[PATCH v5] ASoC:Add support for cs42l73 codec

[PATCH v5] ASoC:Add support for cs42l73 codec

[Brian Austin]

Signed-off-by:Brian Austin <brian.austin@cirrus.com>
Signed-off-by:georgi Vlaev <joe@nucleusys.com>

This patch adds support for Cirrus Logic CS42L73
low power stereo codec
---
 sound/soc/codecs/Kconfig   |    4 +
 sound/soc/codecs/Makefile  |    2 +
 sound/soc/codecs/cs42l73.c | 1070 ++++++++++++++++++++++++++++++++++++++++++++
 sound/soc/codecs/cs42l73.h |  227 ++++++++++
 4 files changed, 1303 insertions(+), 0 deletions(-)
 create mode 100644 sound/soc/codecs/cs42l73.c
 create mode 100644 sound/soc/codecs/cs42l73.h

diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index 4584514..4f0ce61 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -28,6 +28,7 @@ config SND_SOC_ALL_CODECS
 	select SND_SOC_ALC5623 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
@@ -175,6 +176,9 @@ 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 a2c7842..b271eb5 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
@@ -115,6 +116,7 @@ obj-$(CONFIG_SND_SOC_AK4671)	+= snd-soc-ak4671.o
 obj-$(CONFIG_SND_SOC_ALC5623)    += snd-soc-alc5623.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 0000000..6dcc947
--- /dev/null
+++ b/sound/soc/codecs/cs42l73.c
@@ -0,0 +1,1070 @@
+/*
+ * 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/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 {
+	u32 sysclk;
+	u8 mclksel;
+	u32 mclk;
+	struct sp_config config[3];
+};
+
+static const u8 cs42l73_reg[] = {
+/*0*/ 0x00, 0x42, 0xA7, 0x30,
+/*4*/ 0x00, 0x00, 0xF1, 0xDF,
+/*8*/ 0x3F, 0x57, 0x53, 0x00,
+/*C*/ 0x00, 0x15, 0x00, 0x15,
+/*A*/ 0x00, 0x15, 0x00, 0x06,
+/*E*/ 0x00, 0x00, 0x00, 0x00,
+/*18*/ 0x00, 0x00, 0x00, 0x00,
+/*1C*/ 0x00, 0x00, 0x00, 0x00,
+/*20*/ 0x00, 0x00, 0x00, 0x00,
+/*24*/ 0x00, 0x00, 0x00, 0x7F,
+/*28*/ 0x00, 0x00, 0x3F, 0x00,
+/*2C*/ 0x00, 0x3F, 0x00, 0x00,
+/*30*/ 0x3F, 0x00, 0x00, 0x00,
+/*34*/ 0x18, 0x3F, 0x3F, 0x3F,
+/*38*/ 0x3F, 0x3F, 0x3F, 0x3F,
+/*3C*/ 0x3F, 0x3F, 0x3F, 0x3F,
+/*40*/ 0x3F, 0x3F, 0x3F, 0x3F,
+/*44*/ 0x3F, 0x3F, 0x3F, 0x3F,
+/*48*/ 0x3F, 0x3F, 0x3F, 0x3F,
+/*4C*/ 0x3F, 0x3F, 0x3F, 0x3F,
+/*50*/ 0x3F, 0x3F, 0x3F, 0x3F,
+/*54*/ 0x3F, 0xAA, 0x3F, 0x3F,
+/*58*/ 0x3F, 0x3F, 0x3F, 0x3F,
+/*5C*/ 0x3F, 0x3F, 0x00, 0x00,
+/*60*/ 0x00, 0x00
+};
+
+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 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_mixer_text[] = {
+	"Left", "Right", "Mono Mix"};
+
+static const struct soc_enum spk_asp_mono_mixer_enum =
+	SOC_ENUM_SINGLE(CS42L73_MMIXCTL, 6,
+		ARRAY_SIZE(cs42l73_mono_mixer_text), cs42l73_mono_mixer_text);
+
+static const struct soc_enum spk_xsp_mono_mixer_enum =
+	SOC_ENUM_SINGLE(CS42L73_MMIXCTL, 4,
+		ARRAY_SIZE(cs42l73_mono_mixer_text), cs42l73_mono_mixer_text);
+
+static const struct soc_enum esl_asp_mono_mixer_enum =
+	SOC_ENUM_SINGLE(CS42L73_MMIXCTL, 2,
+		ARRAY_SIZE(cs42l73_mono_mixer_text), cs42l73_mono_mixer_text);
+
+static const struct soc_enum esl_xsp_mono_mixer_enum =
+	SOC_ENUM_SINGLE(CS42L73_MMIXCTL, 0,
+		ARRAY_SIZE(cs42l73_mono_mixer_text), cs42l73_mono_mixer_text);
+
+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 vspout_mixer_enum =
+	SOC_ENUM_SINGLE(CS42L73_MIXERCTL, 5,
+		ARRAY_SIZE(cs42l73_spo_mixer_text), cs42l73_spo_mixer_text);
+
+static const struct soc_enum xspout_mixer_enum =
+	SOC_ENUM_SINGLE(CS42L73_MIXERCTL, 4,
+		ARRAY_SIZE(cs42l73_spo_mixer_text), cs42l73_spo_mixer_text);
+
+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, 1,
+			limiter_tlv),
+	SOC_SINGLE_TLV("ALC Min Threshold Volume", CS42L73_ALCMINMAX, 2, 7, 1,
+			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_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, 0x3F, 1, attn_tlv),
+	SOC_SINGLE_TLV("SPK-XSP Mono Volume",
+			CS42L73_SPKMXSPA, 0, 0x3F, 1, attn_tlv),
+	SOC_SINGLE_TLV("SPK-ASP Mono Volume",
+			CS42L73_SPKMASPA, 0, 0x3F, 1, attn_tlv),
+	SOC_SINGLE_TLV("SPK-VSP Mono Volume",
+			CS42L73_SPKMVSPMA, 0, 0x3F, 1, attn_tlv),
+
+	SOC_SINGLE_TLV("ESL-IP Mono Volume",
+			CS42L73_ESLMIPMA, 0, 0x3F, 1, attn_tlv),
+	SOC_SINGLE_TLV("ESL-XSP Mono Volume",
+			CS42L73_ESLMXSPA, 0, 0x3F, 1, attn_tlv),
+	SOC_SINGLE_TLV("ESL-ASP Mono Volume",
+			CS42L73_ESLMASPA, 0, 0x3F, 1, attn_tlv),
+	SOC_SINGLE_TLV("ESL-VSP Mono Volume",
+			CS42L73_ESLMVSPMA, 0, 0x3F, 1, attn_tlv),
+
+	SOC_ENUM("IP Digital Swap/Mono Select", ip_swap_enum),
+
+	SOC_ENUM("ESL-XSP Mono Mixer Select", esl_xsp_mono_mixer_enum),
+	SOC_ENUM("ESL-ASP Mono Mixer Select", esl_asp_mono_mixer_enum),
+
+	SOC_ENUM("SPK-ASP Mono Mixer Select", spk_asp_mono_mixer_enum),
+	SOC_ENUM("SPK-XSP Mono Mixer Select", spk_xsp_mono_mixer_enum),
+
+	SOC_ENUM("VSP Output Mixer Select", vspout_mixer_enum),
+	SOC_ENUM("XSP Output Mixer Select", xspout_mixer_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_INPUT("DMICA"),
+	SND_SOC_DAPM_INPUT("DMICB"),
+
+	SND_SOC_DAPM_AIF_OUT("XSPOUT", "XSP Capture",  0,
+			CS42L73_PWRCTL2, 1, 1),
+	SND_SOC_DAPM_AIF_OUT("ASPOUT", "ASP Capture",  0,
+			CS42L73_PWRCTL2, 3, 1),
+	SND_SOC_DAPM_AIF_OUT("VSPOUT", "VSP Capture",  0,
+			CS42L73_PWRCTL2, 4, 1),
+
+	SND_SOC_DAPM_AIF_IN("XSPIN", "XSP Playback",  0,
+			CS42L73_PWRCTL2, 0, 1),
+	SND_SOC_DAPM_AIF_IN("ASPIN", "ASP Playback",  0,
+			CS42L73_PWRCTL2, 2, 1),
+	SND_SOC_DAPM_AIF_IN("VSPIN", "VSP Playback",  0,
+			CS42L73_PWRCTL2, 4, 1),
+
+	SND_SOC_DAPM_ADC("ADC Left", NULL, CS42L73_PWRCTL1, 5, 1),
+	SND_SOC_DAPM_ADC("ADC Right", NULL, CS42L73_PWRCTL1, 7, 1),
+
+	SND_SOC_DAPM_PGA("DAC1", SND_SOC_NOPM, 0, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("DAC2", SND_SOC_NOPM, 0, 0, NULL, 0),
+
+	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 Mux Left", SND_SOC_NOPM, 0, 0, &pgaa_mux),
+	SND_SOC_DAPM_MUX("PGA Mux Right", SND_SOC_NOPM, 0, 0, &pgab_mux),
+
+	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("HPOUT"),
+	SND_SOC_DAPM_OUTPUT("LINEOUT"),
+	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[] = {
+
+	{"HPOUT", NULL, "HP Amp"},
+	{"LINEOUT", NULL, "LO Amp"},
+	{"SPKOUT", NULL, "SPK Amp"},
+	{"EAROUT", NULL, "EAR Amp"},
+	{"SPKLINEOUT", NULL, "SPKLO Amp"},
+
+	{"HP Amp", "Switch", "DAC1"},
+	{"LO Amp", "Switch", "DAC1"},
+	{"SPK Amp", "Switch", "DAC2"},
+	{"SPKLO Amp", "Switch", "DAC2"},
+	{"EAR Amp", "Switch", "DAC2"},
+
+	{"DAC1", NULL, "ASPIN"},
+	{"DAC1", NULL, "XSPIN"},
+	{"DAC1", NULL, "VSPIN"},
+
+	{"DAC2", NULL, "ASPIN"},
+	{"DAC2", NULL, "XSPIN"},
+	{"DAC2", NULL, "VSPIN"},
+
+	{"PGA Mux Left", "Line A", "LINEINA"},
+	{"PGA Mux Right", "Line B", "LINEINB"},
+
+	{"PGA Mux Left", "Mic 1", "MIC1"},
+	{"PGA Mux Right", "Mic 2", "MIC2"},
+
+	{"PGA Left", NULL, "PGA Mux Left"},
+	{"PGA Right", NULL, "PGA Mux Right"},
+
+	{"ADC Left", "ADC", "PGA Left"},
+	{"ADC Right", "ADC", "PGA Right"},
+
+	{"XSPOUT", NULL, "ADC Left"},
+	{"XSPOUT", NULL, "ADC Right"},
+
+	{"ASPOUT", NULL, "ADC Left"},
+	{"ASPOUT", NULL, "ADC Right"},
+
+	{"VSPOUT", NULL, "ADC Left"},
+	{"VSPOUT", NULL, "ADC Right"},
+
+};
+
+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 */
+		dev_dbg(codec->dev, "DAI[%d]: Slave\n", id);
+		priv->config[id].spc &= 0xFC;
+		priv->config[id].spc |= MCK_SCLK_64FS;
+	}
+	/* Update ASRCs */
+	priv->config[id].srate = srate;
+	cs42l73_update_asrc(codec, id, srate);
+	snd_soc_write(codec, CS42L73_SPC(id), priv->config[id].spc);
+	snd_soc_write(codec, CS42L73_MMCC(id), priv->config[id].mmcc);
+	return 0;
+}
+
+static int cs42l73_set_bias_level(struct snd_soc_codec *codec,
+				  enum snd_soc_bias_level level)
+{
+	int ret;
+
+	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) {
+			ret = snd_soc_cache_sync(codec);
+			if (ret < 0) {
+				dev_err(codec->dev,
+					"Failed to sync cache: %d\n", ret);
+				return ret;
+			}
+		}
+		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)
+{
+
+	cs42l73_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+	return 0;
+}
+
+static int cs42l73_probe(struct snd_soc_codec *codec)
+{
+	int ret;
+	unsigned int devid = 0;
+	struct cs42l73_private *cs42l73 = snd_soc_codec_get_drvdata(codec);
+
+	ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_I2C);
+	if (ret < 0) {
+		dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
+		return ret;
+	}
+
+	cs42l73_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+	/* initialize codec */
+	ret = snd_soc_read(codec, CS42L73_DEVID_AB);
+	devid = (ret & 0xFF) << 12;
+
+	ret = snd_soc_read(codec, CS42L73_DEVID_CD);
+	devid |= (ret & 0xFF) << 4;
+
+	ret = snd_soc_read(codec, CS42L73_DEVID_E);
+	devid |= (ret & 0xF0) >> 4;
+
+
+	if (devid != CS42L73_DEVID) {
+		dev_err(codec->dev,
+			"CS42L73 Device ID (%X). Expected %X\n",
+			devid, CS42L73_DEVID);
+		return ret;
+	}
+
+	ret = snd_soc_read(codec, CS42L73_REVID);
+	if (ret < 0) {
+		dev_err(codec->dev, "Get Revision ID failed\n");
+		return ret;
+	}
+
+	dev_info(codec->dev,
+		 "Cirrus Logic CS42L73, Revision: %02X\n", ret & 0xFF);
+
+	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,
+	.reg_cache_size = ARRAY_SIZE(cs42l73_reg),
+	.reg_cache_default = cs42l73_reg,
+	.reg_word_size = sizeof(u8),
+	.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 __devinit int cs42l73_i2c_probe(struct i2c_client *i2c_client,
+				       const struct i2c_device_id *id)
+{
+	struct cs42l73_private *cs42l73;
+	int ret;
+
+	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);
+
+	ret =  snd_soc_register_codec(&i2c_client->dev,
+			&soc_codec_dev_cs42l73, cs42l73_dai,
+			ARRAY_SIZE(cs42l73_dai));
+	if (ret < 0)
+		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);
+	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) {
+		printk(KERN_ERR "%s: can't add i2c driver\n", __func__);
+		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 0000000..188f06d
--- /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]. */
+
+/* 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__ */
-- 
1.7.6.3

Re: [PATCH v5] ASoC:Add support for cs42l73 codec

[Vinod Koul]

On Tue, 2011-10-18 at 10:46 -0500, Brian Austin wrote:
> +static const struct snd_soc_dapm_route cs42l73_audio_map[] = {
> +
> +       {"HPOUT", NULL, "HP Amp"},
> +       {"LINEOUT", NULL, "LO Amp"},
> +       {"SPKOUT", NULL, "SPK Amp"},
> +       {"EAROUT", NULL, "EAR Amp"},
> +       {"SPKLINEOUT", NULL, "SPKLO Amp"},
> +
> +       {"HP Amp", "Switch", "DAC1"},
> +       {"LO Amp", "Switch", "DAC1"},
> +       {"SPK Amp", "Switch", "DAC2"},
> +       {"SPKLO Amp", "Switch", "DAC2"},
> +       {"EAR Amp", "Switch", "DAC2"},
> + 
Since you have a mixer in codec before each of the amp, who is going to
configure those? And how are they represented here?

-- 
~Vinod

Re: [PATCH v5] ASoC:Add support for cs42l73 codec

[Austin, Brian]

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On Oct 19, 2011, at 3:42 AM, Vinod Koul wrote:

> On Tue, 2011-10-18 at 10:46 -0500, Brian Austin wrote:
>> +static const struct snd_soc_dapm_route cs42l73_audio_map[] = {
>> +
>> +       {"HPOUT", NULL, "HP Amp"},
>> +       {"LINEOUT", NULL, "LO Amp"},
>> +       {"SPKOUT", NULL, "SPK Amp"},
>> +       {"EAROUT", NULL, "EAR Amp"},
>> +       {"SPKLINEOUT", NULL, "SPKLO Amp"},
>> +
>> +       {"HP Amp", "Switch", "DAC1"},
>> +       {"LO Amp", "Switch", "DAC1"},
>> +       {"SPK Amp", "Switch", "DAC2"},
>> +       {"SPKLO Amp", "Switch", "DAC2"},
>> +       {"EAR Amp", "Switch", "DAC2"},
>> + 
> Since you have a mixer in codec before each of the amp, who is going to
> configure those? And how are they represented here?
> 
> -- 
> ~Vinod
> 
> 
The mixer in the codec is configured by the Attenuation mixer controls.
You then select the mono/stereo output with the xSP Output Mixer Select.
I didn't think we needed to represent that in the DAPM map.

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Re: [PATCH v5] ASoC:Add support for cs42l73 codec

[Vinod Koul]

On Thu, 2011-10-20 at 14:58 +0100, Mark Brown wrote:
> On Thu, Oct 20, 2011 at 01:54:06PM +0000, Austin, Brian wrote:
> 
> > The mixer in the codec is configured by the Attenuation mixer controls.
> > You then select the mono/stereo output with the xSP Output Mixer Select.
> > I didn't think we needed to represent that in the DAPM map.
> 
> That would be *extremely* surprising given that DAPM is supposed to
> understand the audio routing through the CODEC so it can tell where the
> connected audio paths go.  Doesn't your current design cause too much of
> the device to be powered up?
Precisely why this should be controlled by DAPM  :)

-- 
~Vinod

Re: [PATCH v5] ASoC:Add support for cs42l73 codec

[Mark Brown]

On Thu, Oct 20, 2011 at 01:54:06PM +0000, Austin, Brian wrote:

> The mixer in the codec is configured by the Attenuation mixer controls.
> You then select the mono/stereo output with the xSP Output Mixer Select.
> I didn't think we needed to represent that in the DAPM map.

That would be *extremely* surprising given that DAPM is supposed to
understand the audio routing through the CODEC so it can tell where the
connected audio paths go.  Doesn't your current design cause too much of
the device to be powered up?

Re: [PATCH v5] ASoC:Add support for cs42l73 codec

[Austin, Brian]

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On Oct 20, 2011, at 8:54 AM, Vinod Koul wrote:

> On Thu, 2011-10-20 at 14:58 +0100, Mark Brown wrote:
>> On Thu, Oct 20, 2011 at 01:54:06PM +0000, Austin, Brian wrote:
>> 
>>> The mixer in the codec is configured by the Attenuation mixer controls.
>>> You then select the mono/stereo output with the xSP Output Mixer Select.
>>> I didn't think we needed to represent that in the DAPM map.
>> 
>> That would be *extremely* surprising given that DAPM is supposed to
>> understand the audio routing through the CODEC so it can tell where the
>> connected audio paths go.  Doesn't your current design cause too much of
>> the device to be powered up?
> Precisely why this should be controlled by DAPM  :)
> 
> -- 
> ~Vinod
> 
> _______________________________________________
> Alsa-devel mailing list
> Alsa-devel@alsa-project.org
> http://mailman.alsa-project.org/mailman/listinfo/alsa-devel
> 
With the current mixer settings enabling the AMP to be used for output, the power is exactly where it is supposed to be.

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Re: [PATCH v5] ASoC:Add support for cs42l73 codec

[Mark Brown]

On Thu, Oct 20, 2011 at 02:11:32PM +0000, Austin, Brian wrote:

> With the current mixer settings enabling the AMP to be used for
> output, the power is exactly where it is supposed to be.

Could you explain how this works please?  It may be that we're just
missing something about how the chip power is managed here but what
you're describing is definitely not normal.

Re: [PATCH v5] ASoC:Add support for cs42l73 codec

[Austin, Brian]

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On Oct 20, 2011, at 9:20 AM, Mark Brown wrote:

> On Thu, Oct 20, 2011 at 02:11:32PM +0000, Austin, Brian wrote:
> 
>> With the current mixer settings enabling the AMP to be used for
>> output, the power is exactly where it is supposed to be.
> 
> Could you explain how this works please?  It may be that we're just
> missing something about how the chip power is managed here but what
> you're describing is definitely not normal.
> 
OK.  Yeah, the chip is different as far as how power is managed internally.
It can also be that I am missing something wrt how this DAPM mapping is supposed to work.

Let me write something up and I'll send it.

Thanks

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Re: [PATCH v5] ASoC:Add support for cs42l73 codec

[Mark Brown]

On Thu, Oct 20, 2011 at 09:31:50PM +0300, Georgi Vlaev wrote:
> Hello,

Don't top post.

> The codec has 2 basic groups of power control bits:

You're describing some features that don't sound at all unusual here...
I'm not sure what point you're trying to make here?

Re: [PATCH v5] ASoC:Add support for cs42l73 codec

[Austin, Brian]

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On Oct 20, 2011, at 3:00 PM, Mark Brown wrote:

> On Thu, Oct 20, 2011 at 09:31:50PM +0300, Georgi Vlaev wrote:
>> Hello,
> 
> Don't top post.
> 
>> The codec has 2 basic groups of power control bits:
> 
> You're describing some features that don't sound at all unusual here...
> I'm not sure what point you're trying to make here?
> 
We have only 2 power domains.  The input PCM and the output AMP.
The routing between these 2 points in the codec is handled by volume levels.

Say I am routing through the XSP port to the HP. If I power the HP Amp and adjust the HP-XSP volume level, the route is then established.
I could also adjust the SPK-XSP volume and power the SPK Amp to route the stream to the Speaker as well.

Given that, why would there be a need to show all routes through the mixer in a DAPM context? How do you show that?

would it look something like this?

{"HPOUT",  NULL, "HP Amp"}
{"HP Amp", "Amp Switch",  "DAC1"}
{"DAC1",  "HP-XSP Volume Control",  "XSPIN"}

You can represent the "HP-XSP Volume Control" control as DAPM? 

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Re: [PATCH v5] ASoC:Add support for cs42l73 codec

[Mark Brown]

On Thu, Oct 20, 2011 at 08:26:11PM +0000, Austin, Brian wrote:

Fix your mailer to word wrap within paragraphs, I've reflowed for
legibility.

> We have only 2 power domains.  The input PCM and the output AMP.
> The routing between these 2 points in the codec is handled by volume levels.

Like I say this is pretty unremarkable - the volumes are your DAPM
routing controls.

> Given that, why would there be a need to show all routes through the
> mixer in a DAPM context? How do you show that?  would it look
> something like this?

Well, the most obvious issue is that even if your device doesn't have
any useful internal power management there may be external devices
connected to the outputs (eg, a high power speaker amp) which do and so
we need to know which outputs are actually active in order to control
the outputs.

> {"HPOUT",  NULL, "HP Amp"}
> {"HP Amp", "Amp Switch",  "DAC1"}
> {"DAC1",  "HP-XSP Volume Control",  "XSPIN"}

This looks wrong, the control on the DAC looks like it actually controls
just the one input path.

> You can represent the "HP-XSP Volume Control" control as DAPM? 

Yes.

Re: [PATCH v5] ASoC:Add support for cs42l73 codec

[Austin, Brian]

On Oct 20, 2011, at 4:28 PM, "Mark Brown" <broonie@opensource.wolfsonmicro.com> wrote:

> On Thu, Oct 20, 2011 at 08:26:11PM +0000, Austin, Brian wrote:
> 
> Fix your mailer to word wrap within paragraphs, I've reflowed for
> legibility.
> 
>> We have only 2 power domains.  The input PCM and the output AMP.
>> The routing between these 2 points in the codec is handled by volume levels.
> 
> Like I say this is pretty unremarkable - the volumes are your DAPM
> routing controls.
> 
>> Given that, why would there be a need to show all routes through the
>> mixer in a DAPM context? How do you show that?  would it look
>> something like this?
> 
> Well, the most obvious issue is that even if your device doesn't have
> any useful internal power management there may be external devices
> connected to the outputs (eg, a high power speaker amp) which do and so
> we need to know which outputs are actually active in order to control
> the outputs.

That makes sense, thanks
> 
>> {"HPOUT",  NULL, "HP Amp"}
>> {"HP Amp", "Amp Switch",  "DAC1"}
>> {"DAC1",  "HP-XSP Volume Control",  "XSPIN"}
> 
> This looks wrong, the control on the DAC looks like it actually controls
> just the one input path.

That is just one example, it would be
for all inputs as the mixer controls are.
> 
>> You can represent the "HP-XSP Volume Control" control as DAPM? 
> 
> Yes.
> ______
Let me rework the routes to show all inputs.

Sorry this has been such a hassle.

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> 

Re: [PATCH v5] ASoC:Add support for cs42l73 codec

[Brian Austin]

>>
>>> {"HPOUT",  NULL, "HP Amp"}
>>> {"HP Amp", "Amp Switch",  "DAC1"}
>>> {"DAC1",  "HP-XSP Volume Control",  "XSPIN"}
>>
>> This looks wrong, the control on the DAC looks like it actually controls
>> just the one input path.
>
> That is just one example, it would be
> for all inputs as the mixer controls are.
>>
>>> You can represent the "HP-XSP Volume Control" control as DAPM?
>>
>> Yes.
>> ______
> Let me rework the routes to show all inputs.
>
> Sorry this has been such a hassle.
>
I get it now! This is going to add a lot of code as there are a lot of
mixer controls. Using volume controls as DAPM I can create all routes in 
the codec.

Thanks for the help.

Re: [PATCH v5] ASoC:Add support for cs42l73 codec

[Mark Brown]

On Fri, Oct 21, 2011 at 08:23:44AM -0500, Brian Austin wrote:

> I get it now! This is going to add a lot of code as there are a lot of
> mixer controls. Using volume controls as DAPM I can create all
> routes in the codec.

I did also mean to say that if you do have the really coarse power
control it might be worth looking at using supply widgets to control
the actual registers and no power management (SND_SOC_NOPM) for the
widgets used for routing.