEESL	0x2C	/* Lmtr Attack Rate  */
+#define CS42L73_LIMRRATEESL	0x2D	/* Lmtr Ctl,Release Rate */
+#define CS42L73_LMAXESL		0x2E	/* Lmtr Thresholds */
+#define CS42L73_ALCARATE	0x2F	/* ALC Enable, Attack Rate AB. */
+#define CS42L73_ALCRRATE	0x30	/* ALC Release Rate AB.  */
+#define CS42L73_ALCMINMAX	0x31	/* ALC Thresholds AB. */
+#define CS42L73_NGCAB		0x32	/* Noise Gate Ctl AB. */
+#define CS42L73_ALCNGMC		0x33	/* ALC & Noise Gate Misc Ctl. */
+#define CS42L73_MIXERCTL	0x34	/* Mixer Control. */
+#define CS42L73_HLAIPAA		0x35	/* HP/LO Left Mixer: L. */
+#define CS42L73_HLBIPBA		0x36	/* HP/LO Right Mixer: R.  */
+#define CS42L73_HLAXSPAA	0x37	/* HP/LO Left Mixer: XSP L */
+#define CS42L73_HLBXSPBA	0x38	/* HP/LO Right Mixer: XSP R */
+#define CS42L73_HLAASPAA	0x39	/* HP/LO Left Mixer: ASP L */
+#define CS42L73_HLBASPBA	0x3A	/* HP/LO Right Mixer: ASP R */
+#define CS42L73_HLAVSPMA	0x3B	/* HP/LO Left Mixer: VSP. */
+#define CS42L73_HLBVSPMA	0x3C	/* HP/LO Right Mixer: VSP */
+#define CS42L73_XSPAIPAA	0x3D	/* XSP Left Mixer: Left */
+#define CS42L73_XSPBIPBA	0x3E	/* XSP Rt. Mixer: Right */
+#define CS42L73_XSPAXSPAA	0x3F	/* XSP Left Mixer: XSP L */
+#define CS42L73_XSPBXSPBA	0x40	/* XSP Rt. Mixer: XSP R */
+#define CS42L73_XSPAASPAA	0x41	/* XSP Left Mixer: ASP L */
+#define CS42L73_XSPAASPBA	0x42	/* XSP Rt. Mixer: ASP R */
+#define CS42L73_XSPAVSPMA	0x43	/* XSP Left Mixer: VSP */
+#define CS42L73_XSPBVSPMA	0x44	/* XSP Rt. Mixer: VSP */
+#define CS42L73_ASPAIPAA	0x45	/* ASP Left Mixer: Left */
+#define CS42L73_ASPBIPBA	0x46	/* ASP Rt. Mixer: Right */
+#define CS42L73_ASPAXSPAA	0x47	/* ASP Left Mixer: XSP L */
+#define CS42L73_ASPBXSPBA	0x48	/* ASP Rt. Mixer: XSP R */
+#define CS42L73_ASPAASPAA	0x49	/* ASP Left Mixer: ASP L */
+#define CS42L73_ASPBASPBA	0x4A	/* ASP Rt. Mixer: ASP R */
+#define CS42L73_ASPAVSPMA	0x4B	/* ASP Left Mixer: VSP */
+#define CS42L73_ASPBVSPMA	0x4C	/* ASP Rt. Mixer: VSP */
+#define CS42L73_VSPAIPAA	0x4D	/* VSP Left Mixer: Left */
+#define CS42L73_VSPBIPBA	0x4E	/* VSP Rt. Mixer: Right */
+#define CS42L73_VSPAXSPAA	0x4F	/* VSP Left Mixer: XSP L */
+#define CS42L73_VSPBXSPBA	0x50	/* VSP Rt. Mixer: XSP R */
+#define CS42L73_VSPAASPAA	0x51	/* VSP Left Mixer: ASP Left */
+#define CS42L73_VSPBASPBA	0x52	/* VSP Rt. Mixer: ASP Right */
+#define CS42L73_VSPAVSPMA	0x53	/* VSP Left Mixer: VSP */
+#define CS42L73_VSPBVSPMA	0x54	/* VSP Rt. Mixer: VSP */
+#define CS42L73_MMIXCTL		0x55	/* Mono Mixer Controls. */
+#define CS42L73_SPKMIPMA	0x56	/* SPK Mono Mixer: In. Path */
+#define CS42L73_SPKMXSPA	0x57	/* SPK Mono Mixer: XSP Mono/L/R Att. */
+#define CS42L73_SPKMASPA	0x58	/* SPK Mono Mixer: ASP Mono/L/R Att. */
+#define CS42L73_SPKMVSPMA	0x59	/* SPK Mono Mixer: VSP Mono Atten. */
+#define CS42L73_ESLMIPMA	0x5A	/* Ear/SpLO Mono Mixer: */
+#define CS42L73_ESLMXSPA	0x5B	/* Ear/SpLO Mono Mixer: XSP */
+#define CS42L73_ESLMASPA	0x5C	/* Ear/SpLO Mono Mixer: ASP */
+#define CS42L73_ESLMVSPMA	0x5D	/* Ear/SpLO Mono Mixer: VSP */
+#define CS42L73_IM1		0x5E	/* Interrupt Mask 1.  */
+#define CS42L73_IM2		0x5F	/* Interrupt Mask 2. */
+#define CS42L73_IS1		0x60	/* Interrupt Status 1 [RO]. */
+#define CS42L73_IS2		0x61	/* Interrupt Status 2 [RO]. */
+#define CS42L73_MAX_REGISTER	0x61	/* Total Registers */
+/* Bitfield Definitions */
+
+/* CS42L73_PWRCTL1 */
+#define PDN_ADCB		(1 << 7)
+#define PDN_DMICB		(1 << 6)
+#define PDN_ADCA		(1 << 5)
+#define PDN_DMICA		(1 << 4)
+#define PDN_LDO			(1 << 2)
+#define DISCHG_FILT		(1 << 1)
+#define PDN			(1 << 0)
+
+/* CS42L73_PWRCTL2 */
+#define PDN_MIC2_BIAS		(1 << 7)
+#define PDN_MIC1_BIAS		(1 << 6)
+#define PDN_VSP			(1 << 4)
+#define PDN_ASP_SDOUT		(1 << 3)
+#define PDN_ASP_SDIN		(1 << 2)
+#define PDN_XSP_SDOUT		(1 << 1)
+#define PDN_XSP_SDIN		(1 << 0)
+
+/* CS42L73_PWRCTL3 */
+#define PDN_THMS		(1 << 5)
+#define PDN_SPKLO		(1 << 4)
+#define PDN_EAR			(1 << 3)
+#define PDN_SPK			(1 << 2)
+#define PDN_LO			(1 << 1)
+#define PDN_HP			(1 << 0)
+
+/* Thermal Overload Detect. Requires interrupt ... */
+#define THMOVLD_150C		0
+#define THMOVLD_132C		1
+#define THMOVLD_115C		2
+#define THMOVLD_098C		3
+
+
+/* CS42L73_ASPC, CS42L73_XSPC, CS42L73_VSPC */
+#define	SP_3ST			(1 << 7)
+#define SPDIF_I2S		0
+#define SPDIF_PCM		(1 << 6)
+#define PCM_MODE0		0
+#define PCM_MODE1		1
+#define PCM_MODE2		2
+#define PCM_BO_MSBLSB		0
+#define PCM_BO_LSBMSB		1
+#define MCK_SCLK_64FS		0
+#define MCK_SCLK_MCLK		2
+#define MCK_SCLK_PREMCLK	3
+
+/* CS42L73_xSPMMCC */
+#define MS_MASTER		(1 << 7)
+
+
+/* CS42L73_DMMCC */
+#define MCLKDIS			(1 << 0)
+#define MCLKSEL_MCLK2		(1 << 4)
+#define MCLKSEL_MCLK1		(0 << 4)
+
+/* CS42L73 MCLK derived from MCLK1 or MCLK2 */
+#define CS42L73_CLKID_MCLK1     0
+#define CS42L73_CLKID_MCLK2     1
+
+#define CS42L73_MCLKXDIV	0
+#define CS42L73_MMCCDIV         1
+
+#define CS42L73_XSP		0
+#define CS42L73_ASP		1
+#define CS42L73_VSP		2
+
+/* IS1, IM1 */
+#define MIC2_SDET		(1 << 6)
+#define THMOVLD			(1 << 4)
+#define DIGMIXOVFL		(1 << 3)
+#define IPBOVFL			(1 << 1)
+#define IPAOVFL			(1 << 0)
+
+/* Analog Softramp */
+#define ANLGOSFT		(1 << 0)
+
+/* HP A/B Analog Mute */
+#define HPA_MUTE		(1 << 7)
+/* LO A/B Analog Mute	*/
+#define LOA_MUTE		(1 << 7)
+/* Digital Mute */
+#define HLAD_MUTE		(1 << 0)
+#define HLBD_MUTE		(1 << 1)
+#define SPKD_MUTE		(1 << 2)
+#define ESLD_MUTE		(1 << 3)
+
+/* Misc defines for codec */
+#define CS42L73_RESET_GPIO 143
+
+#define CS42L73_DEVID		0x00042A73
+#define CS42L73_MCLKX_MIN	5644800
+#define CS42L73_MCLKX_MAX	38400000
+
+#define CS42L73_SPC(id)		(CS42L73_XSPC + (id << 1))
+#define CS42L73_MMCC(id)	(CS42L73_XSPMMCC + (id << 1))
+#define CS42L73_SPFS(id)	((id == CS42L73_ASP) ? CS42L73_ASPC : CS42L73_VXSPFS)
+
+#endif	/* __CS42L73_H__ */