diff --git a/Documentation/devicetree/bindings/sound/imx-audio-sgtl5000.txt b/Documentation/devicetree/bindings/sound/imx-audio-sgtl5000.txt
new file mode 100644
index 000000000000..e4acdd891e49
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/imx-audio-sgtl5000.txt
@@ -0,0 +1,49 @@
+Freescale i.MX audio complex with SGTL5000 codec
+
+Required properties:
+- compatible : "fsl,imx-audio-sgtl5000"
+- model : The user-visible name of this sound complex
+- ssi-controller : The phandle of the i.MX SSI controller
+- audio-codec : The phandle of the SGTL5000 audio codec
+- audio-routing : A list of the connections between audio components.
+  Each entry is a pair of strings, the first being the connection's sink,
+  the second being the connection's source. Valid names could be power
+  supplies, SGTL5000 pins, and the jacks on the board:
+
+  Power supplies:
+   * Mic Bias
+
+  SGTL5000 pins:
+   * MIC_IN
+   * LINE_IN
+   * HP_OUT
+   * LINE_OUT
+
+  Board connectors:
+   * Mic Jack
+   * Line In Jack
+   * Headphone Jack
+   * Line Out Jack
+   * Ext Spk
+
+- mux-int-port : The internal port of the i.MX audio muxer (AUDMUX)
+- mux-ext-port : The external port of the i.MX audio muxer
+
+Note: The AUDMUX port numbering should start at 1, which is consistent with
+hardware manual.
+
+Example:
+
+sound {
+	compatible = "fsl,imx51-babbage-sgtl5000",
+		     "fsl,imx-audio-sgtl5000";
+	model = "imx51-babbage-sgtl5000";
+	ssi-controller = <&ssi1>;
+	audio-codec = <&sgtl5000>;
+	audio-routing =
+		"MIC_IN", "Mic Jack",
+		"Mic Jack", "Mic Bias",
+		"Headphone Jack", "HP_OUT";
+	mux-int-port = <1>;
+	mux-ext-port = <3>;
+};
diff --git a/Documentation/devicetree/bindings/sound/mxs-audio-sgtl5000.txt b/Documentation/devicetree/bindings/sound/mxs-audio-sgtl5000.txt
new file mode 100644
index 000000000000..601c518eddaa
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/mxs-audio-sgtl5000.txt
@@ -0,0 +1,17 @@
+* Freescale MXS audio complex with SGTL5000 codec
+
+Required properties:
+- compatible: "fsl,mxs-audio-sgtl5000"
+- model: The user-visible name of this sound complex
+- saif-controllers: The phandle list of the MXS SAIF controller
+- audio-codec: The phandle of the SGTL5000 audio codec
+
+Example:
+
+sound {
+	compatible = "fsl,imx28-evk-sgtl5000",
+		     "fsl,mxs-audio-sgtl5000";
+	model = "imx28-evk-sgtl5000";
+	saif-controllers = <&saif0 &saif1>;
+	audio-codec = <&sgtl5000>;
+};
diff --git a/Documentation/devicetree/bindings/sound/mxs-saif.txt b/Documentation/devicetree/bindings/sound/mxs-saif.txt
new file mode 100644
index 000000000000..c37ba6143d9b
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/mxs-saif.txt
@@ -0,0 +1,36 @@
+* Freescale MXS Serial Audio Interface (SAIF)
+
+Required properties:
+- compatible: Should be "fsl,<chip>-saif"
+- reg: Should contain registers location and length
+- interrupts: Should contain ERROR and DMA interrupts
+- fsl,saif-dma-channel: APBX DMA channel for the SAIF
+
+Optional properties:
+- fsl,saif-master: phandle to the master SAIF.  It's only required for
+  the slave SAIF.
+
+Note: Each SAIF controller should have an alias correctly numbered
+in "aliases" node.
+
+Example:
+
+aliases {
+	saif0 = &saif0;
+	saif1 = &saif1;
+};
+
+saif0: saif@80042000 {
+	compatible = "fsl,imx28-saif";
+	reg = <0x80042000 2000>;
+	interrupts = <59 80>;
+	fsl,saif-dma-channel = <4>;
+};
+
+saif1: saif@80046000 {
+	compatible = "fsl,imx28-saif";
+	reg = <0x80046000 2000>;
+	interrupts = <58 81>;
+	fsl,saif-dma-channel = <5>;
+	fsl,saif-master = <&saif0>;
+};
diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra30-ahub.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra30-ahub.txt
new file mode 100644
index 000000000000..1ac7b1642186
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/nvidia,tegra30-ahub.txt
@@ -0,0 +1,32 @@
+NVIDIA Tegra30 AHUB (Audio Hub)
+
+Required properties:
+- compatible : "nvidia,tegra30-ahub"
+- reg : Should contain the register physical address and length for each of
+  the AHUB's APBIF registers and the AHUB's own registers.
+- interrupts : Should contain AHUB interrupt
+- nvidia,dma-request-selector : The Tegra DMA controller's phandle and
+  request selector for the first APBIF channel.
+- ranges : The bus address mapping for the configlink register bus.
+  Can be empty since the mapping is 1:1.
+- #address-cells : For the configlink bus. Should be <1>;
+- #size-cells : For the configlink bus. Should be <1>.
+
+AHUB client modules need to specify the IDs of their CIFs (Client InterFaces).
+For RX CIFs, the numbers indicate the register number within AHUB routing
+register space (APBIF 0..3 RX, I2S 0..5 RX, DAM 0..2 RX 0..1, SPDIF RX 0..1).
+For TX CIFs, the numbers indicate the bit position within the AHUB routing
+registers (APBIF 0..3 TX, I2S 0..5 TX, DAM 0..2 TX, SPDIF TX 0..1).
+
+Example:
+
+ahub@70080000 {
+	compatible = "nvidia,tegra30-ahub";
+	reg = <0x70080000 0x200 0x70080200 0x100>;
+	interrupts = < 0 103 0x04 >;
+	nvidia,dma-request-selector = <&apbdma 1>;
+
+	ranges;
+	#address-cells = <1>;
+	#size-cells = <1>;
+};
diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra30-i2s.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra30-i2s.txt
new file mode 100644
index 000000000000..dfa6c037124a
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/nvidia,tegra30-i2s.txt
@@ -0,0 +1,15 @@
+NVIDIA Tegra30 I2S controller
+
+Required properties:
+- compatible : "nvidia,tegra30-i2s"
+- reg : Should contain I2S registers location and length
+- nvidia,ahub-cif-ids : The list of AHUB CIF IDs for this port, rx (playback)
+  first, tx (capture) second. See nvidia,tegra30-ahub.txt for values.
+
+Example:
+
+i2s@70002800 {
+	compatible = "nvidia,tegra30-i2s";
+	reg = <0x70080300 0x100>;
+	nvidia,ahub-cif-ids = <4 4>;
+};
diff --git a/Documentation/devicetree/bindings/sound/tegra-audio-trimslice.txt b/Documentation/devicetree/bindings/sound/tegra-audio-trimslice.txt
new file mode 100644
index 000000000000..04b14cfb1f16
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/tegra-audio-trimslice.txt
@@ -0,0 +1,14 @@
+NVIDIA Tegra audio complex for TrimSlice
+
+Required properties:
+- compatible : "nvidia,tegra-audio-trimslice"
+- nvidia,i2s-controller : The phandle of the Tegra I2S1 controller
+- nvidia,audio-codec : The phandle of the WM8903 audio codec
+
+Example:
+
+sound {
+	compatible = "nvidia,tegra-audio-trimslice";
+	nvidia,i2s-controller = <&tegra_i2s1>;
+	nvidia,audio-codec = <&codec>;
+};
diff --git a/Documentation/devicetree/bindings/sound/tegra-audio-wm8753.txt b/Documentation/devicetree/bindings/sound/tegra-audio-wm8753.txt
new file mode 100644
index 000000000000..c4dd39ce6165
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/tegra-audio-wm8753.txt
@@ -0,0 +1,54 @@
+NVIDIA Tegra audio complex
+
+Required properties:
+- compatible : "nvidia,tegra-audio-wm8753"
+- nvidia,model : The user-visible name of this sound complex.
+- nvidia,audio-routing : A list of the connections between audio components.
+  Each entry is a pair of strings, the first being the connection's sink,
+  the second being the connection's source. Valid names for sources and
+  sinks are the WM8753's pins, and the jacks on the board:
+
+  WM8753 pins:
+
+  * LOUT1
+  * LOUT2
+  * ROUT1
+  * ROUT2
+  * MONO1
+  * MONO2
+  * OUT3
+  * OUT4
+  * LINE1
+  * LINE2
+  * RXP
+  * RXN
+  * ACIN
+  * ACOP
+  * MIC1N
+  * MIC1
+  * MIC2N
+  * MIC2
+  * Mic Bias
+
+  Board connectors:
+
+  * Headphone Jack
+  * Mic Jack
+
+- nvidia,i2s-controller : The phandle of the Tegra I2S1 controller
+- nvidia,audio-codec : The phandle of the WM8753 audio codec
+Example:
+
+sound {
+	compatible = "nvidia,tegra-audio-wm8753-whistler",
+		     "nvidia,tegra-audio-wm8753"
+	nvidia,model = "tegra-wm8753-harmony";
+
+	nvidia,audio-routing =
+		"Headphone Jack", "LOUT1",
+		"Headphone Jack", "ROUT1";
+
+	nvidia,i2s-controller = <&i2s1>;
+	nvidia,audio-codec = <&wm8753>;
+};
+
diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt
index 8c16d50f6cb6..221b81016dba 100644
--- a/Documentation/sound/alsa/ALSA-Configuration.txt
+++ b/Documentation/sound/alsa/ALSA-Configuration.txt
@@ -1545,7 +1545,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
 
     Module for sound cards based on the C-Media CMI8786/8787/8788 chip:
     * Asound A-8788
-    * Asus Xonar DG
+    * Asus Xonar DG/DGX
     * AuzenTech X-Meridian
     * AuzenTech X-Meridian 2G
     * Bgears b-Enspirer
diff --git a/MAINTAINERS b/MAINTAINERS
index 9c293bf340b1..61dd168d349a 100644
--- a/MAINTAINERS
+++ b/MAINTAINERS
@@ -6691,6 +6691,12 @@ F:	drivers/misc/tifm*
 F:	drivers/mmc/host/tifm_sd.c
 F:	include/linux/tifm.h
 
+TI LM49xxx FAMILY ASoC CODEC DRIVERS
+M:	M R Swami Reddy <mr.swami.reddy@ti.com>
+L:	alsa-devel@alsa-project.org (moderated for non-subscribers)
+S:	Maintained
+F:	sound/soc/codecs/lm49453*
+
 TI TWL4030 SERIES SOC CODEC DRIVER
 M:	Peter Ujfalusi <peter.ujfalusi@ti.com>
 L:	alsa-devel@alsa-project.org (moderated for non-subscribers)
diff --git a/arch/arm/mach-imx/Kconfig b/arch/arm/mach-imx/Kconfig
index 7d6322ce5223..cca8c0c74794 100644
--- a/arch/arm/mach-imx/Kconfig
+++ b/arch/arm/mach-imx/Kconfig
@@ -151,6 +151,7 @@ config MACH_MX25_3DS
 	select IMX_HAVE_PLATFORM_IMX2_WDT
 	select IMX_HAVE_PLATFORM_IMXDI_RTC
 	select IMX_HAVE_PLATFORM_IMX_I2C
+	select IMX_HAVE_PLATFORM_IMX_SSI
 	select IMX_HAVE_PLATFORM_IMX_FB
 	select IMX_HAVE_PLATFORM_IMX_KEYPAD
 	select IMX_HAVE_PLATFORM_IMX_UART
@@ -495,6 +496,7 @@ config MACH_MX31MOBOARD
 	select IMX_HAVE_PLATFORM_FSL_USB2_UDC
 	select IMX_HAVE_PLATFORM_IMX2_WDT
 	select IMX_HAVE_PLATFORM_IMX_I2C
+	select IMX_HAVE_PLATFORM_IMX_SSI
 	select IMX_HAVE_PLATFORM_IMX_UART
 	select IMX_HAVE_PLATFORM_IPU_CORE
 	select IMX_HAVE_PLATFORM_MXC_EHCI
diff --git a/arch/arm/mach-imx/mach-mx31_3ds.c b/arch/arm/mach-imx/mach-mx31_3ds.c
index 4d1aab154400..4eafdf275ea2 100644
--- a/arch/arm/mach-imx/mach-mx31_3ds.c
+++ b/arch/arm/mach-imx/mach-mx31_3ds.c
@@ -156,6 +156,11 @@ static int mx31_3ds_pins[] = {
 	MX31_PIN_CSI_VSYNC__CSI_VSYNC,
 	MX31_PIN_CSI_D5__GPIO3_5, /* CMOS PWDN */
 	IOMUX_MODE(MX31_PIN_RI_DTE1, IOMUX_CONFIG_GPIO), /* CMOS reset */
+	/* SSI */
+	MX31_PIN_STXD4__STXD4,
+	MX31_PIN_SRXD4__SRXD4,
+	MX31_PIN_SCK4__SCK4,
+	MX31_PIN_SFS4__SFS4,
 };
 
 /*
@@ -488,12 +493,23 @@ static struct mc13xxx_regulator_init_data mx31_3ds_regulators[] = {
 };
 
 /* MC13783 */
+static struct mc13xxx_codec_platform_data mx31_3ds_codec = {
+	.dac_ssi_port = MC13783_SSI1_PORT,
+	.adc_ssi_port = MC13783_SSI1_PORT,
+};
+
 static struct mc13xxx_platform_data mc13783_pdata = {
 	.regulators = {
 		.regulators = mx31_3ds_regulators,
 		.num_regulators = ARRAY_SIZE(mx31_3ds_regulators),
 	},
-	.flags  = MC13XXX_USE_TOUCHSCREEN | MC13XXX_USE_RTC,
+	.codec = &mx31_3ds_codec,
+	.flags  = MC13XXX_USE_TOUCHSCREEN | MC13XXX_USE_RTC | MC13XXX_USE_CODEC,
+
+};
+
+static struct imx_ssi_platform_data mx31_3ds_ssi_pdata = {
+	.flags = IMX_SSI_DMA | IMX_SSI_NET,
 };
 
 /* SPI */
@@ -741,6 +757,10 @@ static void __init mx31_3ds_init(void)
 	}
 
 	mx31_3ds_init_camera();
+
+	imx31_add_imx_ssi(0, &mx31_3ds_ssi_pdata);
+
+	imx_add_platform_device("imx_mc13783", 0, NULL, 0, NULL, 0);
 }
 
 static void __init mx31_3ds_timer_init(void)
diff --git a/arch/arm/mach-imx/mach-mx31moboard.c b/arch/arm/mach-imx/mach-mx31moboard.c
index 1dfe3c7a7be1..016791f038b0 100644
--- a/arch/arm/mach-imx/mach-mx31moboard.c
+++ b/arch/arm/mach-imx/mach-mx31moboard.c
@@ -47,6 +47,7 @@
 #include <mach/hardware.h>
 #include <mach/iomux-mx3.h>
 #include <mach/ulpi.h>
+#include <mach/ssi.h>
 
 #include "devices-imx31.h"
 
@@ -102,6 +103,9 @@ static unsigned int moboard_pins[] = {
 	MX31_PIN_CSPI3_MOSI__MOSI, MX31_PIN_CSPI3_MISO__MISO,
 	MX31_PIN_CSPI3_SCLK__SCLK, MX31_PIN_CSPI3_SPI_RDY__SPI_RDY,
 	MX31_PIN_CSPI2_SS1__CSPI3_SS1,
+	/* SSI */
+	MX31_PIN_STXD4__STXD4, MX31_PIN_SRXD4__SRXD4,
+	MX31_PIN_SCK4__SCK4, MX31_PIN_SFS4__SFS4,
 };
 
 static struct physmap_flash_data mx31moboard_flash_data = {
@@ -276,6 +280,11 @@ static struct mc13xxx_buttons_platform_data moboard_buttons = {
 	.b1on_key = KEY_POWER,
 };
 
+static struct mc13xxx_codec_platform_data moboard_codec = {
+	.dac_ssi_port = MC13783_SSI1_PORT,
+	.adc_ssi_port = MC13783_SSI1_PORT,
+};
+
 static struct mc13xxx_platform_data moboard_pmic = {
 	.regulators = {
 		.regulators = moboard_regulators,
@@ -283,7 +292,12 @@ static struct mc13xxx_platform_data moboard_pmic = {
 	},
 	.leds = &moboard_leds,
 	.buttons = &moboard_buttons,
-	.flags = MC13XXX_USE_RTC | MC13XXX_USE_ADC,
+	.codec = &moboard_codec,
+	.flags = MC13XXX_USE_RTC | MC13XXX_USE_ADC | MC13XXX_USE_CODEC,
+};
+
+static struct imx_ssi_platform_data moboard_ssi_pdata = {
+	.flags = IMX_SSI_DMA | IMX_SSI_NET,
 };
 
 static struct spi_board_info moboard_spi_board_info[] __initdata = {
@@ -554,6 +568,10 @@ static void __init mx31moboard_init(void)
 
 	moboard_usbh2_init();
 
+	imx31_add_imx_ssi(0, &moboard_ssi_pdata);
+
+	imx_add_platform_device("imx_mc13783", 0, NULL, 0, NULL, 0);
+
 	pm_power_off = mx31moboard_poweroff;
 
 	switch (mx31moboard_baseboard) {
diff --git a/arch/arm/mach-shmobile/Kconfig b/arch/arm/mach-shmobile/Kconfig
index 98327b7a503c..f31383c32f9c 100644
--- a/arch/arm/mach-shmobile/Kconfig
+++ b/arch/arm/mach-shmobile/Kconfig
@@ -64,6 +64,7 @@ config MACH_AP4EVB
 	depends on ARCH_SH7372
 	select ARCH_REQUIRE_GPIOLIB
 	select SH_LCD_MIPI_DSI
+	select SND_SOC_AK4642 if SND_SIMPLE_CARD
 
 choice
 	prompt "AP4EVB LCD panel selection"
@@ -88,6 +89,7 @@ config MACH_MACKEREL
 	bool "mackerel board"
 	depends on ARCH_SH7372
 	select ARCH_REQUIRE_GPIOLIB
+	select SND_SOC_AK4642 if SND_SIMPLE_CARD
 
 config MACH_KOTA2
 	bool "KOTA2 board"
diff --git a/arch/arm/mach-shmobile/board-ap4evb.c b/arch/arm/mach-shmobile/board-ap4evb.c
index 0c3caeba2f3e..b540b8eb20ca 100644
--- a/arch/arm/mach-shmobile/board-ap4evb.c
+++ b/arch/arm/mach-shmobile/board-ap4evb.c
@@ -50,6 +50,7 @@
 #include <media/soc_camera.h>
 
 #include <sound/sh_fsi.h>
+#include <sound/simple_card.h>
 
 #include <video/sh_mobile_hdmi.h>
 #include <video/sh_mobile_lcdc.h>
@@ -785,17 +786,25 @@ static struct platform_device fsi_device = {
 	},
 };
 
-static struct fsi_ak4642_info fsi2_ak4643_info = {
+static struct asoc_simple_dai_init_info fsi2_ak4643_init_info = {
+	.fmt		= SND_SOC_DAIFMT_LEFT_J,
+	.codec_daifmt	= SND_SOC_DAIFMT_CBM_CFM,
+	.cpu_daifmt	= SND_SOC_DAIFMT_CBS_CFS,
+	.sysclk		= 11289600,
+};
+
+static struct asoc_simple_card_info fsi2_ak4643_info = {
 	.name		= "AK4643",
 	.card		= "FSI2A-AK4643",
 	.cpu_dai	= "fsia-dai",
 	.codec		= "ak4642-codec.0-0013",
 	.platform	= "sh_fsi2",
-	.id		= FSI_PORT_A,
+	.codec_dai	= "ak4642-hifi",
+	.init		= &fsi2_ak4643_init_info,
 };
 
 static struct platform_device fsi_ak4643_device = {
-	.name	= "fsi-ak4642-audio",
+	.name	= "asoc-simple-card",
 	.dev	= {
 		.platform_data	= &fsi2_ak4643_info,
 	},
@@ -900,8 +909,26 @@ static struct platform_device lcdc1_device = {
 	},
 };
 
+static struct asoc_simple_dai_init_info fsi2_hdmi_init_info = {
+	.cpu_daifmt	= SND_SOC_DAIFMT_CBM_CFM,
+};
+
+static struct asoc_simple_card_info fsi2_hdmi_info = {
+	.name		= "HDMI",
+	.card		= "FSI2B-HDMI",
+	.cpu_dai	= "fsib-dai",
+	.codec		= "sh-mobile-hdmi",
+	.platform	= "sh_fsi2",
+	.codec_dai	= "sh_mobile_hdmi-hifi",
+	.init		= &fsi2_hdmi_init_info,
+};
+
 static struct platform_device fsi_hdmi_device = {
-	.name		= "sh_fsi2_b_hdmi",
+	.name	= "asoc-simple-card",
+	.id	= 1,
+	.dev	= {
+		.platform_data	= &fsi2_hdmi_info,
+	},
 };
 
 static struct gpio_led ap4evb_leds[] = {
diff --git a/arch/arm/mach-shmobile/board-mackerel.c b/arch/arm/mach-shmobile/board-mackerel.c
index aae2e24fde46..50c67b22d087 100644
--- a/arch/arm/mach-shmobile/board-mackerel.c
+++ b/arch/arm/mach-shmobile/board-mackerel.c
@@ -53,6 +53,7 @@
 #include <media/soc_camera.h>
 #include <media/soc_camera_platform.h>
 #include <sound/sh_fsi.h>
+#include <sound/simple_card.h>
 
 #include <mach/common.h>
 #include <mach/irqs.h>
@@ -502,8 +503,26 @@ static struct platform_device hdmi_lcdc_device = {
 	},
 };
 
+static struct asoc_simple_dai_init_info fsi2_hdmi_init_info = {
+	.cpu_daifmt	= SND_SOC_DAIFMT_CBM_CFM,
+};
+
+static struct asoc_simple_card_info fsi2_hdmi_info = {
+	.name		= "HDMI",
+	.card		= "FSI2B-HDMI",
+	.cpu_dai	= "fsib-dai",
+	.codec		= "sh-mobile-hdmi",
+	.platform	= "sh_fsi2",
+	.codec_dai	= "sh_mobile_hdmi-hifi",
+	.init		= &fsi2_hdmi_init_info,
+};
+
 static struct platform_device fsi_hdmi_device = {
-	.name		= "sh_fsi2_b_hdmi",
+	.name	= "asoc-simple-card",
+	.id	= 1,
+	.dev	= {
+		.platform_data	= &fsi2_hdmi_info,
+	},
 };
 
 static void __init hdmi_init_pm_clock(void)
@@ -945,17 +964,25 @@ static struct platform_device fsi_device = {
 	},
 };
 
-static struct fsi_ak4642_info fsi2_ak4643_info = {
+static struct asoc_simple_dai_init_info fsi2_ak4643_init_info = {
+	.fmt		= SND_SOC_DAIFMT_LEFT_J,
+	.codec_daifmt	= SND_SOC_DAIFMT_CBM_CFM,
+	.cpu_daifmt	= SND_SOC_DAIFMT_CBS_CFS,
+	.sysclk		= 11289600,
+};
+
+static struct asoc_simple_card_info fsi2_ak4643_info = {
 	.name		= "AK4643",
 	.card		= "FSI2A-AK4643",
 	.cpu_dai	= "fsia-dai",
 	.codec		= "ak4642-codec.0-0013",
 	.platform	= "sh_fsi2",
-	.id		= FSI_PORT_A,
+	.codec_dai	= "ak4642-hifi",
+	.init		= &fsi2_ak4643_init_info,
 };
 
 static struct platform_device fsi_ak4643_device = {
-	.name	= "fsi-ak4642-audio",
+	.name	= "asoc-simple-card",
 	.dev	= {
 		.platform_data	= &fsi2_ak4643_info,
 	},
diff --git a/arch/arm/mach-tegra/board-dt-tegra20.c b/arch/arm/mach-tegra/board-dt-tegra20.c
index 8351c4c147ad..fac3eb1af17e 100644
--- a/arch/arm/mach-tegra/board-dt-tegra20.c
+++ b/arch/arm/mach-tegra/board-dt-tegra20.c
@@ -55,9 +55,9 @@ struct of_dev_auxdata tegra20_auxdata_lookup[] __initdata = {
 	OF_DEV_AUXDATA("nvidia,tegra20-i2c", TEGRA_I2C2_BASE, "tegra-i2c.1", NULL),
 	OF_DEV_AUXDATA("nvidia,tegra20-i2c", TEGRA_I2C3_BASE, "tegra-i2c.2", NULL),
 	OF_DEV_AUXDATA("nvidia,tegra20-i2c-dvc", TEGRA_DVC_BASE, "tegra-i2c.3", NULL),
-	OF_DEV_AUXDATA("nvidia,tegra20-i2s", TEGRA_I2S1_BASE, "tegra-i2s.0", NULL),
-	OF_DEV_AUXDATA("nvidia,tegra20-i2s", TEGRA_I2S2_BASE, "tegra-i2s.1", NULL),
-	OF_DEV_AUXDATA("nvidia,tegra20-das", TEGRA_APB_MISC_DAS_BASE, "tegra-das", NULL),
+	OF_DEV_AUXDATA("nvidia,tegra20-i2s", TEGRA_I2S1_BASE, "tegra20-i2s.0", NULL),
+	OF_DEV_AUXDATA("nvidia,tegra20-i2s", TEGRA_I2S2_BASE, "tegra20-i2s.1", NULL),
+	OF_DEV_AUXDATA("nvidia,tegra20-das", TEGRA_APB_MISC_DAS_BASE, "tegra20-das", NULL),
 	OF_DEV_AUXDATA("nvidia,tegra20-ehci", TEGRA_USB_BASE, "tegra-ehci.0",
 		       &tegra_ehci1_pdata),
 	OF_DEV_AUXDATA("nvidia,tegra20-ehci", TEGRA_USB2_BASE, "tegra-ehci.1",
diff --git a/arch/arm/mach-tegra/board-harmony.c b/arch/arm/mach-tegra/board-harmony.c
index 222182e00226..b906b3b6077b 100644
--- a/arch/arm/mach-tegra/board-harmony.c
+++ b/arch/arm/mach-tegra/board-harmony.c
@@ -124,7 +124,6 @@ static struct platform_device *harmony_devices[] __initdata = {
 	&tegra_ehci3_device,
 	&tegra_i2s_device1,
 	&tegra_das_device,
-	&tegra_pcm_device,
 	&harmony_audio_device,
 };
 
diff --git a/arch/arm/mach-tegra/board-seaboard.c b/arch/arm/mach-tegra/board-seaboard.c
index 20743bcec03a..79064c7a7907 100644
--- a/arch/arm/mach-tegra/board-seaboard.c
+++ b/arch/arm/mach-tegra/board-seaboard.c
@@ -156,7 +156,6 @@ static struct platform_device *seaboard_devices[] __initdata = {
 	&seaboard_gpio_keys_device,
 	&tegra_i2s_device1,
 	&tegra_das_device,
-	&tegra_pcm_device,
 	&seaboard_audio_device,
 };
 
diff --git a/arch/arm/mach-tegra/board-trimslice.c b/arch/arm/mach-tegra/board-trimslice.c
index 0a00183feeec..bc59b379c6fe 100644
--- a/arch/arm/mach-tegra/board-trimslice.c
+++ b/arch/arm/mach-tegra/board-trimslice.c
@@ -89,7 +89,6 @@ static struct platform_device *trimslice_devices[] __initdata = {
 	&tegra_sdhci_device4,
 	&tegra_i2s_device1,
 	&tegra_das_device,
-	&tegra_pcm_device,
 	&trimslice_audio_device,
 };
 
diff --git a/arch/arm/mach-tegra/devices.c b/arch/arm/mach-tegra/devices.c
index bd3035e0cea1..2d8dfa2faf8f 100644
--- a/arch/arm/mach-tegra/devices.c
+++ b/arch/arm/mach-tegra/devices.c
@@ -674,14 +674,14 @@ static struct resource i2s_resource2[] = {
 };
 
 struct platform_device tegra_i2s_device1 = {
-	.name		= "tegra-i2s",
+	.name		= "tegra20-i2s",
 	.id		= 0,
 	.resource	= i2s_resource1,
 	.num_resources	= ARRAY_SIZE(i2s_resource1),
 };
 
 struct platform_device tegra_i2s_device2 = {
-	.name		= "tegra-i2s",
+	.name		= "tegra20-i2s",
 	.id		= 1,
 	.resource	= i2s_resource2,
 	.num_resources	= ARRAY_SIZE(i2s_resource2),
@@ -696,13 +696,8 @@ static struct resource tegra_das_resources[] = {
 };
 
 struct platform_device tegra_das_device = {
-	.name		= "tegra-das",
+	.name		= "tegra20-das",
 	.id		= -1,
 	.num_resources	= ARRAY_SIZE(tegra_das_resources),
 	.resource	= tegra_das_resources,
 };
-
-struct platform_device tegra_pcm_device = {
-	.name = "tegra-pcm-audio",
-	.id = -1,
-};
diff --git a/arch/arm/mach-tegra/devices.h b/arch/arm/mach-tegra/devices.h
index ec455679b219..138c642e59f4 100644
--- a/arch/arm/mach-tegra/devices.h
+++ b/arch/arm/mach-tegra/devices.h
@@ -52,6 +52,5 @@ extern struct platform_device tegra_pmu_device;
 extern struct platform_device tegra_i2s_device1;
 extern struct platform_device tegra_i2s_device2;
 extern struct platform_device tegra_das_device;
-extern struct platform_device tegra_pcm_device;
 
 #endif
diff --git a/arch/arm/mach-tegra/tegra2_clocks.c b/arch/arm/mach-tegra/tegra2_clocks.c
index 2cae5cbc20ba..bae09b859891 100644
--- a/arch/arm/mach-tegra/tegra2_clocks.c
+++ b/arch/arm/mach-tegra/tegra2_clocks.c
@@ -2148,8 +2148,8 @@ static struct clk tegra_list_clks[] = {
 	PERIPH_CLK("apbdma",	"tegra-dma",		NULL,	34,	0,	108000000, mux_pclk,			0),
 	PERIPH_CLK("rtc",	"rtc-tegra",		NULL,	4,	0,	32768,     mux_clk_32k,			PERIPH_NO_RESET),
 	PERIPH_CLK("timer",	"timer",		NULL,	5,	0,	26000000,  mux_clk_m,			0),
-	PERIPH_CLK("i2s1",	"tegra-i2s.0",		NULL,	11,	0x100,	26000000,  mux_pllaout0_audio2x_pllp_clkm,	MUX | DIV_U71),
-	PERIPH_CLK("i2s2",	"tegra-i2s.1",		NULL,	18,	0x104,	26000000,  mux_pllaout0_audio2x_pllp_clkm,	MUX | DIV_U71),
+	PERIPH_CLK("i2s1",	"tegra20-i2s.0",	NULL,	11,	0x100,	26000000,  mux_pllaout0_audio2x_pllp_clkm,	MUX | DIV_U71),
+	PERIPH_CLK("i2s2",	"tegra20-i2s.1",	NULL,	18,	0x104,	26000000,  mux_pllaout0_audio2x_pllp_clkm,	MUX | DIV_U71),
 	PERIPH_CLK("spdif_out",	"spdif_out",		NULL,	10,	0x108,	100000000, mux_pllaout0_audio2x_pllp_clkm,	MUX | DIV_U71),
 	PERIPH_CLK("spdif_in",	"spdif_in",		NULL,	10,	0x10c,	100000000, mux_pllp_pllc_pllm,		MUX | DIV_U71),
 	PERIPH_CLK("pwm",	"pwm",			NULL,	17,	0x110,	432000000, mux_pllp_pllc_audio_clkm_clk32,	MUX | DIV_U71),
diff --git a/arch/powerpc/configs/86xx/mpc8610_hpcd_defconfig b/arch/powerpc/configs/86xx/mpc8610_hpcd_defconfig
index 0db9ba0423ff..c09598b31de1 100644
--- a/arch/powerpc/configs/86xx/mpc8610_hpcd_defconfig
+++ b/arch/powerpc/configs/86xx/mpc8610_hpcd_defconfig
@@ -100,6 +100,7 @@ CONFIG_SND_MIXER_OSS=y
 CONFIG_SND_PCM_OSS=y
 # CONFIG_SND_SUPPORT_OLD_API is not set
 CONFIG_SND_SOC=y
+CONFIG_SND_POWERPC_SOC=y
 CONFIG_RTC_CLASS=y
 CONFIG_RTC_DRV_CMOS=y
 CONFIG_EXT2_FS=y
diff --git a/arch/powerpc/configs/mpc85xx_defconfig b/arch/powerpc/configs/mpc85xx_defconfig
index d6b6df5e8743..62bb723c5b54 100644
--- a/arch/powerpc/configs/mpc85xx_defconfig
+++ b/arch/powerpc/configs/mpc85xx_defconfig
@@ -141,6 +141,7 @@ CONFIG_SND_INTEL8X0=y
 # CONFIG_SND_PPC is not set
 # CONFIG_SND_USB is not set
 CONFIG_SND_SOC=y
+CONFIG_SND_POWERPC_SOC=y
 CONFIG_HID_A4TECH=y
 CONFIG_HID_APPLE=y
 CONFIG_HID_BELKIN=y
diff --git a/arch/powerpc/configs/mpc85xx_smp_defconfig b/arch/powerpc/configs/mpc85xx_smp_defconfig
index 5b0e2926becd..d1828427ae55 100644
--- a/arch/powerpc/configs/mpc85xx_smp_defconfig
+++ b/arch/powerpc/configs/mpc85xx_smp_defconfig
@@ -143,6 +143,7 @@ CONFIG_SND_INTEL8X0=y
 # CONFIG_SND_PPC is not set
 # CONFIG_SND_USB is not set
 CONFIG_SND_SOC=y
+CONFIG_SND_POWERPC_SOC=y
 CONFIG_HID_A4TECH=y
 CONFIG_HID_APPLE=y
 CONFIG_HID_BELKIN=y
diff --git a/arch/sh/boards/Kconfig b/arch/sh/boards/Kconfig
index 3a74b10922e6..1f56b35d3248 100644
--- a/arch/sh/boards/Kconfig
+++ b/arch/sh/boards/Kconfig
@@ -54,6 +54,7 @@ config SH_7724_SOLUTION_ENGINE
 	select SOLUTION_ENGINE
 	depends on CPU_SUBTYPE_SH7724
 	select ARCH_REQUIRE_GPIOLIB
+	select SND_SOC_AK4642 if SND_SIMPLE_CARD
 	help
 	  Select 7724 SolutionEngine if configuring for a Hitachi SH7724
 	  evaluation board.
@@ -224,6 +225,7 @@ config SH_ECOVEC
 	bool "EcoVec"
 	depends on CPU_SUBTYPE_SH7724
 	select ARCH_REQUIRE_GPIOLIB
+	select SND_SOC_DA7210 if SND_SIMPLE_CARD
 	help
 	  Renesas "R0P7724LC0011/21RL (EcoVec)" support.
 
diff --git a/arch/sh/boards/mach-ecovec24/setup.c b/arch/sh/boards/mach-ecovec24/setup.c
index 63002c8a0ec8..4158d70c0dea 100644
--- a/arch/sh/boards/mach-ecovec24/setup.c
+++ b/arch/sh/boards/mach-ecovec24/setup.c
@@ -33,6 +33,7 @@
 #include <linux/videodev2.h>
 #include <video/sh_mobile_lcdc.h>
 #include <sound/sh_fsi.h>
+#include <sound/simple_card.h>
 #include <media/sh_mobile_ceu.h>
 #include <media/soc_camera.h>
 #include <media/tw9910.h>
@@ -811,6 +812,30 @@ static struct platform_device fsi_device = {
 	},
 };
 
+static struct asoc_simple_dai_init_info fsi_da7210_init_info = {
+	.fmt		= SND_SOC_DAIFMT_I2S,
+	.codec_daifmt	= SND_SOC_DAIFMT_CBM_CFM,
+	.cpu_daifmt	= SND_SOC_DAIFMT_CBS_CFS,
+};
+
+static struct asoc_simple_card_info fsi_da7210_info = {
+	.name		= "DA7210",
+	.card		= "FSIB-DA7210",
+	.cpu_dai	= "fsib-dai",
+	.codec		= "da7210.0-001a",
+	.platform	= "sh_fsi.0",
+	.codec_dai	= "da7210-hifi",
+	.init		= &fsi_da7210_init_info,
+};
+
+static struct platform_device fsi_da7210_device = {
+	.name	= "asoc-simple-card",
+	.dev	= {
+		.platform_data	= &fsi_da7210_info,
+	},
+};
+
+
 /* IrDA */
 static struct resource irda_resources[] = {
 	[0] = {
@@ -947,6 +972,7 @@ static struct platform_device *ecovec_devices[] __initdata = {
 	&camera_devices[1],
 	&camera_devices[2],
 	&fsi_device,
+	&fsi_da7210_device,
 	&irda_device,
 	&vou_device,
 #if defined(CONFIG_MMC_SH_MMCIF) || defined(CONFIG_MMC_SH_MMCIF_MODULE)
diff --git a/arch/sh/boards/mach-se/7724/setup.c b/arch/sh/boards/mach-se/7724/setup.c
index dd931e36daf8..ffbf5bc7366b 100644
--- a/arch/sh/boards/mach-se/7724/setup.c
+++ b/arch/sh/boards/mach-se/7724/setup.c
@@ -29,6 +29,7 @@
 #include <video/sh_mobile_lcdc.h>
 #include <media/sh_mobile_ceu.h>
 #include <sound/sh_fsi.h>
+#include <sound/simple_card.h>
 #include <asm/io.h>
 #include <asm/heartbeat.h>
 #include <asm/clock.h>
@@ -305,17 +306,25 @@ static struct platform_device fsi_device = {
 	},
 };
 
-static struct fsi_ak4642_info fsi_ak4642_info = {
+static struct asoc_simple_dai_init_info fsi2_ak4642_init_info = {
+	.fmt		= SND_SOC_DAIFMT_LEFT_J,
+	.codec_daifmt	= SND_SOC_DAIFMT_CBM_CFM,
+	.cpu_daifmt	= SND_SOC_DAIFMT_CBS_CFS,
+	.sysclk		= 11289600,
+};
+
+static struct asoc_simple_card_info fsi_ak4642_info = {
 	.name		= "AK4642",
 	.card		= "FSIA-AK4642",
 	.cpu_dai	= "fsia-dai",
 	.codec		= "ak4642-codec.0-0012",
 	.platform	= "sh_fsi.0",
-	.id		= FSI_PORT_A,
+	.codec_dai	= "ak4642-hifi",
+	.init		= &fsi2_ak4642_init_info,
 };
 
 static struct platform_device fsi_ak4642_device = {
-	.name	= "fsi-ak4642-audio",
+	.name	= "asoc-simple-card",
 	.dev	= {
 		.platform_data	= &fsi_ak4642_info,
 	},
diff --git a/drivers/mfd/mc13xxx-core.c b/drivers/mfd/mc13xxx-core.c
index 9fd4f63c45cc..738722cdecaa 100644
--- a/drivers/mfd/mc13xxx-core.c
+++ b/drivers/mfd/mc13xxx-core.c
@@ -813,7 +813,8 @@ err_revision:
 		mc13xxx_add_subdevice(mc13xxx, "%s-adc");
 
 	if (mc13xxx->flags & MC13XXX_USE_CODEC)
-		mc13xxx_add_subdevice(mc13xxx, "%s-codec");
+		mc13xxx_add_subdevice_pdata(mc13xxx, "%s-codec",
+					pdata->codec, sizeof(*pdata->codec));
 
 	if (mc13xxx->flags & MC13XXX_USE_RTC)
 		mc13xxx_add_subdevice(mc13xxx, "%s-rtc");
diff --git a/include/linux/mfd/mc13xxx.h b/include/linux/mfd/mc13xxx.h
index 10e038bac8dd..bf070755982e 100644
--- a/include/linux/mfd/mc13xxx.h
+++ b/include/linux/mfd/mc13xxx.h
@@ -170,6 +170,16 @@ struct mc13xxx_ts_platform_data {
 	bool atox;
 };
 
+enum mc13783_ssi_port {
+	MC13783_SSI1_PORT,
+	MC13783_SSI2_PORT,
+};
+
+struct mc13xxx_codec_platform_data {
+	enum mc13783_ssi_port adc_ssi_port;
+	enum mc13783_ssi_port dac_ssi_port;
+};
+
 struct mc13xxx_platform_data {
 #define MC13XXX_USE_TOUCHSCREEN (1 << 0)
 #define MC13XXX_USE_CODEC	(1 << 1)
@@ -181,6 +191,7 @@ struct mc13xxx_platform_data {
 	struct mc13xxx_leds_platform_data *leds;
 	struct mc13xxx_buttons_platform_data *buttons;
 	struct mc13xxx_ts_platform_data touch;
+	struct mc13xxx_codec_platform_data *codec;
 };
 
 #define MC13XXX_ADC_MODE_TS		1
diff --git a/include/sound/asound.h b/include/sound/asound.h
index a2e4ff5ba9e9..0876a1e76aef 100644
--- a/include/sound/asound.h
+++ b/include/sound/asound.h
@@ -68,6 +68,20 @@ struct snd_aes_iec958 {
 	unsigned char dig_subframe[4];	/* AES/IEC958 subframe bits */
 };
 
+/****************************************************************************
+ *                                                                          *
+ *        CEA-861 Audio InfoFrame. Used in HDMI and DisplayPort		    *
+ *                                                                          *
+ ****************************************************************************/
+
+struct snd_cea_861_aud_if {
+	unsigned char db1_ct_cc; /* coding type and channel count */
+	unsigned char db2_sf_ss; /* sample frequency and size */
+	unsigned char db3; /* not used, all zeros */
+	unsigned char db4_ca; /* channel allocation code */
+	unsigned char db5_dminh_lsv; /* downmix inhibit & level-shit values */
+};
+
 /****************************************************************************
  *                                                                          *
  *      Section for driver hardware dependent interface - /dev/snd/hw?      *
diff --git a/include/sound/asoundef.h b/include/sound/asoundef.h
index 20ebf3298eba..bb05c02f89b0 100644
--- a/include/sound/asoundef.h
+++ b/include/sound/asoundef.h
@@ -170,6 +170,47 @@
 #define IEC958_AES5_CON_CGMSA_COPYNOMORE (2<<0)	/* condition not be used */
 #define IEC958_AES5_CON_CGMSA_COPYNEVER	(3<<0)	/* no copying is permitted */
 
+/****************************************************************************
+ *                                                                          *
+ *        CEA-861 Audio InfoFrame. Used in HDMI and DisplayPort		    *
+ *                                                                          *
+ ****************************************************************************/
+#define CEA861_AUDIO_INFOFRAME_DB1CC		(7<<0) /* mask - channel count */
+#define CEA861_AUDIO_INFOFRAME_DB1CT		(0xf<<4) /* mask - coding type */
+#define CEA861_AUDIO_INFOFRAME_DB1CT_FROM_STREAM (0<<4) /* refer to stream */
+#define CEA861_AUDIO_INFOFRAME_DB1CT_IEC60958	(1<<4) /* IEC-60958 L-PCM */
+#define CEA861_AUDIO_INFOFRAME_DB1CT_AC3	(2<<4) /* AC-3 */
+#define CEA861_AUDIO_INFOFRAME_DB1CT_MPEG1	(3<<4) /* MPEG1 Layers 1 & 2 */
+#define CEA861_AUDIO_INFOFRAME_DB1CT_MP3	(4<<4) /* MPEG1 Layer 3 */
+#define CEA861_AUDIO_INFOFRAME_DB1CT_MPEG2_MULTICH (5<<4) /* MPEG2 Multichannel */
+#define CEA861_AUDIO_INFOFRAME_DB1CT_AAC	(6<<4) /* AAC */
+#define CEA861_AUDIO_INFOFRAME_DB1CT_DTS	(7<<4) /* DTS */
+#define CEA861_AUDIO_INFOFRAME_DB1CT_ATRAC	(8<<4) /* ATRAC */
+#define CEA861_AUDIO_INFOFRAME_DB1CT_ONEBIT	(9<<4) /* One Bit Audio */
+#define CEA861_AUDIO_INFOFRAME_DB1CT_DOLBY_DIG_PLUS (10<<4) /* Dolby Digital + */
+#define CEA861_AUDIO_INFOFRAME_DB1CT_DTS_HD	(11<<4) /* DTS-HD */
+#define CEA861_AUDIO_INFOFRAME_DB1CT_MAT	(12<<4) /* MAT (MLP) */
+#define CEA861_AUDIO_INFOFRAME_DB1CT_DST	(13<<4) /* DST */
+#define CEA861_AUDIO_INFOFRAME_DB1CT_WMA_PRO	(14<<4) /* WMA Pro */
+#define CEA861_AUDIO_INFOFRAME_DB2SF		(7<<2) /* mask - sample frequency */
+#define CEA861_AUDIO_INFOFRAME_DB2SF_FROM_STREAM (0<<2) /* refer to stream */
+#define CEA861_AUDIO_INFOFRAME_DB2SF_32000	(1<<2) /* 32kHz */
+#define CEA861_AUDIO_INFOFRAME_DB2SF_44100	(2<<2) /* 44.1kHz */
+#define CEA861_AUDIO_INFOFRAME_DB2SF_48000	(3<<2) /* 48kHz */
+#define CEA861_AUDIO_INFOFRAME_DB2SF_88200	(4<<2) /* 88.2kHz */
+#define CEA861_AUDIO_INFOFRAME_DB2SF_96000	(5<<2) /* 96kHz */
+#define CEA861_AUDIO_INFOFRAME_DB2SF_176400	(6<<2) /* 176.4kHz */
+#define CEA861_AUDIO_INFOFRAME_DB2SF_192000	(7<<2) /* 192kHz */
+#define CEA861_AUDIO_INFOFRAME_DB2SS		(3<<0) /* mask - sample size */
+#define CEA861_AUDIO_INFOFRAME_DB2SS_FROM_STREAM (0<<0) /* refer to stream */
+#define CEA861_AUDIO_INFOFRAME_DB2SS_16BIT	(1<<0) /* 16 bits */
+#define CEA861_AUDIO_INFOFRAME_DB2SS_20BIT	(2<<0) /* 20 bits */
+#define CEA861_AUDIO_INFOFRAME_DB2SS_24BIT	(3<<0) /* 24 bits */
+#define CEA861_AUDIO_INFOFRAME_DB5_DM_INH	(1<<7) /* mask - inhibit downmixing */
+#define CEA861_AUDIO_INFOFRAME_DB5_DM_INH_PERMITTED (0<<7) /* stereo downmix permitted */
+#define CEA861_AUDIO_INFOFRAME_DB5_DM_INH_PROHIBITED (1<<7) /* stereo downmis prohibited */
+#define CEA861_AUDIO_INFOFRAME_DB5_LSV		(0xf<<3) /* mask - level-shift values */
+
 /*****************************************************************************
  *                                                                           *
  *                            MIDI v1.0 interface                            *
diff --git a/include/sound/cs42l52.h b/include/sound/cs42l52.h
new file mode 100644
index 000000000000..4c68955f7330
--- /dev/null
+++ b/include/sound/cs42l52.h
@@ -0,0 +1,36 @@
+/*
+ * linux/sound/cs42l52.h -- Platform data for CS42L52
+ *
+ * Copyright (c) 2012 Cirrus Logic Inc.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef __CS42L52_H
+#define __CS42L52_H
+
+struct cs42l52_platform_data {
+
+	/* MICBIAS Level. Check datasheet Pg48 */
+	unsigned int micbias_lvl;
+
+	/* MICA mode selection 0=Single 1=Differential */
+	unsigned int mica_cfg;
+
+	/* MICB mode selection 0=Single 1=Differential */
+	unsigned int micb_cfg;
+
+	/* MICA Select 0=MIC1A 1=MIC2A */
+	unsigned int mica_sel;
+
+	/* MICB Select 0=MIC2A 1=MIC2B */
+	unsigned int micb_sel;
+
+	/* Charge Pump Freq. Check datasheet Pg73 */
+	unsigned int chgfreq;
+
+};
+
+#endif /* __CS42L52_H */
diff --git a/include/sound/max98095.h b/include/sound/max98095.h
index 7513a42dd4aa..e87ae67b0a55 100644
--- a/include/sound/max98095.h
+++ b/include/sound/max98095.h
@@ -49,6 +49,18 @@ struct max98095_pdata {
 	 */
 	unsigned int digmic_left_mode:1;
 	unsigned int digmic_right_mode:1;
+
+	/* Pin5 is the mechanical method of sensing jack insertion
+	 * but it is something that might not be supported.
+	 * 0 = PIN5 not supported
+	 * 1 = PIN5 supported
+	 */
+	unsigned int jack_detect_pin5en:1;
+
+	/* Slew amount for jack detection. Calculated as 4 * (delay + 1).
+	 * Default delay is 24 to get a time of 100ms.
+	 */
+	unsigned int jack_detect_delay;
 };
 
 #endif
diff --git a/include/sound/sh_fsi.h b/include/sound/sh_fsi.h
index b457e87fbd08..906010344dd7 100644
--- a/include/sound/sh_fsi.h
+++ b/include/sound/sh_fsi.h
@@ -21,10 +21,11 @@
 /*
  * flags format
  *
- * 0x000000BA
+ * 0x00000CBA
  *
  * A:  inversion
  * B:  format mode
+ * C:  chip specific
  */
 
 /* A: clock inversion */
@@ -39,6 +40,9 @@
 #define SH_FSI_FMT_DAI		(0 << 4)
 #define SH_FSI_FMT_SPDIF	(1 << 4)
 
+/* C: chip specific */
+#define SH_FSI_OPTION_MASK	0x00000F00
+#define SH_FSI_ENABLE_STREAM_MODE	(1 << 8) /* for 16bit data */
 
 /*
  * set_rate return value
@@ -84,16 +88,4 @@ struct sh_fsi_platform_info {
 	struct sh_fsi_port_info port_b;
 };
 
-/*
- * for fsi-ak4642
- */
-struct fsi_ak4642_info {
-	const char *name;
-	const char *card;
-	const char *cpu_dai;
-	const char *codec;
-	const char *platform;
-	int id;
-};
-
 #endif /* __SOUND_FSI_H */
diff --git a/include/sound/simple_card.h b/include/sound/simple_card.h
new file mode 100644
index 000000000000..4b62b8dc6a4f
--- /dev/null
+++ b/include/sound/simple_card.h
@@ -0,0 +1,38 @@
+/*
+ * ASoC simple sound card support
+ *
+ * Copyright (C) 2012 Renesas Solutions Corp.
+ * Kuninori Morimoto <kuninori.morimoto.gx@renesas.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.
+ */
+
+#ifndef __SIMPLE_CARD_H
+#define __SIMPLE_CARD_H
+
+#include <sound/soc.h>
+
+struct asoc_simple_dai_init_info {
+	unsigned int fmt;
+	unsigned int cpu_daifmt;
+	unsigned int codec_daifmt;
+	unsigned int sysclk;
+};
+
+struct asoc_simple_card_info {
+	const char *name;
+	const char *card;
+	const char *cpu_dai;
+	const char *codec;
+	const char *platform;
+	const char *codec_dai;
+	struct asoc_simple_dai_init_info *init; /* for snd_link.init */
+
+	/* used in simple-card.c */
+	struct snd_soc_dai_link snd_link;
+	struct snd_soc_card snd_card;
+};
+
+#endif /* __SIMPLE_CARD_H */
diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h
index c429f248cf4e..1f69e0af2941 100644
--- a/include/sound/soc-dai.h
+++ b/include/sound/soc-dai.h
@@ -173,6 +173,8 @@ struct snd_soc_dai_ops {
 		struct snd_soc_dai *);
 	int (*trigger)(struct snd_pcm_substream *, int,
 		struct snd_soc_dai *);
+	int (*bespoke_trigger)(struct snd_pcm_substream *, int,
+		struct snd_soc_dai *);
 	/*
 	 * For hardware based FIFO caused delay reporting.
 	 * Optional.
@@ -196,6 +198,7 @@ struct snd_soc_dai_driver {
 	const char *name;
 	unsigned int id;
 	int ac97_control;
+	unsigned int base;
 
 	/* DAI driver callbacks */
 	int (*probe)(struct snd_soc_dai *dai);
@@ -241,6 +244,7 @@ struct snd_soc_dai {
 
 	struct snd_soc_dapm_widget *playback_widget;
 	struct snd_soc_dapm_widget *capture_widget;
+	struct snd_soc_dapm_context dapm;
 
 	/* DAI DMA data */
 	void *playback_dma_data;
diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h
index 8da3c2409060..e3833d9f1914 100644
--- a/include/sound/soc-dapm.h
+++ b/include/sound/soc-dapm.h
@@ -141,10 +141,6 @@ struct device;
 {       .id = snd_soc_dapm_mixer, .name = wname, .reg = wreg, .shift = wshift, \
 	.invert = winvert, .kcontrol_news = wcontrols, \
 	.num_kcontrols = wncontrols, .event = wevent, .event_flags = wflags}
-#define SND_SOC_DAPM_MICBIAS_E(wname, wreg, wshift, winvert, wevent, wflags) \
-{	.id = snd_soc_dapm_micbias, .name = wname, .reg = wreg, .shift = wshift, \
-	.invert = winvert, .kcontrol_news = NULL, .num_kcontrols = 0, \
-	.event = wevent, .event_flags = wflags}
 #define SND_SOC_DAPM_SWITCH_E(wname, wreg, wshift, winvert, wcontrols, \
 	wevent, wflags) \
 {	.id = snd_soc_dapm_switch, .name = wname, .reg = wreg, .shift = wshift, \
@@ -324,6 +320,8 @@ struct snd_soc_dapm_path;
 struct snd_soc_dapm_pin;
 struct snd_soc_dapm_route;
 struct snd_soc_dapm_context;
+struct regulator;
+struct snd_soc_dapm_widget_list;
 
 int dapm_reg_event(struct snd_soc_dapm_widget *w,
 		   struct snd_kcontrol *kcontrol, int event);
@@ -359,6 +357,10 @@ int snd_soc_dapm_new_controls(struct snd_soc_dapm_context *dapm,
 int snd_soc_dapm_new_dai_widgets(struct snd_soc_dapm_context *dapm,
 				 struct snd_soc_dai *dai);
 int snd_soc_dapm_link_dai_widgets(struct snd_soc_card *card);
+int snd_soc_dapm_new_pcm(struct snd_soc_card *card,
+			 const struct snd_soc_pcm_stream *params,
+			 struct snd_soc_dapm_widget *source,
+			 struct snd_soc_dapm_widget *sink);
 
 /* dapm path setup */
 int snd_soc_dapm_new_widgets(struct snd_soc_dapm_context *dapm);
@@ -369,8 +371,8 @@ int snd_soc_dapm_weak_routes(struct snd_soc_dapm_context *dapm,
 			     const struct snd_soc_dapm_route *route, int num);
 
 /* dapm events */
-int snd_soc_dapm_stream_event(struct snd_soc_pcm_runtime *rtd, int stream,
-			      struct snd_soc_dai *dai, int event);
+void snd_soc_dapm_stream_event(struct snd_soc_pcm_runtime *rtd, int stream,
+	int event);
 void snd_soc_dapm_shutdown(struct snd_soc_card *card);
 
 /* external DAPM widget events */
@@ -402,6 +404,10 @@ void snd_soc_dapm_auto_nc_codec_pins(struct snd_soc_codec *codec);
 /* Mostly internal - should not normally be used */
 void dapm_mark_dirty(struct snd_soc_dapm_widget *w, const char *reason);
 
+/* dapm path query */
+int snd_soc_dapm_dai_get_connected_widgets(struct snd_soc_dai *dai, int stream,
+	struct snd_soc_dapm_widget_list **list);
+
 /* dapm widget types */
 enum snd_soc_dapm_type {
 	snd_soc_dapm_input = 0,		/* input pin */
@@ -430,6 +436,12 @@ enum snd_soc_dapm_type {
 	snd_soc_dapm_aif_out,		/* audio interface output */
 	snd_soc_dapm_siggen,		/* signal generator */
 	snd_soc_dapm_dai,		/* link to DAI structure */
+	snd_soc_dapm_dai_link,		/* link between two DAI structures */
+};
+
+enum snd_soc_dapm_subclass {
+	SND_SOC_DAPM_CLASS_INIT		= 0,
+	SND_SOC_DAPM_CLASS_RUNTIME	= 1,
 };
 
 /*
@@ -482,9 +494,11 @@ struct snd_soc_dapm_widget {
 	struct snd_soc_dapm_context *dapm;
 
 	void *priv;				/* widget specific data */
+	struct regulator *regulator;		/* attached regulator */
+	const struct snd_soc_pcm_stream *params; /* params for dai links */
 
 	/* dapm control */
-	short reg;						/* negative reg = no direct dapm */
+	int reg;				/* negative reg = no direct dapm */
 	unsigned char shift;			/* bits to shift */
 	unsigned int saved_value;		/* widget saved value */
 	unsigned int value;				/* widget current value */
diff --git a/include/sound/soc-dpcm.h b/include/sound/soc-dpcm.h
new file mode 100644
index 000000000000..04598f1efd77
--- /dev/null
+++ b/include/sound/soc-dpcm.h
@@ -0,0 +1,138 @@
+/*
+ * linux/sound/soc-dpcm.h -- ALSA SoC Dynamic PCM Support
+ *
+ * Author:		Liam Girdwood <lrg@ti.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.
+ */
+
+#ifndef __LINUX_SND_SOC_DPCM_H
+#define __LINUX_SND_SOC_DPCM_H
+
+#include <linux/list.h>
+#include <sound/pcm.h>
+
+struct snd_soc_pcm_runtime;
+
+/*
+ * Types of runtime_update to perform. e.g. originated from FE PCM ops
+ * or audio route changes triggered by muxes/mixers.
+ */
+enum snd_soc_dpcm_update {
+	SND_SOC_DPCM_UPDATE_NO	= 0,
+	SND_SOC_DPCM_UPDATE_BE,
+	SND_SOC_DPCM_UPDATE_FE,
+};
+
+/*
+ * Dynamic PCM Frontend -> Backend link management states.
+ */
+enum snd_soc_dpcm_link_state {
+	SND_SOC_DPCM_LINK_STATE_NEW	= 0,	/* newly created link */
+	SND_SOC_DPCM_LINK_STATE_FREE,		/* link to be dismantled */
+};
+
+/*
+ * Dynamic PCM Frontend -> Backend link PCM states.
+ */
+enum snd_soc_dpcm_state {
+	SND_SOC_DPCM_STATE_NEW	= 0,
+	SND_SOC_DPCM_STATE_OPEN,
+	SND_SOC_DPCM_STATE_HW_PARAMS,
+	SND_SOC_DPCM_STATE_PREPARE,
+	SND_SOC_DPCM_STATE_START,
+	SND_SOC_DPCM_STATE_STOP,
+	SND_SOC_DPCM_STATE_PAUSED,
+	SND_SOC_DPCM_STATE_SUSPEND,
+	SND_SOC_DPCM_STATE_HW_FREE,
+	SND_SOC_DPCM_STATE_CLOSE,
+};
+
+/*
+ * Dynamic PCM trigger ordering. Triggering flexibility is required as some
+ * DSPs require triggering before/after their CPU platform and DAIs.
+ *
+ * i.e. some clients may want to manually order this call in their PCM
+ * trigger() whilst others will just use the regular core ordering.
+ */
+enum snd_soc_dpcm_trigger {
+	SND_SOC_DPCM_TRIGGER_PRE		= 0,
+	SND_SOC_DPCM_TRIGGER_POST,
+	SND_SOC_DPCM_TRIGGER_BESPOKE,
+};
+
+/*
+ * Dynamic PCM link
+ * This links together a FE and BE DAI at runtime and stores the link
+ * state information and the hw_params configuration.
+ */
+struct snd_soc_dpcm {
+	/* FE and BE DAIs*/
+	struct snd_soc_pcm_runtime *be;
+	struct snd_soc_pcm_runtime *fe;
+
+	/* link state */
+	enum snd_soc_dpcm_link_state state;
+
+	/* list of BE and FE for this DPCM link */
+	struct list_head list_be;
+	struct list_head list_fe;
+
+	/* hw params for this link - may be different for each link */
+	struct snd_pcm_hw_params hw_params;
+#ifdef CONFIG_DEBUG_FS
+	struct dentry *debugfs_state;
+#endif
+};
+
+/*
+ * Dynamic PCM runtime data.
+ */
+struct snd_soc_dpcm_runtime {
+	struct list_head be_clients;
+	struct list_head fe_clients;
+
+	int users;
+	struct snd_pcm_runtime *runtime;
+	struct snd_pcm_hw_params hw_params;
+
+	/* state and update */
+	enum snd_soc_dpcm_update runtime_update;
+	enum snd_soc_dpcm_state state;
+};
+
+/* can this BE stop and free */
+int snd_soc_dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe,
+		struct snd_soc_pcm_runtime *be, int stream);
+
+/* can this BE perform a hw_params() */
+int snd_soc_dpcm_can_be_params(struct snd_soc_pcm_runtime *fe,
+		struct snd_soc_pcm_runtime *be, int stream);
+
+/* is the current PCM operation for this FE ? */
+int snd_soc_dpcm_fe_can_update(struct snd_soc_pcm_runtime *fe, int stream);
+
+/* is the current PCM operation for this BE ? */
+int snd_soc_dpcm_be_can_update(struct snd_soc_pcm_runtime *fe,
+		struct snd_soc_pcm_runtime *be, int stream);
+
+/* get the substream for this BE */
+struct snd_pcm_substream *
+	snd_soc_dpcm_get_substream(struct snd_soc_pcm_runtime *be, int stream);
+
+/* get the BE runtime state */
+enum snd_soc_dpcm_state
+	snd_soc_dpcm_be_get_state(struct snd_soc_pcm_runtime *be, int stream);
+
+/* set the BE runtime state */
+void snd_soc_dpcm_be_set_state(struct snd_soc_pcm_runtime *be, int stream,
+	enum snd_soc_dpcm_state state);
+
+/* internal use only */
+int soc_dpcm_be_digital_mute(struct snd_soc_pcm_runtime *fe, int mute);
+int soc_dpcm_debugfs_add(struct snd_soc_pcm_runtime *rtd);
+int soc_dpcm_runtime_update(struct snd_soc_dapm_widget *);
+
+#endif
diff --git a/include/sound/soc.h b/include/sound/soc.h
index 2ebf7877c148..c703871f5f65 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -55,6 +55,18 @@
 	.info = snd_soc_info_volsw, .get = snd_soc_get_volsw,\
 	.put = snd_soc_put_volsw, \
 	.private_value =  SOC_SINGLE_VALUE(reg, shift, max, invert) }
+#define SOC_SINGLE_SX_TLV(xname, xreg, xshift, xmin, xmax, tlv_array) \
+{       .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
+	.access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \
+	SNDRV_CTL_ELEM_ACCESS_READWRITE, \
+	.tlv.p  = (tlv_array),\
+	.info = snd_soc_info_volsw, \
+	.get = snd_soc_get_volsw_sx,\
+	.put = snd_soc_put_volsw_sx, \
+	.private_value = (unsigned long)&(struct soc_mixer_control) \
+		{.reg = xreg, .rreg = xreg, \
+		.shift = xshift, .rshift = xshift, \
+		.max = xmax, .min = xmin} }
 #define SOC_DOUBLE(xname, reg, shift_left, shift_right, max, invert) \
 {	.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\
 	.info = snd_soc_info_volsw, .get = snd_soc_get_volsw, \
@@ -85,6 +97,18 @@
 	.get = snd_soc_get_volsw, .put = snd_soc_put_volsw, \
 	.private_value = SOC_DOUBLE_R_VALUE(reg_left, reg_right, xshift, \
 					    xmax, xinvert) }
+#define SOC_DOUBLE_R_SX_TLV(xname, xreg, xrreg, xshift, xmin, xmax, tlv_array) \
+{       .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \
+	.access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \
+	SNDRV_CTL_ELEM_ACCESS_READWRITE, \
+	.tlv.p  = (tlv_array), \
+	.info = snd_soc_info_volsw, \
+	.get = snd_soc_get_volsw_sx, \
+	.put = snd_soc_put_volsw_sx, \
+	.private_value = (unsigned long)&(struct soc_mixer_control) \
+		{.reg = xreg, .rreg = xrreg, \
+		.shift = xshift, .rshift = xshift, \
+		.max = xmax, .min = xmin} }
 #define SOC_DOUBLE_S8_TLV(xname, xreg, xmin, xmax, tlv_array) \
 {	.iface  = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \
 	.access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \
@@ -171,20 +195,6 @@
 	.get = xhandler_get, .put = xhandler_put, \
 	.private_value = (unsigned long)&xenum }
 
-#define SOC_DOUBLE_R_SX_TLV(xname, xreg_left, xreg_right, xshift,\
-		xmin, xmax, tlv_array) \
-{	.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \
-	.access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \
-		  SNDRV_CTL_ELEM_ACCESS_READWRITE, \
-	.tlv.p = (tlv_array), \
-	.info = snd_soc_info_volsw_2r_sx, \
-	.get = snd_soc_get_volsw_2r_sx, \
-	.put = snd_soc_put_volsw_2r_sx, \
-	.private_value = (unsigned long)&(struct soc_mixer_control) \
-		{.reg = xreg_left, \
-		 .rreg = xreg_right, .shift = xshift, \
-		 .min = xmin, .max = xmax} }
-
 #define SND_SOC_BYTES(xname, xbase, xregs)		      \
 {	.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname,   \
 	.info = snd_soc_bytes_info, .get = snd_soc_bytes_get, \
@@ -200,6 +210,19 @@
 		{.base = xbase, .num_regs = xregs,	      \
 		 .mask = xmask }) }
 
+#define SOC_SINGLE_XR_SX(xname, xregbase, xregcount, xnbits, \
+		xmin, xmax, xinvert) \
+{	.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \
+	.info = snd_soc_info_xr_sx, .get = snd_soc_get_xr_sx, \
+	.put = snd_soc_put_xr_sx, \
+	.private_value = (unsigned long)&(struct soc_mreg_control) \
+		{.regbase = xregbase, .regcount = xregcount, .nbits = xnbits, \
+		.invert = xinvert, .min = xmin, .max = xmax} }
+
+#define SOC_SINGLE_STROBE(xname, xreg, xshift, xinvert) \
+	SOC_SINGLE_EXT(xname, xreg, xshift, 1, xinvert, \
+		snd_soc_get_strobe, snd_soc_put_strobe)
+
 /*
  * Simplified versions of above macros, declaring a struct and calculating
  * ARRAY_SIZE internally
@@ -264,6 +287,7 @@ struct snd_soc_jack_zone;
 struct snd_soc_jack_pin;
 struct snd_soc_cache_ops;
 #include <sound/soc-dapm.h>
+#include <sound/soc-dpcm.h>
 
 #ifdef CONFIG_GPIOLIB
 struct snd_soc_jack_gpio;
@@ -288,6 +312,11 @@ enum snd_soc_pcm_subclass {
 	SND_SOC_PCM_CLASS_BE	= 1,
 };
 
+enum snd_soc_card_subclass {
+	SND_SOC_CARD_CLASS_INIT		= 0,
+	SND_SOC_CARD_CLASS_RUNTIME	= 1,
+};
+
 int snd_soc_codec_set_sysclk(struct snd_soc_codec *codec, int clk_id,
 			     int source, unsigned int freq, int dir);
 int snd_soc_codec_set_pll(struct snd_soc_codec *codec, int pll_id, int source,
@@ -333,6 +362,11 @@ int snd_soc_platform_write(struct snd_soc_platform *platform,
 					unsigned int reg, unsigned int val);
 int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num);
 
+struct snd_pcm_substream *snd_soc_get_dai_substream(struct snd_soc_card *card,
+		const char *dai_link, int stream);
+struct snd_soc_pcm_runtime *snd_soc_get_pcm_runtime(struct snd_soc_card *card,
+		const char *dai_link);
+
 /* Utility functions to get clock rates from various things */
 int snd_soc_calc_frame_size(int sample_size, int channels, int tdm_slots);
 int snd_soc_params_to_frame_size(struct snd_pcm_hw_params *params);
@@ -343,6 +377,9 @@ int snd_soc_params_to_bclk(struct snd_pcm_hw_params *parms);
 int snd_soc_set_runtime_hwparams(struct snd_pcm_substream *substream,
 	const struct snd_pcm_hardware *hw);
 
+int snd_soc_platform_trigger(struct snd_pcm_substream *substream,
+		int cmd, struct snd_soc_platform *platform);
+
 /* Jack reporting */
 int snd_soc_jack_new(struct snd_soc_codec *codec, const char *id, int type,
 		     struct snd_soc_jack *jack);
@@ -413,6 +450,10 @@ int snd_soc_put_volsw(struct snd_kcontrol *kcontrol,
 	struct snd_ctl_elem_value *ucontrol);
 #define snd_soc_get_volsw_2r snd_soc_get_volsw
 #define snd_soc_put_volsw_2r snd_soc_put_volsw
+int snd_soc_get_volsw_sx(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol);
+int snd_soc_put_volsw_sx(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol);
 int snd_soc_info_volsw_s8(struct snd_kcontrol *kcontrol,
 	struct snd_ctl_elem_info *uinfo);
 int snd_soc_get_volsw_s8(struct snd_kcontrol *kcontrol,
@@ -421,19 +462,22 @@ int snd_soc_put_volsw_s8(struct snd_kcontrol *kcontrol,
 	struct snd_ctl_elem_value *ucontrol);
 int snd_soc_limit_volume(struct snd_soc_codec *codec,
 	const char *name, int max);
-int snd_soc_info_volsw_2r_sx(struct snd_kcontrol *kcontrol,
-	struct snd_ctl_elem_info *uinfo);
-int snd_soc_get_volsw_2r_sx(struct snd_kcontrol *kcontrol,
-	struct snd_ctl_elem_value *ucontrol);
-int snd_soc_put_volsw_2r_sx(struct snd_kcontrol *kcontrol,
-	struct snd_ctl_elem_value *ucontrol);
 int snd_soc_bytes_info(struct snd_kcontrol *kcontrol,
 		       struct snd_ctl_elem_info *uinfo);
 int snd_soc_bytes_get(struct snd_kcontrol *kcontrol,
 		      struct snd_ctl_elem_value *ucontrol);
 int snd_soc_bytes_put(struct snd_kcontrol *kcontrol,
 		      struct snd_ctl_elem_value *ucontrol);
-
+int snd_soc_info_xr_sx(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_info *uinfo);
+int snd_soc_get_xr_sx(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol);
+int snd_soc_put_xr_sx(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol);
+int snd_soc_get_strobe(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol);
+int snd_soc_put_strobe(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol);
 
 /**
  * struct snd_soc_reg_access - Describes whether a given register is
@@ -513,6 +557,7 @@ struct snd_soc_jack_gpio {
 #endif
 
 struct snd_soc_jack {
+	struct mutex mutex;
 	struct snd_jack *jack;
 	struct snd_soc_codec *codec;
 	struct list_head pins;
@@ -711,6 +756,7 @@ struct snd_soc_platform_driver {
 	/* platform IO - used for platform DAPM */
 	unsigned int (*read)(struct snd_soc_platform *, unsigned int);
 	int (*write)(struct snd_soc_platform *, unsigned int, unsigned int);
+	int (*bespoke_trigger)(struct snd_pcm_substream *, int);
 };
 
 struct snd_soc_platform {
@@ -746,21 +792,36 @@ struct snd_soc_dai_link {
 	const char *cpu_dai_name;
 	const struct device_node *cpu_dai_of_node;
 	const char *codec_dai_name;
+	int be_id;	/* optional ID for machine driver BE identification */
+
+	const struct snd_soc_pcm_stream *params;
 
 	unsigned int dai_fmt;           /* format to set on init */
 
+	enum snd_soc_dpcm_trigger trigger[2]; /* trigger type for DPCM */
+
 	/* Keep DAI active over suspend */
 	unsigned int ignore_suspend:1;
 
 	/* Symmetry requirements */
 	unsigned int symmetric_rates:1;
 
+	/* Do not create a PCM for this DAI link (Backend link) */
+	unsigned int no_pcm:1;
+
+	/* This DAI link can route to other DAI links at runtime (Frontend)*/
+	unsigned int dynamic:1;
+
 	/* pmdown_time is ignored at stop */
 	unsigned int ignore_pmdown_time:1;
 
 	/* codec/machine specific init - e.g. add machine controls */
 	int (*init)(struct snd_soc_pcm_runtime *rtd);
 
+	/* optional hw_params re-writing for BE and FE sync */
+	int (*be_hw_params_fixup)(struct snd_soc_pcm_runtime *rtd,
+			struct snd_pcm_hw_params *params);
+
 	/* machine stream operations */
 	struct snd_soc_ops *ops;
 };
@@ -800,6 +861,7 @@ struct snd_soc_card {
 
 	struct list_head list;
 	struct mutex mutex;
+	struct mutex dapm_mutex;
 
 	bool instantiated;
 
@@ -889,9 +951,11 @@ struct snd_soc_pcm_runtime {
 	enum snd_soc_pcm_subclass pcm_subclass;
 	struct snd_pcm_ops ops;
 
-	unsigned int complete:1;
 	unsigned int dev_registered:1;
 
+	/* Dynamic PCM BE runtime data */
+	struct snd_soc_dpcm_runtime dpcm[2];
+
 	long pmdown_time;
 
 	/* runtime devices */
@@ -902,6 +966,10 @@ struct snd_soc_pcm_runtime {
 	struct snd_soc_dai *cpu_dai;
 
 	struct delayed_work delayed_work;
+#ifdef CONFIG_DEBUG_FS
+	struct dentry *debugfs_dpcm_root;
+	struct dentry *debugfs_dpcm_state;
+#endif
 };
 
 /* mixer control */
@@ -916,6 +984,12 @@ struct soc_bytes {
 	u32 mask;
 };
 
+/* multi register control */
+struct soc_mreg_control {
+	long min, max;
+	unsigned int regbase, regcount, nbits, invert;
+};
+
 /* enumerated kcontrol */
 struct soc_enum {
 	unsigned short reg;
diff --git a/include/trace/events/asoc.h b/include/trace/events/asoc.h
index ab26f8aa3c78..5fc2dcdd21cd 100644
--- a/include/trace/events/asoc.h
+++ b/include/trace/events/asoc.h
@@ -7,6 +7,8 @@
 #include <linux/ktime.h>
 #include <linux/tracepoint.h>
 
+#define DAPM_DIRECT "(direct)"
+
 struct snd_soc_jack;
 struct snd_soc_codec;
 struct snd_soc_platform;
@@ -241,6 +243,84 @@ TRACE_EVENT(snd_soc_dapm_walk_done,
 		  (int)__entry->path_checks, (int)__entry->neighbour_checks)
 );
 
+TRACE_EVENT(snd_soc_dapm_output_path,
+
+	TP_PROTO(struct snd_soc_dapm_widget *widget,
+		struct snd_soc_dapm_path *path),
+
+	TP_ARGS(widget, path),
+
+	TP_STRUCT__entry(
+		__string(	wname,	widget->name		)
+		__string(	pname,	path->name ? path->name : DAPM_DIRECT)
+		__string(	psname,	path->sink->name	)
+		__field(	int,	path_sink		)
+		__field(	int,	path_connect		)
+	),
+
+	TP_fast_assign(
+		__assign_str(wname, widget->name);
+		__assign_str(pname, path->name ? path->name : DAPM_DIRECT);
+		__assign_str(psname, path->sink->name);
+		__entry->path_connect = path->connect;
+		__entry->path_sink = (long)path->sink;
+	),
+
+	TP_printk("%c%s -> %s -> %s\n",
+		(int) __entry->path_sink &&
+		(int) __entry->path_connect ? '*' : ' ',
+		__get_str(wname), __get_str(pname), __get_str(psname))
+);
+
+TRACE_EVENT(snd_soc_dapm_input_path,
+
+	TP_PROTO(struct snd_soc_dapm_widget *widget,
+		struct snd_soc_dapm_path *path),
+
+	TP_ARGS(widget, path),
+
+	TP_STRUCT__entry(
+		__string(	wname,	widget->name		)
+		__string(	pname,	path->name ? path->name : DAPM_DIRECT)
+		__string(	psname,	path->source->name	)
+		__field(	int,	path_source		)
+		__field(	int,	path_connect		)
+	),
+
+	TP_fast_assign(
+		__assign_str(wname, widget->name);
+		__assign_str(pname, path->name ? path->name : DAPM_DIRECT);
+		__assign_str(psname, path->source->name);
+		__entry->path_connect = path->connect;
+		__entry->path_source = (long)path->source;
+	),
+
+	TP_printk("%c%s <- %s <- %s\n",
+		(int) __entry->path_source &&
+		(int) __entry->path_connect ? '*' : ' ',
+		__get_str(wname), __get_str(pname), __get_str(psname))
+);
+
+TRACE_EVENT(snd_soc_dapm_connected,
+
+	TP_PROTO(int paths, int stream),
+
+	TP_ARGS(paths, stream),
+
+	TP_STRUCT__entry(
+		__field(	int,	paths		)
+		__field(	int,	stream		)
+	),
+
+	TP_fast_assign(
+		__entry->paths = paths;
+		__entry->stream = stream;
+	),
+
+	TP_printk("%s: found %d paths\n",
+		__entry->stream ? "capture" : "playback", __entry->paths)
+);
+
 TRACE_EVENT(snd_soc_jack_irq,
 
 	TP_PROTO(const char *name),
diff --git a/sound/atmel/ac97c.c b/sound/atmel/ac97c.c
index 115313ef54d6..f5ded640b395 100644
--- a/sound/atmel/ac97c.c
+++ b/sound/atmel/ac97c.c
@@ -991,6 +991,8 @@ static int __devinit atmel_ac97c_probe(struct platform_device *pdev)
 			gpio_direction_output(pdata->reset_pin, 1);
 			chip->reset_pin = pdata->reset_pin;
 		}
+	} else {
+		chip->reset_pin = -EINVAL;
 	}
 
 	snd_card_set_dev(card, &pdev->dev);
diff --git a/sound/core/jack.c b/sound/core/jack.c
index 471e1e3b0a99..a06b1651fcba 100644
--- a/sound/core/jack.c
+++ b/sound/core/jack.c
@@ -155,7 +155,7 @@ EXPORT_SYMBOL(snd_jack_new);
  * @jack:   The jack to configure
  * @parent: The device to set as parent for the jack.
  *
- * Set the parent for the jack input device in the device tree.  This
+ * Set the parent for the jack devices in the device tree.  This
  * function is only valid prior to registration of the jack.  If no
  * parent is configured then the parent device will be the sound card.
  */
@@ -179,6 +179,9 @@ EXPORT_SYMBOL(snd_jack_set_parent);
  * mapping is provided but keys are enabled in the jack type then
  * BTN_n numeric buttons will be reported.
  *
+ * If jacks are not reporting via the input API this call will have no
+ * effect.
+ *
  * Note that this is intended to be use by simple devices with small
  * numbers of keys that can be reported.  It is also possible to
  * access the input device directly - devices with complex input
diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c
index 4d18941178e6..faedb1481b24 100644
--- a/sound/core/pcm_lib.c
+++ b/sound/core/pcm_lib.c
@@ -1894,6 +1894,7 @@ static snd_pcm_sframes_t snd_pcm_lib_write1(struct snd_pcm_substream *substream,
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	snd_pcm_uframes_t xfer = 0;
 	snd_pcm_uframes_t offset = 0;
+	snd_pcm_uframes_t avail;
 	int err = 0;
 
 	if (size == 0)
@@ -1917,13 +1918,12 @@ static snd_pcm_sframes_t snd_pcm_lib_write1(struct snd_pcm_substream *substream,
 	}
 
 	runtime->twake = runtime->control->avail_min ? : 1;
+	if (runtime->status->state == SNDRV_PCM_STATE_RUNNING)
+		snd_pcm_update_hw_ptr(substream);
+	avail = snd_pcm_playback_avail(runtime);
 	while (size > 0) {
 		snd_pcm_uframes_t frames, appl_ptr, appl_ofs;
-		snd_pcm_uframes_t avail;
 		snd_pcm_uframes_t cont;
-		if (runtime->status->state == SNDRV_PCM_STATE_RUNNING)
-			snd_pcm_update_hw_ptr(substream);
-		avail = snd_pcm_playback_avail(runtime);
 		if (!avail) {
 			if (nonblock) {
 				err = -EAGAIN;
@@ -1971,6 +1971,7 @@ static snd_pcm_sframes_t snd_pcm_lib_write1(struct snd_pcm_substream *substream,
 		offset += frames;
 		size -= frames;
 		xfer += frames;
+		avail -= frames;
 		if (runtime->status->state == SNDRV_PCM_STATE_PREPARED &&
 		    snd_pcm_playback_hw_avail(runtime) >= (snd_pcm_sframes_t)runtime->start_threshold) {
 			err = snd_pcm_start(substream);
@@ -2111,6 +2112,7 @@ static snd_pcm_sframes_t snd_pcm_lib_read1(struct snd_pcm_substream *substream,
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	snd_pcm_uframes_t xfer = 0;
 	snd_pcm_uframes_t offset = 0;
+	snd_pcm_uframes_t avail;
 	int err = 0;
 
 	if (size == 0)
@@ -2141,13 +2143,12 @@ static snd_pcm_sframes_t snd_pcm_lib_read1(struct snd_pcm_substream *substream,
 	}
 
 	runtime->twake = runtime->control->avail_min ? : 1;
+	if (runtime->status->state == SNDRV_PCM_STATE_RUNNING)
+		snd_pcm_update_hw_ptr(substream);
+	avail = snd_pcm_capture_avail(runtime);
 	while (size > 0) {
 		snd_pcm_uframes_t frames, appl_ptr, appl_ofs;
-		snd_pcm_uframes_t avail;
 		snd_pcm_uframes_t cont;
-		if (runtime->status->state == SNDRV_PCM_STATE_RUNNING)
-			snd_pcm_update_hw_ptr(substream);
-		avail = snd_pcm_capture_avail(runtime);
 		if (!avail) {
 			if (runtime->status->state ==
 			    SNDRV_PCM_STATE_DRAINING) {
@@ -2202,6 +2203,7 @@ static snd_pcm_sframes_t snd_pcm_lib_read1(struct snd_pcm_substream *substream,
 		offset += frames;
 		size -= frames;
 		xfer += frames;
+		avail -= frames;
 	}
  _end_unlock:
 	runtime->twake = 0;
diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c
index 3fe99e644eb8..53b5ada8f7c3 100644
--- a/sound/core/pcm_native.c
+++ b/sound/core/pcm_native.c
@@ -1360,7 +1360,14 @@ static int snd_pcm_prepare(struct snd_pcm_substream *substream,
 
 static int snd_pcm_pre_drain_init(struct snd_pcm_substream *substream, int state)
 {
-	substream->runtime->trigger_master = substream;
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	switch (runtime->status->state) {
+	case SNDRV_PCM_STATE_OPEN:
+	case SNDRV_PCM_STATE_DISCONNECTED:
+	case SNDRV_PCM_STATE_SUSPENDED:
+		return -EBADFD;
+	}
+	runtime->trigger_master = substream;
 	return 0;
 }
 
@@ -1379,6 +1386,9 @@ static int snd_pcm_do_drain_init(struct snd_pcm_substream *substream, int state)
 		case SNDRV_PCM_STATE_RUNNING:
 			runtime->status->state = SNDRV_PCM_STATE_DRAINING;
 			break;
+		case SNDRV_PCM_STATE_XRUN:
+			runtime->status->state = SNDRV_PCM_STATE_SETUP;
+			break;
 		default:
 			break;
 		}
diff --git a/sound/core/sound_oss.c b/sound/core/sound_oss.c
index c70092043061..e9528333e36d 100644
--- a/sound/core/sound_oss.c
+++ b/sound/core/sound_oss.c
@@ -35,7 +35,7 @@
 #include <linux/sound.h>
 #include <linux/mutex.h>
 
-#define SNDRV_OSS_MINORS 128
+#define SNDRV_OSS_MINORS 256
 
 static struct snd_minor *snd_oss_minors[SNDRV_OSS_MINORS];
 static DEFINE_MUTEX(sound_oss_mutex);
@@ -111,7 +111,7 @@ int snd_register_oss_device(int type, struct snd_card *card, int dev,
 	int register1 = -1, register2 = -1;
 	struct device *carddev = snd_card_get_device_link(card);
 
-	if (card && card->number >= 8)
+	if (card && card->number >= SNDRV_MINOR_OSS_DEVICES)
 		return 0; /* ignore silently */
 	if (minor < 0)
 		return minor;
@@ -170,7 +170,7 @@ int snd_unregister_oss_device(int type, struct snd_card *card, int dev)
 	int track2 = -1;
 	struct snd_minor *mptr;
 
-	if (card && card->number >= 8)
+	if (card && card->number >= SNDRV_MINOR_OSS_DEVICES)
 		return 0;
 	if (minor < 0)
 		return minor;
diff --git a/sound/drivers/aloop.c b/sound/drivers/aloop.c
index ad079b63b8ba..8b5c36f4d303 100644
--- a/sound/drivers/aloop.c
+++ b/sound/drivers/aloop.c
@@ -117,6 +117,7 @@ struct loopback_pcm {
 	/* timer stuff */
 	unsigned int irq_pos;		/* fractional IRQ position */
 	unsigned int period_size_frac;
+	unsigned int last_drift;
 	unsigned long last_jiffies;
 	struct timer_list timer;
 };
@@ -264,6 +265,7 @@ static int loopback_trigger(struct snd_pcm_substream *substream, int cmd)
 			return err;
 		dpcm->last_jiffies = jiffies;
 		dpcm->pcm_rate_shift = 0;
+		dpcm->last_drift = 0;
 		spin_lock(&cable->lock);	
 		cable->running |= stream;
 		cable->pause &= ~stream;
@@ -444,34 +446,30 @@ static void copy_play_buf(struct loopback_pcm *play,
 	}
 }
 
-#define BYTEPOS_UPDATE_POSONLY	0
-#define BYTEPOS_UPDATE_CLEAR	1
-#define BYTEPOS_UPDATE_COPY	2
-
-static void loopback_bytepos_update(struct loopback_pcm *dpcm,
-				    unsigned int delta,
-				    unsigned int cmd)
+static inline unsigned int bytepos_delta(struct loopback_pcm *dpcm,
+					 unsigned int jiffies_delta)
 {
-	unsigned int count;
 	unsigned long last_pos;
+	unsigned int delta;
 
 	last_pos = byte_pos(dpcm, dpcm->irq_pos);
-	dpcm->irq_pos += delta * dpcm->pcm_bps;
-	count = byte_pos(dpcm, dpcm->irq_pos) - last_pos;
-	if (!count)
-		return;
-	if (cmd == BYTEPOS_UPDATE_CLEAR)
-		clear_capture_buf(dpcm, count);
-	else if (cmd == BYTEPOS_UPDATE_COPY)
-		copy_play_buf(dpcm->cable->streams[SNDRV_PCM_STREAM_PLAYBACK],
-			      dpcm->cable->streams[SNDRV_PCM_STREAM_CAPTURE],
-			      count);
-	dpcm->buf_pos += count;
-	dpcm->buf_pos %= dpcm->pcm_buffer_size;
+	dpcm->irq_pos += jiffies_delta * dpcm->pcm_bps;
+	delta = byte_pos(dpcm, dpcm->irq_pos) - last_pos;
+	if (delta >= dpcm->last_drift)
+		delta -= dpcm->last_drift;
+	dpcm->last_drift = 0;
 	if (dpcm->irq_pos >= dpcm->period_size_frac) {
 		dpcm->irq_pos %= dpcm->period_size_frac;
 		dpcm->period_update_pending = 1;
 	}
+	return delta;
+}
+
+static inline void bytepos_finish(struct loopback_pcm *dpcm,
+				  unsigned int delta)
+{
+	dpcm->buf_pos += delta;
+	dpcm->buf_pos %= dpcm->pcm_buffer_size;
 }
 
 static unsigned int loopback_pos_update(struct loopback_cable *cable)
@@ -481,7 +479,7 @@ static unsigned int loopback_pos_update(struct loopback_cable *cable)
 	struct loopback_pcm *dpcm_capt =
 			cable->streams[SNDRV_PCM_STREAM_CAPTURE];
 	unsigned long delta_play = 0, delta_capt = 0;
-	unsigned int running;
+	unsigned int running, count1, count2;
 	unsigned long flags;
 
 	spin_lock_irqsave(&cable->lock, flags);
@@ -500,12 +498,13 @@ static unsigned int loopback_pos_update(struct loopback_cable *cable)
 		goto unlock;
 		
 	if (delta_play > delta_capt) {
-		loopback_bytepos_update(dpcm_play, delta_play - delta_capt,
-					BYTEPOS_UPDATE_POSONLY);
+		count1 = bytepos_delta(dpcm_play, delta_play - delta_capt);
+		bytepos_finish(dpcm_play, count1);
 		delta_play = delta_capt;
 	} else if (delta_play < delta_capt) {
-		loopback_bytepos_update(dpcm_capt, delta_capt - delta_play,
-					BYTEPOS_UPDATE_CLEAR);
+		count1 = bytepos_delta(dpcm_capt, delta_capt - delta_play);
+		clear_capture_buf(dpcm_capt, count1);
+		bytepos_finish(dpcm_capt, count1);
 		delta_capt = delta_play;
 	}
 
@@ -513,8 +512,17 @@ static unsigned int loopback_pos_update(struct loopback_cable *cable)
 		goto unlock;
 
 	/* note delta_capt == delta_play at this moment */
-	loopback_bytepos_update(dpcm_capt, delta_capt, BYTEPOS_UPDATE_COPY);
-	loopback_bytepos_update(dpcm_play, delta_play, BYTEPOS_UPDATE_POSONLY);
+	count1 = bytepos_delta(dpcm_play, delta_play);
+	count2 = bytepos_delta(dpcm_capt, delta_capt);
+	if (count1 < count2) {
+		dpcm_capt->last_drift = count2 - count1;
+		count1 = count2;
+	} else if (count1 > count2) {
+		dpcm_play->last_drift = count1 - count2;
+	}
+	copy_play_buf(dpcm_play, dpcm_capt, count1);
+	bytepos_finish(dpcm_play, count1);
+	bytepos_finish(dpcm_capt, count1);
  unlock:
 	spin_unlock_irqrestore(&cable->lock, flags);
 	return running;
diff --git a/sound/firewire/amdtp.c b/sound/firewire/amdtp.c
index 87657dd7714c..ea995af6d049 100644
--- a/sound/firewire/amdtp.c
+++ b/sound/firewire/amdtp.c
@@ -31,6 +31,8 @@
 #define INTERRUPT_INTERVAL	16
 #define QUEUE_LENGTH		48
 
+static void pcm_period_tasklet(unsigned long data);
+
 /**
  * amdtp_out_stream_init - initialize an AMDTP output stream structure
  * @s: the AMDTP output stream to initialize
@@ -47,6 +49,7 @@ int amdtp_out_stream_init(struct amdtp_out_stream *s, struct fw_unit *unit,
 	s->flags = flags;
 	s->context = ERR_PTR(-1);
 	mutex_init(&s->mutex);
+	tasklet_init(&s->period_tasklet, pcm_period_tasklet, (unsigned long)s);
 	s->packet_index = 0;
 
 	return 0;
@@ -164,6 +167,21 @@ void amdtp_out_stream_set_pcm_format(struct amdtp_out_stream *s,
 }
 EXPORT_SYMBOL(amdtp_out_stream_set_pcm_format);
 
+/**
+ * amdtp_out_stream_pcm_prepare - prepare PCM device for running
+ * @s: the AMDTP output stream
+ *
+ * This function should be called from the PCM device's .prepare callback.
+ */
+void amdtp_out_stream_pcm_prepare(struct amdtp_out_stream *s)
+{
+	tasklet_kill(&s->period_tasklet);
+	s->pcm_buffer_pointer = 0;
+	s->pcm_period_pointer = 0;
+	s->pointer_flush = true;
+}
+EXPORT_SYMBOL(amdtp_out_stream_pcm_prepare);
+
 static unsigned int calculate_data_blocks(struct amdtp_out_stream *s)
 {
 	unsigned int phase, data_blocks;
@@ -376,11 +394,21 @@ static void queue_out_packet(struct amdtp_out_stream *s, unsigned int cycle)
 		s->pcm_period_pointer += data_blocks;
 		if (s->pcm_period_pointer >= pcm->runtime->period_size) {
 			s->pcm_period_pointer -= pcm->runtime->period_size;
-			snd_pcm_period_elapsed(pcm);
+			s->pointer_flush = false;
+			tasklet_hi_schedule(&s->period_tasklet);
 		}
 	}
 }
 
+static void pcm_period_tasklet(unsigned long data)
+{
+	struct amdtp_out_stream *s = (void *)data;
+	struct snd_pcm_substream *pcm = ACCESS_ONCE(s->pcm);
+
+	if (pcm)
+		snd_pcm_period_elapsed(pcm);
+}
+
 static void out_packet_callback(struct fw_iso_context *context, u32 cycle,
 				size_t header_length, void *header, void *data)
 {
@@ -505,6 +533,24 @@ err_unlock:
 }
 EXPORT_SYMBOL(amdtp_out_stream_start);
 
+/**
+ * amdtp_out_stream_pcm_pointer - get the PCM buffer position
+ * @s: the AMDTP output stream that transports the PCM data
+ *
+ * Returns the current buffer position, in frames.
+ */
+unsigned long amdtp_out_stream_pcm_pointer(struct amdtp_out_stream *s)
+{
+	/* this optimization is allowed to be racy */
+	if (s->pointer_flush)
+		fw_iso_context_flush_completions(s->context);
+	else
+		s->pointer_flush = true;
+
+	return ACCESS_ONCE(s->pcm_buffer_pointer);
+}
+EXPORT_SYMBOL(amdtp_out_stream_pcm_pointer);
+
 /**
  * amdtp_out_stream_update - update the stream after a bus reset
  * @s: the AMDTP output stream
@@ -532,6 +578,7 @@ void amdtp_out_stream_stop(struct amdtp_out_stream *s)
 		return;
 	}
 
+	tasklet_kill(&s->period_tasklet);
 	fw_iso_context_stop(s->context);
 	fw_iso_context_destroy(s->context);
 	s->context = ERR_PTR(-1);
diff --git a/sound/firewire/amdtp.h b/sound/firewire/amdtp.h
index 537a9cb83581..b680c5ef01d6 100644
--- a/sound/firewire/amdtp.h
+++ b/sound/firewire/amdtp.h
@@ -1,6 +1,7 @@
 #ifndef SOUND_FIREWIRE_AMDTP_H_INCLUDED
 #define SOUND_FIREWIRE_AMDTP_H_INCLUDED
 
+#include <linux/interrupt.h>
 #include <linux/mutex.h>
 #include <linux/spinlock.h>
 #include "packets-buffer.h"
@@ -55,6 +56,7 @@ struct amdtp_out_stream {
 	struct iso_packets_buffer buffer;
 
 	struct snd_pcm_substream *pcm;
+	struct tasklet_struct period_tasklet;
 
 	int packet_index;
 	unsigned int data_block_counter;
@@ -66,6 +68,7 @@ struct amdtp_out_stream {
 
 	unsigned int pcm_buffer_pointer;
 	unsigned int pcm_period_pointer;
+	bool pointer_flush;
 };
 
 int amdtp_out_stream_init(struct amdtp_out_stream *s, struct fw_unit *unit,
@@ -81,6 +84,8 @@ void amdtp_out_stream_stop(struct amdtp_out_stream *s);
 
 void amdtp_out_stream_set_pcm_format(struct amdtp_out_stream *s,
 				     snd_pcm_format_t format);
+void amdtp_out_stream_pcm_prepare(struct amdtp_out_stream *s);
+unsigned long amdtp_out_stream_pcm_pointer(struct amdtp_out_stream *s);
 void amdtp_out_stream_pcm_abort(struct amdtp_out_stream *s);
 
 /**
@@ -122,18 +127,6 @@ static inline bool amdtp_out_streaming_error(struct amdtp_out_stream *s)
 	return s->packet_index < 0;
 }
 
-/**
- * amdtp_out_stream_pcm_prepare - prepare PCM device for running
- * @s: the AMDTP output stream
- *
- * This function should be called from the PCM device's .prepare callback.
- */
-static inline void amdtp_out_stream_pcm_prepare(struct amdtp_out_stream *s)
-{
-	s->pcm_buffer_pointer = 0;
-	s->pcm_period_pointer = 0;
-}
-
 /**
  * amdtp_out_stream_pcm_trigger - start/stop playback from a PCM device
  * @s: the AMDTP output stream
@@ -149,18 +142,6 @@ static inline void amdtp_out_stream_pcm_trigger(struct amdtp_out_stream *s,
 	ACCESS_ONCE(s->pcm) = pcm;
 }
 
-/**
- * amdtp_out_stream_pcm_pointer - get the PCM buffer position
- * @s: the AMDTP output stream that transports the PCM data
- *
- * Returns the current buffer position, in frames.
- */
-static inline unsigned long
-amdtp_out_stream_pcm_pointer(struct amdtp_out_stream *s)
-{
-	return ACCESS_ONCE(s->pcm_buffer_pointer);
-}
-
 static inline bool cip_sfc_is_base_44100(enum cip_sfc sfc)
 {
 	return sfc & 1;
diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig
index 5ca0939e4223..ff3af6e77d61 100644
--- a/sound/pci/Kconfig
+++ b/sound/pci/Kconfig
@@ -228,7 +228,7 @@ config SND_OXYGEN
 	  Say Y here to include support for sound cards based on the
 	  C-Media CMI8788 (Oxygen HD Audio) chip:
 	   * Asound A-8788
-	   * Asus Xonar DG
+	   * Asus Xonar DG/DGX
 	   * AuzenTech X-Meridian
 	   * AuzenTech X-Meridian 2G
 	   * Bgears b-Enspirer
diff --git a/sound/pci/ad1889.c b/sound/pci/ad1889.c
index 9d91d61902b4..e672ff4df2da 100644
--- a/sound/pci/ad1889.c
+++ b/sound/pci/ad1889.c
@@ -1062,17 +1062,4 @@ static struct pci_driver ad1889_pci_driver = {
 	.remove = __devexit_p(snd_ad1889_remove),
 };
 
-static int __init
-alsa_ad1889_init(void)
-{
-	return pci_register_driver(&ad1889_pci_driver);
-}
-
-static void __exit
-alsa_ad1889_fini(void)
-{
-	pci_unregister_driver(&ad1889_pci_driver);
-}
-
-module_init(alsa_ad1889_init);
-module_exit(alsa_ad1889_fini);
+module_pci_driver(ad1889_pci_driver);
diff --git a/sound/pci/ali5451/ali5451.c b/sound/pci/ali5451/ali5451.c
index bdd6164e9c7e..9dfc27bf6cc6 100644
--- a/sound/pci/ali5451/ali5451.c
+++ b/sound/pci/ali5451/ali5451.c
@@ -2294,7 +2294,7 @@ static void __devexit snd_ali_remove(struct pci_dev *pci)
 	pci_set_drvdata(pci, NULL);
 }
 
-static struct pci_driver driver = {
+static struct pci_driver ali5451_driver = {
 	.name = KBUILD_MODNAME,
 	.id_table = snd_ali_ids,
 	.probe = snd_ali_probe,
@@ -2305,15 +2305,4 @@ static struct pci_driver driver = {
 #endif
 };                                
 
-static int __init alsa_card_ali_init(void)
-{
-	return pci_register_driver(&driver);
-}
-
-static void __exit alsa_card_ali_exit(void)
-{
-	pci_unregister_driver(&driver);
-}
-
-module_init(alsa_card_ali_init)
-module_exit(alsa_card_ali_exit)
+module_pci_driver(ali5451_driver);
diff --git a/sound/pci/als300.c b/sound/pci/als300.c
index 8196e229b2df..59d65388faf5 100644
--- a/sound/pci/als300.c
+++ b/sound/pci/als300.c
@@ -852,7 +852,7 @@ static int __devinit snd_als300_probe(struct pci_dev *pci,
 	return 0;
 }
 
-static struct pci_driver driver = {
+static struct pci_driver als300_driver = {
 	.name = KBUILD_MODNAME,
 	.id_table = snd_als300_ids,
 	.probe = snd_als300_probe,
@@ -863,15 +863,4 @@ static struct pci_driver driver = {
 #endif
 };
 
-static int __init alsa_card_als300_init(void)
-{
-	return pci_register_driver(&driver);
-}
-
-static void __exit alsa_card_als300_exit(void)
-{
-	pci_unregister_driver(&driver);
-}
-
-module_init(alsa_card_als300_init)
-module_exit(alsa_card_als300_exit)
+module_pci_driver(als300_driver);
diff --git a/sound/pci/als4000.c b/sound/pci/als4000.c
index 3269b8011ea9..7d7f2598c748 100644
--- a/sound/pci/als4000.c
+++ b/sound/pci/als4000.c
@@ -1036,7 +1036,7 @@ static int snd_als4000_resume(struct pci_dev *pci)
 #endif /* CONFIG_PM */
 
 
-static struct pci_driver driver = {
+static struct pci_driver als4000_driver = {
 	.name = KBUILD_MODNAME,
 	.id_table = snd_als4000_ids,
 	.probe = snd_card_als4000_probe,
@@ -1047,15 +1047,4 @@ static struct pci_driver driver = {
 #endif
 };
 
-static int __init alsa_card_als4000_init(void)
-{
-	return pci_register_driver(&driver);
-}
-
-static void __exit alsa_card_als4000_exit(void)
-{
-	pci_unregister_driver(&driver);
-}
-
-module_init(alsa_card_als4000_init)
-module_exit(alsa_card_als4000_exit)
+module_pci_driver(als4000_driver);
diff --git a/sound/pci/atiixp.c b/sound/pci/atiixp.c
index 590682f115ef..156a94f8a123 100644
--- a/sound/pci/atiixp.c
+++ b/sound/pci/atiixp.c
@@ -1700,7 +1700,7 @@ static void __devexit snd_atiixp_remove(struct pci_dev *pci)
 	pci_set_drvdata(pci, NULL);
 }
 
-static struct pci_driver driver = {
+static struct pci_driver atiixp_driver = {
 	.name = KBUILD_MODNAME,
 	.id_table = snd_atiixp_ids,
 	.probe = snd_atiixp_probe,
@@ -1711,16 +1711,4 @@ static struct pci_driver driver = {
 #endif
 };
 
-
-static int __init alsa_card_atiixp_init(void)
-{
-	return pci_register_driver(&driver);
-}
-
-static void __exit alsa_card_atiixp_exit(void)
-{
-	pci_unregister_driver(&driver);
-}
-
-module_init(alsa_card_atiixp_init)
-module_exit(alsa_card_atiixp_exit)
+module_pci_driver(atiixp_driver);
diff --git a/sound/pci/atiixp_modem.c b/sound/pci/atiixp_modem.c
index 524d35f31232..30a4fd96ce73 100644
--- a/sound/pci/atiixp_modem.c
+++ b/sound/pci/atiixp_modem.c
@@ -1331,7 +1331,7 @@ static void __devexit snd_atiixp_remove(struct pci_dev *pci)
 	pci_set_drvdata(pci, NULL);
 }
 
-static struct pci_driver driver = {
+static struct pci_driver atiixp_modem_driver = {
 	.name = KBUILD_MODNAME,
 	.id_table = snd_atiixp_ids,
 	.probe = snd_atiixp_probe,
@@ -1342,16 +1342,4 @@ static struct pci_driver driver = {
 #endif
 };
 
-
-static int __init alsa_card_atiixp_init(void)
-{
-	return pci_register_driver(&driver);
-}
-
-static void __exit alsa_card_atiixp_exit(void)
-{
-	pci_unregister_driver(&driver);
-}
-
-module_init(alsa_card_atiixp_init)
-module_exit(alsa_card_atiixp_exit)
+module_pci_driver(atiixp_modem_driver);
diff --git a/sound/pci/au88x0/au88x0.c b/sound/pci/au88x0/au88x0.c
index f13ad536b2d5..ffc376f9f4e4 100644
--- a/sound/pci/au88x0/au88x0.c
+++ b/sound/pci/au88x0/au88x0.c
@@ -375,24 +375,11 @@ static void __devexit snd_vortex_remove(struct pci_dev *pci)
 }
 
 // pci_driver definition
-static struct pci_driver driver = {
+static struct pci_driver vortex_driver = {
 	.name = KBUILD_MODNAME,
 	.id_table = snd_vortex_ids,
 	.probe = snd_vortex_probe,
 	.remove = __devexit_p(snd_vortex_remove),
 };
 
-// initialization of the module
-static int __init alsa_card_vortex_init(void)
-{
-	return pci_register_driver(&driver);
-}
-
-// clean up the module
-static void __exit alsa_card_vortex_exit(void)
-{
-	pci_unregister_driver(&driver);
-}
-
-module_init(alsa_card_vortex_init)
-module_exit(alsa_card_vortex_exit)
+module_pci_driver(vortex_driver);
diff --git a/sound/pci/aw2/aw2-alsa.c b/sound/pci/aw2/aw2-alsa.c
index 1c5231931462..0f804741825f 100644
--- a/sound/pci/aw2/aw2-alsa.c
+++ b/sound/pci/aw2/aw2-alsa.c
@@ -112,8 +112,6 @@ struct aw2 {
 /*********************************
  * FUNCTION DECLARATIONS
  ********************************/
-static int __init alsa_card_aw2_init(void);
-static void __exit alsa_card_aw2_exit(void);
 static int snd_aw2_dev_free(struct snd_device *device);
 static int __devinit snd_aw2_create(struct snd_card *card,
 				    struct pci_dev *pci, struct aw2 **rchip);
@@ -171,13 +169,15 @@ static DEFINE_PCI_DEVICE_TABLE(snd_aw2_ids) = {
 MODULE_DEVICE_TABLE(pci, snd_aw2_ids);
 
 /* pci_driver definition */
-static struct pci_driver driver = {
+static struct pci_driver aw2_driver = {
 	.name = KBUILD_MODNAME,
 	.id_table = snd_aw2_ids,
 	.probe = snd_aw2_probe,
 	.remove = __devexit_p(snd_aw2_remove),
 };
 
+module_pci_driver(aw2_driver);
+
 /* operators for playback PCM alsa interface */
 static struct snd_pcm_ops snd_aw2_playback_ops = {
 	.open = snd_aw2_pcm_playback_open,
@@ -217,23 +217,6 @@ static struct snd_kcontrol_new aw2_control __devinitdata = {
  * FUNCTION IMPLEMENTATIONS
  ********************************/
 
-/* initialization of the module */
-static int __init alsa_card_aw2_init(void)
-{
-	snd_printdd(KERN_DEBUG "aw2: Load aw2 module\n");
-	return pci_register_driver(&driver);
-}
-
-/* clean up the module */
-static void __exit alsa_card_aw2_exit(void)
-{
-	snd_printdd(KERN_DEBUG "aw2: Unload aw2 module\n");
-	pci_unregister_driver(&driver);
-}
-
-module_init(alsa_card_aw2_init);
-module_exit(alsa_card_aw2_exit);
-
 /* component-destructor */
 static int snd_aw2_dev_free(struct snd_device *device)
 {
diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c
index 496f14c1a731..f0b4d7493af5 100644
--- a/sound/pci/azt3328.c
+++ b/sound/pci/azt3328.c
@@ -2862,7 +2862,7 @@ snd_azf3328_resume(struct pci_dev *pci)
 #endif /* CONFIG_PM */
 
 
-static struct pci_driver driver = {
+static struct pci_driver azf3328_driver = {
 	.name = KBUILD_MODNAME,
 	.id_table = snd_azf3328_ids,
 	.probe = snd_azf3328_probe,
@@ -2873,23 +2873,4 @@ static struct pci_driver driver = {
 #endif
 };
 
-static int __init
-alsa_card_azf3328_init(void)
-{
-	int err;
-	snd_azf3328_dbgcallenter();
-	err = pci_register_driver(&driver);
-	snd_azf3328_dbgcallleave();
-	return err;
-}
-
-static void __exit
-alsa_card_azf3328_exit(void)
-{
-	snd_azf3328_dbgcallenter();
-	pci_unregister_driver(&driver);
-	snd_azf3328_dbgcallleave();
-}
-
-module_init(alsa_card_azf3328_init)
-module_exit(alsa_card_azf3328_exit)
+module_pci_driver(azf3328_driver);
diff --git a/sound/pci/bt87x.c b/sound/pci/bt87x.c
index 62d6163fc9d9..b6a95eeca095 100644
--- a/sound/pci/bt87x.c
+++ b/sound/pci/bt87x.c
@@ -836,8 +836,6 @@ static struct {
 	{0x7063, 0x2000}, /* pcHDTV HD-2000 TV */
 };
 
-static struct pci_driver driver;
-
 /* return the id of the card, or a negative value if it's blacklisted */
 static int __devinit snd_bt87x_detect_card(struct pci_dev *pci)
 {
@@ -964,24 +962,11 @@ static DEFINE_PCI_DEVICE_TABLE(snd_bt87x_default_ids) = {
 	{ }
 };
 
-static struct pci_driver driver = {
+static struct pci_driver bt87x_driver = {
 	.name = KBUILD_MODNAME,
 	.id_table = snd_bt87x_ids,
 	.probe = snd_bt87x_probe,
 	.remove = __devexit_p(snd_bt87x_remove),
 };
 
-static int __init alsa_card_bt87x_init(void)
-{
-	if (load_all)
-		driver.id_table = snd_bt87x_default_ids;
-	return pci_register_driver(&driver);
-}
-
-static void __exit alsa_card_bt87x_exit(void)
-{
-	pci_unregister_driver(&driver);
-}
-
-module_init(alsa_card_bt87x_init)
-module_exit(alsa_card_bt87x_exit)
+module_pci_driver(bt87x_driver);
diff --git a/sound/pci/ca0106/ca0106_main.c b/sound/pci/ca0106/ca0106_main.c
index 08d6ebfe5a61..e76d68a7081f 100644
--- a/sound/pci/ca0106/ca0106_main.c
+++ b/sound/pci/ca0106/ca0106_main.c
@@ -1932,7 +1932,7 @@ static DEFINE_PCI_DEVICE_TABLE(snd_ca0106_ids) = {
 MODULE_DEVICE_TABLE(pci, snd_ca0106_ids);
 
 // pci_driver definition
-static struct pci_driver driver = {
+static struct pci_driver ca0106_driver = {
 	.name = KBUILD_MODNAME,
 	.id_table = snd_ca0106_ids,
 	.probe = snd_ca0106_probe,
@@ -1943,17 +1943,4 @@ static struct pci_driver driver = {
 #endif
 };
 
-// initialization of the module
-static int __init alsa_card_ca0106_init(void)
-{
-	return pci_register_driver(&driver);
-}
-
-// clean up the module
-static void __exit alsa_card_ca0106_exit(void)
-{
-	pci_unregister_driver(&driver);
-}
-
-module_init(alsa_card_ca0106_init)
-module_exit(alsa_card_ca0106_exit)
+module_pci_driver(ca0106_driver);
diff --git a/sound/pci/cmipci.c b/sound/pci/cmipci.c
index 19b06269adc2..3815bd4c6779 100644
--- a/sound/pci/cmipci.c
+++ b/sound/pci/cmipci.c
@@ -3398,7 +3398,7 @@ static int snd_cmipci_resume(struct pci_dev *pci)
 }
 #endif /* CONFIG_PM */
 
-static struct pci_driver driver = {
+static struct pci_driver cmipci_driver = {
 	.name = KBUILD_MODNAME,
 	.id_table = snd_cmipci_ids,
 	.probe = snd_cmipci_probe,
@@ -3409,15 +3409,4 @@ static struct pci_driver driver = {
 #endif
 };
 	
-static int __init alsa_card_cmipci_init(void)
-{
-	return pci_register_driver(&driver);
-}
-
-static void __exit alsa_card_cmipci_exit(void)
-{
-	pci_unregister_driver(&driver);
-}
-
-module_init(alsa_card_cmipci_init)
-module_exit(alsa_card_cmipci_exit)
+module_pci_driver(cmipci_driver);
diff --git a/sound/pci/cs4281.c b/sound/pci/cs4281.c
index a9f368f60df6..33506ee569bd 100644
--- a/sound/pci/cs4281.c
+++ b/sound/pci/cs4281.c
@@ -2084,7 +2084,7 @@ static int cs4281_resume(struct pci_dev *pci)
 }
 #endif /* CONFIG_PM */
 
-static struct pci_driver driver = {
+static struct pci_driver cs4281_driver = {
 	.name = KBUILD_MODNAME,
 	.id_table = snd_cs4281_ids,
 	.probe = snd_cs4281_probe,
@@ -2095,15 +2095,4 @@ static struct pci_driver driver = {
 #endif
 };
 	
-static int __init alsa_card_cs4281_init(void)
-{
-	return pci_register_driver(&driver);
-}
-
-static void __exit alsa_card_cs4281_exit(void)
-{
-	pci_unregister_driver(&driver);
-}
-
-module_init(alsa_card_cs4281_init)
-module_exit(alsa_card_cs4281_exit)
+module_pci_driver(cs4281_driver);
diff --git a/sound/pci/cs46xx/cs46xx.c b/sound/pci/cs46xx/cs46xx.c
index 819d79d0586d..6cc7404e0e8f 100644
--- a/sound/pci/cs46xx/cs46xx.c
+++ b/sound/pci/cs46xx/cs46xx.c
@@ -161,7 +161,7 @@ static void __devexit snd_card_cs46xx_remove(struct pci_dev *pci)
 	pci_set_drvdata(pci, NULL);
 }
 
-static struct pci_driver driver = {
+static struct pci_driver cs46xx_driver = {
 	.name = KBUILD_MODNAME,
 	.id_table = snd_cs46xx_ids,
 	.probe = snd_card_cs46xx_probe,
@@ -172,15 +172,4 @@ static struct pci_driver driver = {
 #endif
 };
 
-static int __init alsa_card_cs46xx_init(void)
-{
-	return pci_register_driver(&driver);
-}
-
-static void __exit alsa_card_cs46xx_exit(void)
-{
-	pci_unregister_driver(&driver);
-}
-
-module_init(alsa_card_cs46xx_init)
-module_exit(alsa_card_cs46xx_exit)
+module_pci_driver(cs46xx_driver);
diff --git a/sound/pci/cs5530.c b/sound/pci/cs5530.c
index c47cabff2bfa..f1e4229993af 100644
--- a/sound/pci/cs5530.c
+++ b/sound/pci/cs5530.c
@@ -291,23 +291,11 @@ static int __devinit snd_cs5530_probe(struct pci_dev *pci,
 	return 0;
 }
 
-static struct pci_driver driver = {
+static struct pci_driver cs5530_driver = {
 	.name = KBUILD_MODNAME,
 	.id_table = snd_cs5530_ids,
 	.probe = snd_cs5530_probe,
 	.remove = __devexit_p(snd_cs5530_remove),
 };
 
-static int __init alsa_card_cs5530_init(void)
-{
-	return pci_register_driver(&driver);
-}
-
-static void __exit alsa_card_cs5530_exit(void)
-{
-	pci_unregister_driver(&driver);
-}
-
-module_init(alsa_card_cs5530_init)
-module_exit(alsa_card_cs5530_exit)
-
+module_pci_driver(cs5530_driver);
diff --git a/sound/pci/cs5535audio/cs5535audio.c b/sound/pci/cs5535audio/cs5535audio.c
index a2fb2173e980..2c9697cf0a1a 100644
--- a/sound/pci/cs5535audio/cs5535audio.c
+++ b/sound/pci/cs5535audio/cs5535audio.c
@@ -394,7 +394,7 @@ static void __devexit snd_cs5535audio_remove(struct pci_dev *pci)
 	pci_set_drvdata(pci, NULL);
 }
 
-static struct pci_driver driver = {
+static struct pci_driver cs5535audio_driver = {
 	.name = KBUILD_MODNAME,
 	.id_table = snd_cs5535audio_ids,
 	.probe = snd_cs5535audio_probe,
@@ -405,18 +405,7 @@ static struct pci_driver driver = {
 #endif
 };
 
-static int __init alsa_card_cs5535audio_init(void)
-{
-	return pci_register_driver(&driver);
-}
-
-static void __exit alsa_card_cs5535audio_exit(void)
-{
-	pci_unregister_driver(&driver);
-}
-
-module_init(alsa_card_cs5535audio_init)
-module_exit(alsa_card_cs5535audio_exit)
+module_pci_driver(cs5535audio_driver);
 
 MODULE_AUTHOR("Jaya Kumar");
 MODULE_LICENSE("GPL");
diff --git a/sound/pci/ctxfi/xfi.c b/sound/pci/ctxfi/xfi.c
index 15d95d2bacee..75aa2c338410 100644
--- a/sound/pci/ctxfi/xfi.c
+++ b/sound/pci/ctxfi/xfi.c
@@ -154,15 +154,4 @@ static struct pci_driver ct_driver = {
 #endif
 };
 
-static int __init ct_card_init(void)
-{
-	return pci_register_driver(&ct_driver);
-}
-
-static void __exit ct_card_exit(void)
-{
-	pci_unregister_driver(&ct_driver);
-}
-
-module_init(ct_card_init)
-module_exit(ct_card_exit)
+module_pci_driver(ct_driver);
diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c
index 595c11f904bb..0f8eda1dafdb 100644
--- a/sound/pci/echoaudio/echoaudio.c
+++ b/sound/pci/echoaudio/echoaudio.c
@@ -2328,7 +2328,7 @@ static void __devexit snd_echo_remove(struct pci_dev *pci)
 ******************************************************************************/
 
 /* pci_driver definition */
-static struct pci_driver driver = {
+static struct pci_driver echo_driver = {
 	.name = KBUILD_MODNAME,
 	.id_table = snd_echo_ids,
 	.probe = snd_echo_probe,
@@ -2339,22 +2339,4 @@ static struct pci_driver driver = {
 #endif /* CONFIG_PM */
 };
 
-
-
-/* initialization of the module */
-static int __init alsa_card_echo_init(void)
-{
-	return pci_register_driver(&driver);
-}
-
-
-
-/* clean up the module */
-static void __exit alsa_card_echo_exit(void)
-{
-	pci_unregister_driver(&driver);
-}
-
-
-module_init(alsa_card_echo_init)
-module_exit(alsa_card_echo_exit)
+module_pci_driver(echo_driver);
diff --git a/sound/pci/emu10k1/emu10k1.c b/sound/pci/emu10k1/emu10k1.c
index 790c65d980c8..7fdbbe4d9965 100644
--- a/sound/pci/emu10k1/emu10k1.c
+++ b/sound/pci/emu10k1/emu10k1.c
@@ -263,7 +263,7 @@ static int snd_emu10k1_resume(struct pci_dev *pci)
 }
 #endif
 
-static struct pci_driver driver = {
+static struct pci_driver emu10k1_driver = {
 	.name = KBUILD_MODNAME,
 	.id_table = snd_emu10k1_ids,
 	.probe = snd_card_emu10k1_probe,
@@ -274,15 +274,4 @@ static struct pci_driver driver = {
 #endif
 };
 
-static int __init alsa_card_emu10k1_init(void)
-{
-	return pci_register_driver(&driver);
-}
-
-static void __exit alsa_card_emu10k1_exit(void)
-{
-	pci_unregister_driver(&driver);
-}
-
-module_init(alsa_card_emu10k1_init)
-module_exit(alsa_card_emu10k1_exit)
+module_pci_driver(emu10k1_driver);
diff --git a/sound/pci/emu10k1/emu10k1x.c b/sound/pci/emu10k1/emu10k1x.c
index 47a651cb6e84..5c8978b2c4d9 100644
--- a/sound/pci/emu10k1/emu10k1x.c
+++ b/sound/pci/emu10k1/emu10k1x.c
@@ -1612,24 +1612,11 @@ static DEFINE_PCI_DEVICE_TABLE(snd_emu10k1x_ids) = {
 MODULE_DEVICE_TABLE(pci, snd_emu10k1x_ids);
 
 // pci_driver definition
-static struct pci_driver driver = {
+static struct pci_driver emu10k1x_driver = {
 	.name = KBUILD_MODNAME,
 	.id_table = snd_emu10k1x_ids,
 	.probe = snd_emu10k1x_probe,
 	.remove = __devexit_p(snd_emu10k1x_remove),
 };
 
-// initialization of the module
-static int __init alsa_card_emu10k1x_init(void)
-{
-	return pci_register_driver(&driver);
-}
-
-// clean up the module
-static void __exit alsa_card_emu10k1x_exit(void)
-{
-	pci_unregister_driver(&driver);
-}
-
-module_init(alsa_card_emu10k1x_init)
-module_exit(alsa_card_emu10k1x_exit)
+module_pci_driver(emu10k1x_driver);
diff --git a/sound/pci/ens1370.c b/sound/pci/ens1370.c
index 47a245e84190..3821c81d1c99 100644
--- a/sound/pci/ens1370.c
+++ b/sound/pci/ens1370.c
@@ -2488,7 +2488,7 @@ static void __devexit snd_audiopci_remove(struct pci_dev *pci)
 	pci_set_drvdata(pci, NULL);
 }
 
-static struct pci_driver driver = {
+static struct pci_driver ens137x_driver = {
 	.name = KBUILD_MODNAME,
 	.id_table = snd_audiopci_ids,
 	.probe = snd_audiopci_probe,
@@ -2499,15 +2499,4 @@ static struct pci_driver driver = {
 #endif
 };
 	
-static int __init alsa_card_ens137x_init(void)
-{
-	return pci_register_driver(&driver);
-}
-
-static void __exit alsa_card_ens137x_exit(void)
-{
-	pci_unregister_driver(&driver);
-}
-
-module_init(alsa_card_ens137x_init)
-module_exit(alsa_card_ens137x_exit)
+module_pci_driver(ens137x_driver);
diff --git a/sound/pci/es1938.c b/sound/pci/es1938.c
index 53eb76b41108..82c8d8c5c52a 100644
--- a/sound/pci/es1938.c
+++ b/sound/pci/es1938.c
@@ -1882,7 +1882,7 @@ static void __devexit snd_es1938_remove(struct pci_dev *pci)
 	pci_set_drvdata(pci, NULL);
 }
 
-static struct pci_driver driver = {
+static struct pci_driver es1938_driver = {
 	.name = KBUILD_MODNAME,
 	.id_table = snd_es1938_ids,
 	.probe = snd_es1938_probe,
@@ -1893,15 +1893,4 @@ static struct pci_driver driver = {
 #endif
 };
 
-static int __init alsa_card_es1938_init(void)
-{
-	return pci_register_driver(&driver);
-}
-
-static void __exit alsa_card_es1938_exit(void)
-{
-	pci_unregister_driver(&driver);
-}
-
-module_init(alsa_card_es1938_init)
-module_exit(alsa_card_es1938_exit)
+module_pci_driver(es1938_driver);
diff --git a/sound/pci/es1968.c b/sound/pci/es1968.c
index a8faae1c85e4..67f47d891959 100644
--- a/sound/pci/es1968.c
+++ b/sound/pci/es1968.c
@@ -2898,7 +2898,7 @@ static void __devexit snd_es1968_remove(struct pci_dev *pci)
 	pci_set_drvdata(pci, NULL);
 }
 
-static struct pci_driver driver = {
+static struct pci_driver es1968_driver = {
 	.name = KBUILD_MODNAME,
 	.id_table = snd_es1968_ids,
 	.probe = snd_es1968_probe,
@@ -2909,15 +2909,4 @@ static struct pci_driver driver = {
 #endif
 };
 
-static int __init alsa_card_es1968_init(void)
-{
-	return pci_register_driver(&driver);
-}
-
-static void __exit alsa_card_es1968_exit(void)
-{
-	pci_unregister_driver(&driver);
-}
-
-module_init(alsa_card_es1968_init)
-module_exit(alsa_card_es1968_exit)
+module_pci_driver(es1968_driver);
diff --git a/sound/pci/fm801.c b/sound/pci/fm801.c
index a416ea8af3e9..f69662322750 100644
--- a/sound/pci/fm801.c
+++ b/sound/pci/fm801.c
@@ -1416,7 +1416,7 @@ static int snd_fm801_resume(struct pci_dev *pci)
 }
 #endif
 
-static struct pci_driver driver = {
+static struct pci_driver fm801_driver = {
 	.name = KBUILD_MODNAME,
 	.id_table = snd_fm801_ids,
 	.probe = snd_card_fm801_probe,
@@ -1427,15 +1427,4 @@ static struct pci_driver driver = {
 #endif
 };
 
-static int __init alsa_card_fm801_init(void)
-{
-	return pci_register_driver(&driver);
-}
-
-static void __exit alsa_card_fm801_exit(void)
-{
-	pci_unregister_driver(&driver);
-}
-
-module_init(alsa_card_fm801_init)
-module_exit(alsa_card_fm801_exit)
+module_pci_driver(fm801_driver);
diff --git a/sound/pci/hda/Makefile b/sound/pci/hda/Makefile
index ace157cc3d15..bd4149f1aaf4 100644
--- a/sound/pci/hda/Makefile
+++ b/sound/pci/hda/Makefile
@@ -1,6 +1,6 @@
 snd-hda-intel-objs := hda_intel.o
 
-snd-hda-codec-y := hda_codec.o hda_jack.o
+snd-hda-codec-y := hda_codec.o hda_jack.o hda_auto_parser.o
 snd-hda-codec-$(CONFIG_SND_HDA_GENERIC) += hda_generic.o
 snd-hda-codec-$(CONFIG_PROC_FS) += hda_proc.o
 snd-hda-codec-$(CONFIG_SND_HDA_HWDEP) += hda_hwdep.o
diff --git a/sound/pci/hda/hda_auto_parser.c b/sound/pci/hda/hda_auto_parser.c
new file mode 100644
index 000000000000..6e9ef3e25093
--- /dev/null
+++ b/sound/pci/hda/hda_auto_parser.c
@@ -0,0 +1,760 @@
+/*
+ * BIOS auto-parser helper functions for HD-audio
+ *
+ * Copyright (c) 2012 Takashi Iwai <tiwai@suse.de>
+ *
+ * This driver is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ */
+
+#include <linux/slab.h>
+#include <linux/export.h>
+#include <sound/core.h>
+#include "hda_codec.h"
+#include "hda_local.h"
+#include "hda_auto_parser.h"
+
+#define SFX	"hda_codec: "
+
+/*
+ * Helper for automatic pin configuration
+ */
+
+static int is_in_nid_list(hda_nid_t nid, const hda_nid_t *list)
+{
+	for (; *list; list++)
+		if (*list == nid)
+			return 1;
+	return 0;
+}
+
+
+/*
+ * Sort an associated group of pins according to their sequence numbers.
+ */
+static void sort_pins_by_sequence(hda_nid_t *pins, short *sequences,
+				  int num_pins)
+{
+	int i, j;
+	short seq;
+	hda_nid_t nid;
+
+	for (i = 0; i < num_pins; i++) {
+		for (j = i + 1; j < num_pins; j++) {
+			if (sequences[i] > sequences[j]) {
+				seq = sequences[i];
+				sequences[i] = sequences[j];
+				sequences[j] = seq;
+				nid = pins[i];
+				pins[i] = pins[j];
+				pins[j] = nid;
+			}
+		}
+	}
+}
+
+
+/* add the found input-pin to the cfg->inputs[] table */
+static void add_auto_cfg_input_pin(struct auto_pin_cfg *cfg, hda_nid_t nid,
+				   int type)
+{
+	if (cfg->num_inputs < AUTO_CFG_MAX_INS) {
+		cfg->inputs[cfg->num_inputs].pin = nid;
+		cfg->inputs[cfg->num_inputs].type = type;
+		cfg->num_inputs++;
+	}
+}
+
+/* sort inputs in the order of AUTO_PIN_* type */
+static void sort_autocfg_input_pins(struct auto_pin_cfg *cfg)
+{
+	int i, j;
+
+	for (i = 0; i < cfg->num_inputs; i++) {
+		for (j = i + 1; j < cfg->num_inputs; j++) {
+			if (cfg->inputs[i].type > cfg->inputs[j].type) {
+				struct auto_pin_cfg_item tmp;
+				tmp = cfg->inputs[i];
+				cfg->inputs[i] = cfg->inputs[j];
+				cfg->inputs[j] = tmp;
+			}
+		}
+	}
+}
+
+/* Reorder the surround channels
+ * ALSA sequence is front/surr/clfe/side
+ * HDA sequence is:
+ *    4-ch: front/surr  =>  OK as it is
+ *    6-ch: front/clfe/surr
+ *    8-ch: front/clfe/rear/side|fc
+ */
+static void reorder_outputs(unsigned int nums, hda_nid_t *pins)
+{
+	hda_nid_t nid;
+
+	switch (nums) {
+	case 3:
+	case 4:
+		nid = pins[1];
+		pins[1] = pins[2];
+		pins[2] = nid;
+		break;
+	}
+}
+
+/*
+ * Parse all pin widgets and store the useful pin nids to cfg
+ *
+ * The number of line-outs or any primary output is stored in line_outs,
+ * and the corresponding output pins are assigned to line_out_pins[],
+ * in the order of front, rear, CLFE, side, ...
+ *
+ * If more extra outputs (speaker and headphone) are found, the pins are
+ * assisnged to hp_pins[] and speaker_pins[], respectively.  If no line-out jack
+ * is detected, one of speaker of HP pins is assigned as the primary
+ * output, i.e. to line_out_pins[0].  So, line_outs is always positive
+ * if any analog output exists.
+ *
+ * The analog input pins are assigned to inputs array.
+ * The digital input/output pins are assigned to dig_in_pin and dig_out_pin,
+ * respectively.
+ */
+int snd_hda_parse_pin_defcfg(struct hda_codec *codec,
+			     struct auto_pin_cfg *cfg,
+			     const hda_nid_t *ignore_nids,
+			     unsigned int cond_flags)
+{
+	hda_nid_t nid, end_nid;
+	short seq, assoc_line_out;
+	short sequences_line_out[ARRAY_SIZE(cfg->line_out_pins)];
+	short sequences_speaker[ARRAY_SIZE(cfg->speaker_pins)];
+	short sequences_hp[ARRAY_SIZE(cfg->hp_pins)];
+	int i;
+
+	memset(cfg, 0, sizeof(*cfg));
+
+	memset(sequences_line_out, 0, sizeof(sequences_line_out));
+	memset(sequences_speaker, 0, sizeof(sequences_speaker));
+	memset(sequences_hp, 0, sizeof(sequences_hp));
+	assoc_line_out = 0;
+
+	codec->ignore_misc_bit = true;
+	end_nid = codec->start_nid + codec->num_nodes;
+	for (nid = codec->start_nid; nid < end_nid; nid++) {
+		unsigned int wid_caps = get_wcaps(codec, nid);
+		unsigned int wid_type = get_wcaps_type(wid_caps);
+		unsigned int def_conf;
+		short assoc, loc, conn, dev;
+
+		/* read all default configuration for pin complex */
+		if (wid_type != AC_WID_PIN)
+			continue;
+		/* ignore the given nids (e.g. pc-beep returns error) */
+		if (ignore_nids && is_in_nid_list(nid, ignore_nids))
+			continue;
+
+		def_conf = snd_hda_codec_get_pincfg(codec, nid);
+		if (!(get_defcfg_misc(snd_hda_codec_get_pincfg(codec, nid)) &
+		      AC_DEFCFG_MISC_NO_PRESENCE))
+			codec->ignore_misc_bit = false;
+		conn = get_defcfg_connect(def_conf);
+		if (conn == AC_JACK_PORT_NONE)
+			continue;
+		loc = get_defcfg_location(def_conf);
+		dev = get_defcfg_device(def_conf);
+
+		/* workaround for buggy BIOS setups */
+		if (dev == AC_JACK_LINE_OUT) {
+			if (conn == AC_JACK_PORT_FIXED)
+				dev = AC_JACK_SPEAKER;
+		}
+
+		switch (dev) {
+		case AC_JACK_LINE_OUT:
+			seq = get_defcfg_sequence(def_conf);
+			assoc = get_defcfg_association(def_conf);
+
+			if (!(wid_caps & AC_WCAP_STEREO))
+				if (!cfg->mono_out_pin)
+					cfg->mono_out_pin = nid;
+			if (!assoc)
+				continue;
+			if (!assoc_line_out)
+				assoc_line_out = assoc;
+			else if (assoc_line_out != assoc)
+				continue;
+			if (cfg->line_outs >= ARRAY_SIZE(cfg->line_out_pins))
+				continue;
+			cfg->line_out_pins[cfg->line_outs] = nid;
+			sequences_line_out[cfg->line_outs] = seq;
+			cfg->line_outs++;
+			break;
+		case AC_JACK_SPEAKER:
+			seq = get_defcfg_sequence(def_conf);
+			assoc = get_defcfg_association(def_conf);
+			if (cfg->speaker_outs >= ARRAY_SIZE(cfg->speaker_pins))
+				continue;
+			cfg->speaker_pins[cfg->speaker_outs] = nid;
+			sequences_speaker[cfg->speaker_outs] = (assoc << 4) | seq;
+			cfg->speaker_outs++;
+			break;
+		case AC_JACK_HP_OUT:
+			seq = get_defcfg_sequence(def_conf);
+			assoc = get_defcfg_association(def_conf);
+			if (cfg->hp_outs >= ARRAY_SIZE(cfg->hp_pins))
+				continue;
+			cfg->hp_pins[cfg->hp_outs] = nid;
+			sequences_hp[cfg->hp_outs] = (assoc << 4) | seq;
+			cfg->hp_outs++;
+			break;
+		case AC_JACK_MIC_IN:
+			add_auto_cfg_input_pin(cfg, nid, AUTO_PIN_MIC);
+			break;
+		case AC_JACK_LINE_IN:
+			add_auto_cfg_input_pin(cfg, nid, AUTO_PIN_LINE_IN);
+			break;
+		case AC_JACK_CD:
+			add_auto_cfg_input_pin(cfg, nid, AUTO_PIN_CD);
+			break;
+		case AC_JACK_AUX:
+			add_auto_cfg_input_pin(cfg, nid, AUTO_PIN_AUX);
+			break;
+		case AC_JACK_SPDIF_OUT:
+		case AC_JACK_DIG_OTHER_OUT:
+			if (cfg->dig_outs >= ARRAY_SIZE(cfg->dig_out_pins))
+				continue;
+			cfg->dig_out_pins[cfg->dig_outs] = nid;
+			cfg->dig_out_type[cfg->dig_outs] =
+				(loc == AC_JACK_LOC_HDMI) ?
+				HDA_PCM_TYPE_HDMI : HDA_PCM_TYPE_SPDIF;
+			cfg->dig_outs++;
+			break;
+		case AC_JACK_SPDIF_IN:
+		case AC_JACK_DIG_OTHER_IN:
+			cfg->dig_in_pin = nid;
+			if (loc == AC_JACK_LOC_HDMI)
+				cfg->dig_in_type = HDA_PCM_TYPE_HDMI;
+			else
+				cfg->dig_in_type = HDA_PCM_TYPE_SPDIF;
+			break;
+		}
+	}
+
+	/* FIX-UP:
+	 * If no line-out is defined but multiple HPs are found,
+	 * some of them might be the real line-outs.
+	 */
+	if (!cfg->line_outs && cfg->hp_outs > 1 &&
+	    !(cond_flags & HDA_PINCFG_NO_HP_FIXUP)) {
+		int i = 0;
+		while (i < cfg->hp_outs) {
+			/* The real HPs should have the sequence 0x0f */
+			if ((sequences_hp[i] & 0x0f) == 0x0f) {
+				i++;
+				continue;
+			}
+			/* Move it to the line-out table */
+			cfg->line_out_pins[cfg->line_outs] = cfg->hp_pins[i];
+			sequences_line_out[cfg->line_outs] = sequences_hp[i];
+			cfg->line_outs++;
+			cfg->hp_outs--;
+			memmove(cfg->hp_pins + i, cfg->hp_pins + i + 1,
+				sizeof(cfg->hp_pins[0]) * (cfg->hp_outs - i));
+			memmove(sequences_hp + i, sequences_hp + i + 1,
+				sizeof(sequences_hp[0]) * (cfg->hp_outs - i));
+		}
+		memset(cfg->hp_pins + cfg->hp_outs, 0,
+		       sizeof(hda_nid_t) * (AUTO_CFG_MAX_OUTS - cfg->hp_outs));
+		if (!cfg->hp_outs)
+			cfg->line_out_type = AUTO_PIN_HP_OUT;
+
+	}
+
+	/* sort by sequence */
+	sort_pins_by_sequence(cfg->line_out_pins, sequences_line_out,
+			      cfg->line_outs);
+	sort_pins_by_sequence(cfg->speaker_pins, sequences_speaker,
+			      cfg->speaker_outs);
+	sort_pins_by_sequence(cfg->hp_pins, sequences_hp,
+			      cfg->hp_outs);
+
+	/*
+	 * FIX-UP: if no line-outs are detected, try to use speaker or HP pin
+	 * as a primary output
+	 */
+	if (!cfg->line_outs &&
+	    !(cond_flags & HDA_PINCFG_NO_LO_FIXUP)) {
+		if (cfg->speaker_outs) {
+			cfg->line_outs = cfg->speaker_outs;
+			memcpy(cfg->line_out_pins, cfg->speaker_pins,
+			       sizeof(cfg->speaker_pins));
+			cfg->speaker_outs = 0;
+			memset(cfg->speaker_pins, 0, sizeof(cfg->speaker_pins));
+			cfg->line_out_type = AUTO_PIN_SPEAKER_OUT;
+		} else if (cfg->hp_outs) {
+			cfg->line_outs = cfg->hp_outs;
+			memcpy(cfg->line_out_pins, cfg->hp_pins,
+			       sizeof(cfg->hp_pins));
+			cfg->hp_outs = 0;
+			memset(cfg->hp_pins, 0, sizeof(cfg->hp_pins));
+			cfg->line_out_type = AUTO_PIN_HP_OUT;
+		}
+	}
+
+	reorder_outputs(cfg->line_outs, cfg->line_out_pins);
+	reorder_outputs(cfg->hp_outs, cfg->hp_pins);
+	reorder_outputs(cfg->speaker_outs, cfg->speaker_pins);
+
+	sort_autocfg_input_pins(cfg);
+
+	/*
+	 * debug prints of the parsed results
+	 */
+	snd_printd("autoconfig: line_outs=%d (0x%x/0x%x/0x%x/0x%x/0x%x) type:%s\n",
+		   cfg->line_outs, cfg->line_out_pins[0], cfg->line_out_pins[1],
+		   cfg->line_out_pins[2], cfg->line_out_pins[3],
+		   cfg->line_out_pins[4],
+		   cfg->line_out_type == AUTO_PIN_HP_OUT ? "hp" :
+		   (cfg->line_out_type == AUTO_PIN_SPEAKER_OUT ?
+		    "speaker" : "line"));
+	snd_printd("   speaker_outs=%d (0x%x/0x%x/0x%x/0x%x/0x%x)\n",
+		   cfg->speaker_outs, cfg->speaker_pins[0],
+		   cfg->speaker_pins[1], cfg->speaker_pins[2],
+		   cfg->speaker_pins[3], cfg->speaker_pins[4]);
+	snd_printd("   hp_outs=%d (0x%x/0x%x/0x%x/0x%x/0x%x)\n",
+		   cfg->hp_outs, cfg->hp_pins[0],
+		   cfg->hp_pins[1], cfg->hp_pins[2],
+		   cfg->hp_pins[3], cfg->hp_pins[4]);
+	snd_printd("   mono: mono_out=0x%x\n", cfg->mono_out_pin);
+	if (cfg->dig_outs)
+		snd_printd("   dig-out=0x%x/0x%x\n",
+			   cfg->dig_out_pins[0], cfg->dig_out_pins[1]);
+	snd_printd("   inputs:");
+	for (i = 0; i < cfg->num_inputs; i++) {
+		snd_printd(" %s=0x%x",
+			    hda_get_autocfg_input_label(codec, cfg, i),
+			    cfg->inputs[i].pin);
+	}
+	snd_printd("\n");
+	if (cfg->dig_in_pin)
+		snd_printd("   dig-in=0x%x\n", cfg->dig_in_pin);
+
+	return 0;
+}
+EXPORT_SYMBOL_HDA(snd_hda_parse_pin_defcfg);
+
+int snd_hda_get_input_pin_attr(unsigned int def_conf)
+{
+	unsigned int loc = get_defcfg_location(def_conf);
+	unsigned int conn = get_defcfg_connect(def_conf);
+	if (conn == AC_JACK_PORT_NONE)
+		return INPUT_PIN_ATTR_UNUSED;
+	/* Windows may claim the internal mic to be BOTH, too */
+	if (conn == AC_JACK_PORT_FIXED || conn == AC_JACK_PORT_BOTH)
+		return INPUT_PIN_ATTR_INT;
+	if ((loc & 0x30) == AC_JACK_LOC_INTERNAL)
+		return INPUT_PIN_ATTR_INT;
+	if ((loc & 0x30) == AC_JACK_LOC_SEPARATE)
+		return INPUT_PIN_ATTR_DOCK;
+	if (loc == AC_JACK_LOC_REAR)
+		return INPUT_PIN_ATTR_REAR;
+	if (loc == AC_JACK_LOC_FRONT)
+		return INPUT_PIN_ATTR_FRONT;
+	return INPUT_PIN_ATTR_NORMAL;
+}
+EXPORT_SYMBOL_HDA(snd_hda_get_input_pin_attr);
+
+/**
+ * hda_get_input_pin_label - Give a label for the given input pin
+ *
+ * When check_location is true, the function checks the pin location
+ * for mic and line-in pins, and set an appropriate prefix like "Front",
+ * "Rear", "Internal".
+ */
+
+static const char *hda_get_input_pin_label(struct hda_codec *codec,
+					   hda_nid_t pin, bool check_location)
+{
+	unsigned int def_conf;
+	static const char * const mic_names[] = {
+		"Internal Mic", "Dock Mic", "Mic", "Front Mic", "Rear Mic",
+	};
+	int attr;
+
+	def_conf = snd_hda_codec_get_pincfg(codec, pin);
+
+	switch (get_defcfg_device(def_conf)) {
+	case AC_JACK_MIC_IN:
+		if (!check_location)
+			return "Mic";
+		attr = snd_hda_get_input_pin_attr(def_conf);
+		if (!attr)
+			return "None";
+		return mic_names[attr - 1];
+	case AC_JACK_LINE_IN:
+		if (!check_location)
+			return "Line";
+		attr = snd_hda_get_input_pin_attr(def_conf);
+		if (!attr)
+			return "None";
+		if (attr == INPUT_PIN_ATTR_DOCK)
+			return "Dock Line";
+		return "Line";
+	case AC_JACK_AUX:
+		return "Aux";
+	case AC_JACK_CD:
+		return "CD";
+	case AC_JACK_SPDIF_IN:
+		return "SPDIF In";
+	case AC_JACK_DIG_OTHER_IN:
+		return "Digital In";
+	default:
+		return "Misc";
+	}
+}
+
+/* Check whether the location prefix needs to be added to the label.
+ * If all mic-jacks are in the same location (e.g. rear panel), we don't
+ * have to put "Front" prefix to each label.  In such a case, returns false.
+ */
+static int check_mic_location_need(struct hda_codec *codec,
+				   const struct auto_pin_cfg *cfg,
+				   int input)
+{
+	unsigned int defc;
+	int i, attr, attr2;
+
+	defc = snd_hda_codec_get_pincfg(codec, cfg->inputs[input].pin);
+	attr = snd_hda_get_input_pin_attr(defc);
+	/* for internal or docking mics, we need locations */
+	if (attr <= INPUT_PIN_ATTR_NORMAL)
+		return 1;
+
+	attr = 0;
+	for (i = 0; i < cfg->num_inputs; i++) {
+		defc = snd_hda_codec_get_pincfg(codec, cfg->inputs[i].pin);
+		attr2 = snd_hda_get_input_pin_attr(defc);
+		if (attr2 >= INPUT_PIN_ATTR_NORMAL) {
+			if (attr && attr != attr2)
+				return 1; /* different locations found */
+			attr = attr2;
+		}
+	}
+	return 0;
+}
+
+/**
+ * hda_get_autocfg_input_label - Get a label for the given input
+ *
+ * Get a label for the given input pin defined by the autocfg item.
+ * Unlike hda_get_input_pin_label(), this function checks all inputs
+ * defined in autocfg and avoids the redundant mic/line prefix as much as
+ * possible.
+ */
+const char *hda_get_autocfg_input_label(struct hda_codec *codec,
+					const struct auto_pin_cfg *cfg,
+					int input)
+{
+	int type = cfg->inputs[input].type;
+	int has_multiple_pins = 0;
+
+	if ((input > 0 && cfg->inputs[input - 1].type == type) ||
+	    (input < cfg->num_inputs - 1 && cfg->inputs[input + 1].type == type))
+		has_multiple_pins = 1;
+	if (has_multiple_pins && type == AUTO_PIN_MIC)
+		has_multiple_pins &= check_mic_location_need(codec, cfg, input);
+	return hda_get_input_pin_label(codec, cfg->inputs[input].pin,
+				       has_multiple_pins);
+}
+EXPORT_SYMBOL_HDA(hda_get_autocfg_input_label);
+
+/* return the position of NID in the list, or -1 if not found */
+static int find_idx_in_nid_list(hda_nid_t nid, const hda_nid_t *list, int nums)
+{
+	int i;
+	for (i = 0; i < nums; i++)
+		if (list[i] == nid)
+			return i;
+	return -1;
+}
+
+/* get a unique suffix or an index number */
+static const char *check_output_sfx(hda_nid_t nid, const hda_nid_t *pins,
+				    int num_pins, int *indexp)
+{
+	static const char * const channel_sfx[] = {
+		" Front", " Surround", " CLFE", " Side"
+	};
+	int i;
+
+	i = find_idx_in_nid_list(nid, pins, num_pins);
+	if (i < 0)
+		return NULL;
+	if (num_pins == 1)
+		return "";
+	if (num_pins > ARRAY_SIZE(channel_sfx)) {
+		if (indexp)
+			*indexp = i;
+		return "";
+	}
+	return channel_sfx[i];
+}
+
+static int fill_audio_out_name(struct hda_codec *codec, hda_nid_t nid,
+			       const struct auto_pin_cfg *cfg,
+			       const char *name, char *label, int maxlen,
+			       int *indexp)
+{
+	unsigned int def_conf = snd_hda_codec_get_pincfg(codec, nid);
+	int attr = snd_hda_get_input_pin_attr(def_conf);
+	const char *pfx = "", *sfx = "";
+
+	/* handle as a speaker if it's a fixed line-out */
+	if (!strcmp(name, "Line Out") && attr == INPUT_PIN_ATTR_INT)
+		name = "Speaker";
+	/* check the location */
+	switch (attr) {
+	case INPUT_PIN_ATTR_DOCK:
+		pfx = "Dock ";
+		break;
+	case INPUT_PIN_ATTR_FRONT:
+		pfx = "Front ";
+		break;
+	}
+	if (cfg) {
+		/* try to give a unique suffix if needed */
+		sfx = check_output_sfx(nid, cfg->line_out_pins, cfg->line_outs,
+				       indexp);
+		if (!sfx)
+			sfx = check_output_sfx(nid, cfg->speaker_pins, cfg->speaker_outs,
+					       indexp);
+		if (!sfx) {
+			/* don't add channel suffix for Headphone controls */
+			int idx = find_idx_in_nid_list(nid, cfg->hp_pins,
+						       cfg->hp_outs);
+			if (idx >= 0)
+				*indexp = idx;
+			sfx = "";
+		}
+	}
+	snprintf(label, maxlen, "%s%s%s", pfx, name, sfx);
+	return 1;
+}
+
+/**
+ * snd_hda_get_pin_label - Get a label for the given I/O pin
+ *
+ * Get a label for the given pin.  This function works for both input and
+ * output pins.  When @cfg is given as non-NULL, the function tries to get
+ * an optimized label using hda_get_autocfg_input_label().
+ *
+ * This function tries to give a unique label string for the pin as much as
+ * possible.  For example, when the multiple line-outs are present, it adds
+ * the channel suffix like "Front", "Surround", etc (only when @cfg is given).
+ * If no unique name with a suffix is available and @indexp is non-NULL, the
+ * index number is stored in the pointer.
+ */
+int snd_hda_get_pin_label(struct hda_codec *codec, hda_nid_t nid,
+			  const struct auto_pin_cfg *cfg,
+			  char *label, int maxlen, int *indexp)
+{
+	unsigned int def_conf = snd_hda_codec_get_pincfg(codec, nid);
+	const char *name = NULL;
+	int i;
+
+	if (indexp)
+		*indexp = 0;
+	if (get_defcfg_connect(def_conf) == AC_JACK_PORT_NONE)
+		return 0;
+
+	switch (get_defcfg_device(def_conf)) {
+	case AC_JACK_LINE_OUT:
+		return fill_audio_out_name(codec, nid, cfg, "Line Out",
+					   label, maxlen, indexp);
+	case AC_JACK_SPEAKER:
+		return fill_audio_out_name(codec, nid, cfg, "Speaker",
+					   label, maxlen, indexp);
+	case AC_JACK_HP_OUT:
+		return fill_audio_out_name(codec, nid, cfg, "Headphone",
+					   label, maxlen, indexp);
+	case AC_JACK_SPDIF_OUT:
+	case AC_JACK_DIG_OTHER_OUT:
+		if (get_defcfg_location(def_conf) == AC_JACK_LOC_HDMI)
+			name = "HDMI";
+		else
+			name = "SPDIF";
+		if (cfg && indexp) {
+			i = find_idx_in_nid_list(nid, cfg->dig_out_pins,
+						 cfg->dig_outs);
+			if (i >= 0)
+				*indexp = i;
+		}
+		break;
+	default:
+		if (cfg) {
+			for (i = 0; i < cfg->num_inputs; i++) {
+				if (cfg->inputs[i].pin != nid)
+					continue;
+				name = hda_get_autocfg_input_label(codec, cfg, i);
+				if (name)
+					break;
+			}
+		}
+		if (!name)
+			name = hda_get_input_pin_label(codec, nid, true);
+		break;
+	}
+	if (!name)
+		return 0;
+	strlcpy(label, name, maxlen);
+	return 1;
+}
+EXPORT_SYMBOL_HDA(snd_hda_get_pin_label);
+
+int snd_hda_gen_add_verbs(struct hda_gen_spec *spec,
+			  const struct hda_verb *list)
+{
+	const struct hda_verb **v;
+	snd_array_init(&spec->verbs, sizeof(struct hda_verb *), 8);
+	v = snd_array_new(&spec->verbs);
+	if (!v)
+		return -ENOMEM;
+	*v = list;
+	return 0;
+}
+EXPORT_SYMBOL_HDA(snd_hda_gen_add_verbs);
+
+void snd_hda_gen_apply_verbs(struct hda_codec *codec)
+{
+	struct hda_gen_spec *spec = codec->spec;
+	int i;
+	for (i = 0; i < spec->verbs.used; i++) {
+		struct hda_verb **v = snd_array_elem(&spec->verbs, i);
+		snd_hda_sequence_write(codec, *v);
+	}
+}
+EXPORT_SYMBOL_HDA(snd_hda_gen_apply_verbs);
+
+void snd_hda_apply_pincfgs(struct hda_codec *codec,
+			   const struct hda_pintbl *cfg)
+{
+	for (; cfg->nid; cfg++)
+		snd_hda_codec_set_pincfg(codec, cfg->nid, cfg->val);
+}
+EXPORT_SYMBOL_HDA(snd_hda_apply_pincfgs);
+
+void snd_hda_apply_fixup(struct hda_codec *codec, int action)
+{
+	struct hda_gen_spec *spec = codec->spec;
+	int id = spec->fixup_id;
+#ifdef CONFIG_SND_DEBUG_VERBOSE
+	const char *modelname = spec->fixup_name;
+#endif
+	int depth = 0;
+
+	if (!spec->fixup_list)
+		return;
+
+	while (id >= 0) {
+		const struct hda_fixup *fix = spec->fixup_list + id;
+
+		switch (fix->type) {
+		case HDA_FIXUP_PINS:
+			if (action != HDA_FIXUP_ACT_PRE_PROBE || !fix->v.pins)
+				break;
+			snd_printdd(KERN_INFO SFX
+				    "%s: Apply pincfg for %s\n",
+				    codec->chip_name, modelname);
+			snd_hda_apply_pincfgs(codec, fix->v.pins);
+			break;
+		case HDA_FIXUP_VERBS:
+			if (action != HDA_FIXUP_ACT_PROBE || !fix->v.verbs)
+				break;
+			snd_printdd(KERN_INFO SFX
+				    "%s: Apply fix-verbs for %s\n",
+				    codec->chip_name, modelname);
+			snd_hda_gen_add_verbs(codec->spec, fix->v.verbs);
+			break;
+		case HDA_FIXUP_FUNC:
+			if (!fix->v.func)
+				break;
+			snd_printdd(KERN_INFO SFX
+				    "%s: Apply fix-func for %s\n",
+				    codec->chip_name, modelname);
+			fix->v.func(codec, fix, action);
+			break;
+		default:
+			snd_printk(KERN_ERR SFX
+				   "%s: Invalid fixup type %d\n",
+				   codec->chip_name, fix->type);
+			break;
+		}
+		if (!fix->chained)
+			break;
+		if (++depth > 10)
+			break;
+		id = fix->chain_id;
+	}
+}
+EXPORT_SYMBOL_HDA(snd_hda_apply_fixup);
+
+void snd_hda_pick_fixup(struct hda_codec *codec,
+			const struct hda_model_fixup *models,
+			const struct snd_pci_quirk *quirk,
+			const struct hda_fixup *fixlist)
+{
+	struct hda_gen_spec *spec = codec->spec;
+	const struct snd_pci_quirk *q;
+	int id = -1;
+	const char *name = NULL;
+
+	/* when model=nofixup is given, don't pick up any fixups */
+	if (codec->modelname && !strcmp(codec->modelname, "nofixup")) {
+		spec->fixup_list = NULL;
+		spec->fixup_id = -1;
+		return;
+	}
+
+	if (codec->modelname && models) {
+		while (models->name) {
+			if (!strcmp(codec->modelname, models->name)) {
+				id = models->id;
+				name = models->name;
+				break;
+			}
+			models++;
+		}
+	}
+	if (id < 0) {
+		q = snd_pci_quirk_lookup(codec->bus->pci, quirk);
+		if (q) {
+			id = q->value;
+#ifdef CONFIG_SND_DEBUG_VERBOSE
+			name = q->name;
+#endif
+		}
+	}
+	if (id < 0) {
+		for (q = quirk; q->subvendor; q++) {
+			unsigned int vendorid =
+				q->subdevice | (q->subvendor << 16);
+			if (vendorid == codec->subsystem_id) {
+				id = q->value;
+#ifdef CONFIG_SND_DEBUG_VERBOSE
+				name = q->name;
+#endif
+				break;
+			}
+		}
+	}
+
+	spec->fixup_id = id;
+	if (id >= 0) {
+		spec->fixup_list = fixlist;
+		spec->fixup_name = name;
+	}
+}
+EXPORT_SYMBOL_HDA(snd_hda_pick_fixup);
diff --git a/sound/pci/hda/hda_auto_parser.h b/sound/pci/hda/hda_auto_parser.h
new file mode 100644
index 000000000000..2a7889dfbd1b
--- /dev/null
+++ b/sound/pci/hda/hda_auto_parser.h
@@ -0,0 +1,160 @@
+/*
+ * BIOS auto-parser helper functions for HD-audio
+ *
+ * Copyright (c) 2012 Takashi Iwai <tiwai@suse.de>
+ *
+ * This driver is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ */
+
+#ifndef __SOUND_HDA_AUTO_PARSER_H
+#define __SOUND_HDA_AUTO_PARSER_H
+
+/*
+ * Helper for automatic pin configuration
+ */
+
+enum {
+	AUTO_PIN_MIC,
+	AUTO_PIN_LINE_IN,
+	AUTO_PIN_CD,
+	AUTO_PIN_AUX,
+	AUTO_PIN_LAST
+};
+
+enum {
+	AUTO_PIN_LINE_OUT,
+	AUTO_PIN_SPEAKER_OUT,
+	AUTO_PIN_HP_OUT
+};
+
+#define AUTO_CFG_MAX_OUTS	HDA_MAX_OUTS
+#define AUTO_CFG_MAX_INS	8
+
+struct auto_pin_cfg_item {
+	hda_nid_t pin;
+	int type;
+};
+
+struct auto_pin_cfg;
+const char *hda_get_autocfg_input_label(struct hda_codec *codec,
+					const struct auto_pin_cfg *cfg,
+					int input);
+int snd_hda_get_pin_label(struct hda_codec *codec, hda_nid_t nid,
+			  const struct auto_pin_cfg *cfg,
+			  char *label, int maxlen, int *indexp);
+
+enum {
+	INPUT_PIN_ATTR_UNUSED,	/* pin not connected */
+	INPUT_PIN_ATTR_INT,	/* internal mic/line-in */
+	INPUT_PIN_ATTR_DOCK,	/* docking mic/line-in */
+	INPUT_PIN_ATTR_NORMAL,	/* mic/line-in jack */
+	INPUT_PIN_ATTR_FRONT,	/* mic/line-in jack in front */
+	INPUT_PIN_ATTR_REAR,	/* mic/line-in jack in rear */
+};
+
+int snd_hda_get_input_pin_attr(unsigned int def_conf);
+
+struct auto_pin_cfg {
+	int line_outs;
+	/* sorted in the order of Front/Surr/CLFE/Side */
+	hda_nid_t line_out_pins[AUTO_CFG_MAX_OUTS];
+	int speaker_outs;
+	hda_nid_t speaker_pins[AUTO_CFG_MAX_OUTS];
+	int hp_outs;
+	int line_out_type;	/* AUTO_PIN_XXX_OUT */
+	hda_nid_t hp_pins[AUTO_CFG_MAX_OUTS];
+	int num_inputs;
+	struct auto_pin_cfg_item inputs[AUTO_CFG_MAX_INS];
+	int dig_outs;
+	hda_nid_t dig_out_pins[2];
+	hda_nid_t dig_in_pin;
+	hda_nid_t mono_out_pin;
+	int dig_out_type[2]; /* HDA_PCM_TYPE_XXX */
+	int dig_in_type; /* HDA_PCM_TYPE_XXX */
+};
+
+/* bit-flags for snd_hda_parse_pin_def_config() behavior */
+#define HDA_PINCFG_NO_HP_FIXUP	(1 << 0) /* no HP-split */
+#define HDA_PINCFG_NO_LO_FIXUP	(1 << 1) /* don't take other outs as LO */
+
+int snd_hda_parse_pin_defcfg(struct hda_codec *codec,
+			     struct auto_pin_cfg *cfg,
+			     const hda_nid_t *ignore_nids,
+			     unsigned int cond_flags);
+
+/* older function */
+#define snd_hda_parse_pin_def_config(codec, cfg, ignore) \
+	snd_hda_parse_pin_defcfg(codec, cfg, ignore, 0)
+
+/*
+ */
+
+struct hda_gen_spec {
+	/* fix-up list */
+	int fixup_id;
+	const struct hda_fixup *fixup_list;
+	const char *fixup_name;
+
+	/* additional init verbs */
+	struct snd_array verbs;
+};
+
+
+/*
+ * Fix-up pin default configurations and add default verbs
+ */
+
+struct hda_pintbl {
+	hda_nid_t nid;
+	u32 val;
+};
+
+struct hda_model_fixup {
+	const int id;
+	const char *name;
+};
+
+struct hda_fixup {
+	int type;
+	bool chained;
+	int chain_id;
+	union {
+		const struct hda_pintbl *pins;
+		const struct hda_verb *verbs;
+		void (*func)(struct hda_codec *codec,
+			     const struct hda_fixup *fix,
+			     int action);
+	} v;
+};
+
+/* fixup types */
+enum {
+	HDA_FIXUP_INVALID,
+	HDA_FIXUP_PINS,
+	HDA_FIXUP_VERBS,
+	HDA_FIXUP_FUNC,
+};
+
+/* fixup action definitions */
+enum {
+	HDA_FIXUP_ACT_PRE_PROBE,
+	HDA_FIXUP_ACT_PROBE,
+	HDA_FIXUP_ACT_INIT,
+	HDA_FIXUP_ACT_BUILD,
+};
+
+int snd_hda_gen_add_verbs(struct hda_gen_spec *spec,
+			  const struct hda_verb *list);
+void snd_hda_gen_apply_verbs(struct hda_codec *codec);
+void snd_hda_apply_pincfgs(struct hda_codec *codec,
+			   const struct hda_pintbl *cfg);
+void snd_hda_apply_fixup(struct hda_codec *codec, int action);
+void snd_hda_pick_fixup(struct hda_codec *codec,
+			const struct hda_model_fixup *models,
+			const struct snd_pci_quirk *quirk,
+			const struct hda_fixup *fixlist);
+
+#endif /* __SOUND_HDA_AUTO_PARSER_H */
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 841475cc13b6..eb09a3348325 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -334,54 +334,23 @@ static hda_nid_t *lookup_conn_list(struct snd_array *array, hda_nid_t nid)
 	return NULL;
 }
 
-/**
- * snd_hda_get_conn_list - get connection list
- * @codec: the HDA codec
- * @nid: NID to parse
- * @listp: the pointer to store NID list
- *
- * Parses the connection list of the given widget and stores the list
- * of NIDs.
- *
- * Returns the number of connections, or a negative error code.
- */
-int snd_hda_get_conn_list(struct hda_codec *codec, hda_nid_t nid,
-			  const hda_nid_t **listp)
+/* read the connection and add to the cache */
+static int read_and_add_raw_conns(struct hda_codec *codec, hda_nid_t nid)
 {
-	struct snd_array *array = &codec->conn_lists;
-	int len, err;
 	hda_nid_t list[HDA_MAX_CONNECTIONS];
-	hda_nid_t *p;
-	bool added = false;
+	int len;
 
- again:
-	/* if the connection-list is already cached, read it */
-	p = lookup_conn_list(array, nid);
-	if (p) {
-		if (listp)
-			*listp = p + 2;
-		return p[1];
-	}
-	if (snd_BUG_ON(added))
-		return -EINVAL;
-
-	/* read the connection and add to the cache */
-	len = snd_hda_get_raw_connections(codec, nid, list, HDA_MAX_CONNECTIONS);
+	len = snd_hda_get_raw_connections(codec, nid, list, ARRAY_SIZE(list));
 	if (len < 0)
 		return len;
-	err = snd_hda_override_conn_list(codec, nid, len, list);
-	if (err < 0)
-		return err;
-	added = true;
-	goto again;
+	return snd_hda_override_conn_list(codec, nid, len, list);
 }
-EXPORT_SYMBOL_HDA(snd_hda_get_conn_list);
 
 /**
  * snd_hda_get_connections - copy connection list
  * @codec: the HDA codec
  * @nid: NID to parse
- * @conn_list: connection list array
+ * @conn_list: connection list array; when NULL, checks only the size
  * @max_conns: max. number of connections to store
  *
  * Parses the connection list of the given widget and stores the list
@@ -390,21 +359,41 @@ EXPORT_SYMBOL_HDA(snd_hda_get_conn_list);
  * Returns the number of connections, or a negative error code.
  */
 int snd_hda_get_connections(struct hda_codec *codec, hda_nid_t nid,
-			     hda_nid_t *conn_list, int max_conns)
+			    hda_nid_t *conn_list, int max_conns)
 {
-	const hda_nid_t *list;
-	int len = snd_hda_get_conn_list(codec, nid, &list);
+	struct snd_array *array = &codec->conn_lists;
+	int len;
+	hda_nid_t *p;
+	bool added = false;
 
-	if (len <= 0)
-		return len;
-	if (len > max_conns) {
-		snd_printk(KERN_ERR "hda_codec: "
-			   "Too many connections %d for NID 0x%x\n",
-			   len, nid);
-		return -EINVAL;
+ again:
+	mutex_lock(&codec->hash_mutex);
+	len = -1;
+	/* if the connection-list is already cached, read it */
+	p = lookup_conn_list(array, nid);
+	if (p) {
+		len = p[1];
+		if (conn_list && len > max_conns) {
+			snd_printk(KERN_ERR "hda_codec: "
+				   "Too many connections %d for NID 0x%x\n",
+				   len, nid);
+			mutex_unlock(&codec->hash_mutex);
+			return -EINVAL;
+		}
+		if (conn_list && len)
+			memcpy(conn_list, p + 2, len * sizeof(hda_nid_t));
 	}
-	memcpy(conn_list, list, len * sizeof(hda_nid_t));
-	return len;
+	mutex_unlock(&codec->hash_mutex);
+	if (len >= 0)
+		return len;
+	if (snd_BUG_ON(added))
+		return -EINVAL;
+
+	len = read_and_add_raw_conns(codec, nid);
+	if (len < 0)
+		return len;
+	added = true;
+	goto again;
 }
 EXPORT_SYMBOL_HDA(snd_hda_get_connections);
 
@@ -543,6 +532,7 @@ int snd_hda_override_conn_list(struct hda_codec *codec, hda_nid_t nid, int len,
 	hda_nid_t *p;
 	int i, old_used;
 
+	mutex_lock(&codec->hash_mutex);
 	p = lookup_conn_list(array, nid);
 	if (p)
 		*p = -1; /* invalidate the old entry */
@@ -553,10 +543,12 @@ int snd_hda_override_conn_list(struct hda_codec *codec, hda_nid_t nid, int len,
 	for (i = 0; i < len; i++)
 		if (!add_conn_list(array, list[i]))
 			goto error_add;
+	mutex_unlock(&codec->hash_mutex);
 	return 0;
 
  error_add:
 	array->used = old_used;
+	mutex_unlock(&codec->hash_mutex);
 	return -ENOMEM;
 }
 EXPORT_SYMBOL_HDA(snd_hda_override_conn_list);
@@ -1255,6 +1247,7 @@ int /*__devinit*/ snd_hda_codec_new(struct hda_bus *bus,
 	codec->addr = codec_addr;
 	mutex_init(&codec->spdif_mutex);
 	mutex_init(&codec->control_mutex);
+	mutex_init(&codec->hash_mutex);
 	init_hda_cache(&codec->amp_cache, sizeof(struct hda_amp_info));
 	init_hda_cache(&codec->cmd_cache, sizeof(struct hda_cache_head));
 	snd_array_init(&codec->mixers, sizeof(struct hda_nid_item), 32);
@@ -1264,15 +1257,9 @@ int /*__devinit*/ snd_hda_codec_new(struct hda_bus *bus,
 	snd_array_init(&codec->cvt_setups, sizeof(struct hda_cvt_setup), 8);
 	snd_array_init(&codec->conn_lists, sizeof(hda_nid_t), 64);
 	snd_array_init(&codec->spdif_out, sizeof(struct hda_spdif_out), 16);
-	if (codec->bus->modelname) {
-		codec->modelname = kstrdup(codec->bus->modelname, GFP_KERNEL);
-		if (!codec->modelname) {
-			snd_hda_codec_free(codec);
-			return -ENODEV;
-		}
-	}
 
 #ifdef CONFIG_SND_HDA_POWER_SAVE
+	spin_lock_init(&codec->power_lock);
 	INIT_DELAYED_WORK(&codec->power_work, hda_power_work);
 	/* snd_hda_codec_new() marks the codec as power-up, and leave it as is.
 	 * the caller has to power down appropriatley after initialization
@@ -1281,6 +1268,14 @@ int /*__devinit*/ snd_hda_codec_new(struct hda_bus *bus,
 	hda_keep_power_on(codec);
 #endif
 
+	if (codec->bus->modelname) {
+		codec->modelname = kstrdup(codec->bus->modelname, GFP_KERNEL);
+		if (!codec->modelname) {
+			snd_hda_codec_free(codec);
+			return -ENODEV;
+		}
+	}
+
 	list_add_tail(&codec->list, &bus->codec_list);
 	bus->caddr_tbl[codec_addr] = codec;
 
@@ -1603,6 +1598,60 @@ get_alloc_amp_hash(struct hda_codec *codec, u32 key)
 	return (struct hda_amp_info *)get_alloc_hash(&codec->amp_cache, key);
 }
 
+/* overwrite the value with the key in the caps hash */
+static int write_caps_hash(struct hda_codec *codec, u32 key, unsigned int val)
+{
+	struct hda_amp_info *info;
+
+	mutex_lock(&codec->hash_mutex);
+	info = get_alloc_amp_hash(codec, key);
+	if (!info) {
+		mutex_unlock(&codec->hash_mutex);
+		return -EINVAL;
+	}
+	info->amp_caps = val;
+	info->head.val |= INFO_AMP_CAPS;
+	mutex_unlock(&codec->hash_mutex);
+	return 0;
+}
+
+/* query the value from the caps hash; if not found, fetch the current
+ * value from the given function and store in the hash
+ */
+static unsigned int
+query_caps_hash(struct hda_codec *codec, hda_nid_t nid, int dir, u32 key,
+		unsigned int (*func)(struct hda_codec *, hda_nid_t, int))
+{
+	struct hda_amp_info *info;
+	unsigned int val;
+
+	mutex_lock(&codec->hash_mutex);
+	info = get_alloc_amp_hash(codec, key);
+	if (!info) {
+		mutex_unlock(&codec->hash_mutex);
+		return 0;
+	}
+	if (!(info->head.val & INFO_AMP_CAPS)) {
+		mutex_unlock(&codec->hash_mutex); /* for reentrance */
+		val = func(codec, nid, dir);
+		write_caps_hash(codec, key, val);
+	} else {
+		val = info->amp_caps;
+		mutex_unlock(&codec->hash_mutex);
+	}
+	return val;
+}
+
+static unsigned int read_amp_cap(struct hda_codec *codec, hda_nid_t nid,
+				 int direction)
+{
+	if (!(get_wcaps(codec, nid) & AC_WCAP_AMP_OVRD))
+		nid = codec->afg;
+	return snd_hda_param_read(codec, nid,
+				  direction == HDA_OUTPUT ?
+				  AC_PAR_AMP_OUT_CAP : AC_PAR_AMP_IN_CAP);
+}
+
 /**
  * query_amp_caps - query AMP capabilities
  * @codec: the HD-auio codec
@@ -1617,22 +1666,9 @@ get_alloc_amp_hash(struct hda_codec *codec, u32 key)
  */
 u32 query_amp_caps(struct hda_codec *codec, hda_nid_t nid, int direction)
 {
-	struct hda_amp_info *info;
-
-	info = get_alloc_amp_hash(codec, HDA_HASH_KEY(nid, direction, 0));
-	if (!info)
-		return 0;
-	if (!(info->head.val & INFO_AMP_CAPS)) {
-		if (!(get_wcaps(codec, nid) & AC_WCAP_AMP_OVRD))
-			nid = codec->afg;
-		info->amp_caps = snd_hda_param_read(codec, nid,
-						    direction == HDA_OUTPUT ?
-						    AC_PAR_AMP_OUT_CAP :
-						    AC_PAR_AMP_IN_CAP);
-		if (info->amp_caps)
-			info->head.val |= INFO_AMP_CAPS;
-	}
-	return info->amp_caps;
+	return query_caps_hash(codec, nid, direction,
+			       HDA_HASH_KEY(nid, direction, 0),
+			       read_amp_cap);
 }
 EXPORT_SYMBOL_HDA(query_amp_caps);
 
@@ -1652,34 +1688,12 @@ EXPORT_SYMBOL_HDA(query_amp_caps);
 int snd_hda_override_amp_caps(struct hda_codec *codec, hda_nid_t nid, int dir,
 			      unsigned int caps)
 {
-	struct hda_amp_info *info;
-
-	info = get_alloc_amp_hash(codec, HDA_HASH_KEY(nid, dir, 0));
-	if (!info)
-		return -EINVAL;
-	info->amp_caps = caps;
-	info->head.val |= INFO_AMP_CAPS;
-	return 0;
+	return write_caps_hash(codec, HDA_HASH_KEY(nid, dir, 0), caps);
 }
 EXPORT_SYMBOL_HDA(snd_hda_override_amp_caps);
 
-static unsigned int
-query_caps_hash(struct hda_codec *codec, hda_nid_t nid, u32 key,
-		unsigned int (*func)(struct hda_codec *, hda_nid_t))
-{
-	struct hda_amp_info *info;
-
-	info = get_alloc_amp_hash(codec, key);
-	if (!info)
-		return 0;
-	if (!info->head.val) {
-		info->head.val |= INFO_AMP_CAPS;
-		info->amp_caps = func(codec, nid);
-	}
-	return info->amp_caps;
-}
-
-static unsigned int read_pin_cap(struct hda_codec *codec, hda_nid_t nid)
+static unsigned int read_pin_cap(struct hda_codec *codec, hda_nid_t nid,
+				 int dir)
 {
 	return snd_hda_param_read(codec, nid, AC_PAR_PIN_CAP);
 }
@@ -1697,7 +1711,7 @@ static unsigned int read_pin_cap(struct hda_codec *codec, hda_nid_t nid)
  */
 u32 snd_hda_query_pin_caps(struct hda_codec *codec, hda_nid_t nid)
 {
-	return query_caps_hash(codec, nid, HDA_HASH_PINCAP_KEY(nid),
+	return query_caps_hash(codec, nid, 0, HDA_HASH_PINCAP_KEY(nid),
 			       read_pin_cap);
 }
 EXPORT_SYMBOL_HDA(snd_hda_query_pin_caps);
@@ -1715,41 +1729,47 @@ EXPORT_SYMBOL_HDA(snd_hda_query_pin_caps);
 int snd_hda_override_pin_caps(struct hda_codec *codec, hda_nid_t nid,
 			      unsigned int caps)
 {
-	struct hda_amp_info *info;
-	info = get_alloc_amp_hash(codec, HDA_HASH_PINCAP_KEY(nid));
-	if (!info)
-		return -ENOMEM;
-	info->amp_caps = caps;
-	info->head.val |= INFO_AMP_CAPS;
-	return 0;
+	return write_caps_hash(codec, HDA_HASH_PINCAP_KEY(nid), caps);
 }
 EXPORT_SYMBOL_HDA(snd_hda_override_pin_caps);
 
-/*
- * read the current volume to info
- * if the cache exists, read the cache value.
+/* read or sync the hash value with the current value;
+ * call within hash_mutex
  */
-static unsigned int get_vol_mute(struct hda_codec *codec,
-				 struct hda_amp_info *info, hda_nid_t nid,
-				 int ch, int direction, int index)
+static struct hda_amp_info *
+update_amp_hash(struct hda_codec *codec, hda_nid_t nid, int ch,
+		int direction, int index)
 {
-	u32 val, parm;
+	struct hda_amp_info *info;
+	unsigned int parm, val = 0;
+	bool val_read = false;
 
-	if (info->head.val & INFO_AMP_VOL(ch))
-		return info->vol[ch];
-
-	parm = ch ? AC_AMP_GET_RIGHT : AC_AMP_GET_LEFT;
-	parm |= direction == HDA_OUTPUT ? AC_AMP_GET_OUTPUT : AC_AMP_GET_INPUT;
-	parm |= index;
-	val = snd_hda_codec_read(codec, nid, 0,
+ retry:
+	info = get_alloc_amp_hash(codec, HDA_HASH_KEY(nid, direction, index));
+	if (!info)
+		return NULL;
+	if (!(info->head.val & INFO_AMP_VOL(ch))) {
+		if (!val_read) {
+			mutex_unlock(&codec->hash_mutex);
+			parm = ch ? AC_AMP_GET_RIGHT : AC_AMP_GET_LEFT;
+			parm |= direction == HDA_OUTPUT ?
+				AC_AMP_GET_OUTPUT : AC_AMP_GET_INPUT;
+			parm |= index;
+			val = snd_hda_codec_read(codec, nid, 0,
 				 AC_VERB_GET_AMP_GAIN_MUTE, parm);
-	info->vol[ch] = val & 0xff;
-	info->head.val |= INFO_AMP_VOL(ch);
-	return info->vol[ch];
+			val &= 0xff;
+			val_read = true;
+			mutex_lock(&codec->hash_mutex);
+			goto retry;
+		}
+		info->vol[ch] = val;
+		info->head.val |= INFO_AMP_VOL(ch);
+	}
+	return info;
 }
 
 /*
- * write the current volume in info to the h/w and update the cache
+ * write the current volume in info to the h/w
  */
 static void put_vol_mute(struct hda_codec *codec, struct hda_amp_info *info,
 			 hda_nid_t nid, int ch, int direction, int index,
@@ -1766,7 +1786,6 @@ static void put_vol_mute(struct hda_codec *codec, struct hda_amp_info *info,
 	else
 		parm |= val;
 	snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, parm);
-	info->vol[ch] = val;
 }
 
 /**
@@ -1783,10 +1802,14 @@ int snd_hda_codec_amp_read(struct hda_codec *codec, hda_nid_t nid, int ch,
 			   int direction, int index)
 {
 	struct hda_amp_info *info;
-	info = get_alloc_amp_hash(codec, HDA_HASH_KEY(nid, direction, index));
-	if (!info)
-		return 0;
-	return get_vol_mute(codec, info, nid, ch, direction, index);
+	unsigned int val = 0;
+
+	mutex_lock(&codec->hash_mutex);
+	info = update_amp_hash(codec, nid, ch, direction, index);
+	if (info)
+		val = info->vol[ch];
+	mutex_unlock(&codec->hash_mutex);
+	return val;
 }
 EXPORT_SYMBOL_HDA(snd_hda_codec_amp_read);
 
@@ -1808,15 +1831,23 @@ int snd_hda_codec_amp_update(struct hda_codec *codec, hda_nid_t nid, int ch,
 {
 	struct hda_amp_info *info;
 
-	info = get_alloc_amp_hash(codec, HDA_HASH_KEY(nid, direction, idx));
-	if (!info)
-		return 0;
 	if (snd_BUG_ON(mask & ~0xff))
 		mask &= 0xff;
 	val &= mask;
-	val |= get_vol_mute(codec, info, nid, ch, direction, idx) & ~mask;
-	if (info->vol[ch] == val)
+
+	mutex_lock(&codec->hash_mutex);
+	info = update_amp_hash(codec, nid, ch, direction, idx);
+	if (!info) {
+		mutex_unlock(&codec->hash_mutex);
 		return 0;
+	}
+	val |= info->vol[ch] & ~mask;
+	if (info->vol[ch] == val) {
+		mutex_unlock(&codec->hash_mutex);
+		return 0;
+	}
+	info->vol[ch] = val;
+	mutex_unlock(&codec->hash_mutex);
 	put_vol_mute(codec, info, nid, ch, direction, idx, val);
 	return 1;
 }
@@ -2263,7 +2294,10 @@ int snd_hda_codec_reset(struct hda_codec *codec)
 	/* OK, let it free */
 
 #ifdef CONFIG_SND_HDA_POWER_SAVE
-	cancel_delayed_work(&codec->power_work);
+	cancel_delayed_work_sync(&codec->power_work);
+	codec->power_on = 0;
+	codec->power_transition = 0;
+	codec->power_jiffies = jiffies;
 	flush_workqueue(codec->bus->workq);
 #endif
 	snd_hda_ctls_clear(codec);
@@ -2859,12 +2893,15 @@ static int snd_hda_spdif_default_get(struct snd_kcontrol *kcontrol,
 {
 	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
 	int idx = kcontrol->private_value;
-	struct hda_spdif_out *spdif = snd_array_elem(&codec->spdif_out, idx);
+	struct hda_spdif_out *spdif;
 
+	mutex_lock(&codec->spdif_mutex);
+	spdif = snd_array_elem(&codec->spdif_out, idx);
 	ucontrol->value.iec958.status[0] = spdif->status & 0xff;
 	ucontrol->value.iec958.status[1] = (spdif->status >> 8) & 0xff;
 	ucontrol->value.iec958.status[2] = (spdif->status >> 16) & 0xff;
 	ucontrol->value.iec958.status[3] = (spdif->status >> 24) & 0xff;
+	mutex_unlock(&codec->spdif_mutex);
 
 	return 0;
 }
@@ -2950,12 +2987,14 @@ static int snd_hda_spdif_default_put(struct snd_kcontrol *kcontrol,
 {
 	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
 	int idx = kcontrol->private_value;
-	struct hda_spdif_out *spdif = snd_array_elem(&codec->spdif_out, idx);
-	hda_nid_t nid = spdif->nid;
+	struct hda_spdif_out *spdif;
+	hda_nid_t nid;
 	unsigned short val;
 	int change;
 
 	mutex_lock(&codec->spdif_mutex);
+	spdif = snd_array_elem(&codec->spdif_out, idx);
+	nid = spdif->nid;
 	spdif->status = ucontrol->value.iec958.status[0] |
 		((unsigned int)ucontrol->value.iec958.status[1] << 8) |
 		((unsigned int)ucontrol->value.iec958.status[2] << 16) |
@@ -2977,9 +3016,12 @@ static int snd_hda_spdif_out_switch_get(struct snd_kcontrol *kcontrol,
 {
 	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
 	int idx = kcontrol->private_value;
-	struct hda_spdif_out *spdif = snd_array_elem(&codec->spdif_out, idx);
+	struct hda_spdif_out *spdif;
 
+	mutex_lock(&codec->spdif_mutex);
+	spdif = snd_array_elem(&codec->spdif_out, idx);
 	ucontrol->value.integer.value[0] = spdif->ctls & AC_DIG1_ENABLE;
+	mutex_unlock(&codec->spdif_mutex);
 	return 0;
 }
 
@@ -2999,12 +3041,14 @@ static int snd_hda_spdif_out_switch_put(struct snd_kcontrol *kcontrol,
 {
 	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
 	int idx = kcontrol->private_value;
-	struct hda_spdif_out *spdif = snd_array_elem(&codec->spdif_out, idx);
-	hda_nid_t nid = spdif->nid;
+	struct hda_spdif_out *spdif;
+	hda_nid_t nid;
 	unsigned short val;
 	int change;
 
 	mutex_lock(&codec->spdif_mutex);
+	spdif = snd_array_elem(&codec->spdif_out, idx);
+	nid = spdif->nid;
 	val = spdif->ctls & ~AC_DIG1_ENABLE;
 	if (ucontrol->value.integer.value[0])
 		val |= AC_DIG1_ENABLE;
@@ -3092,6 +3136,9 @@ int snd_hda_create_spdif_out_ctls(struct hda_codec *codec,
 }
 EXPORT_SYMBOL_HDA(snd_hda_create_spdif_out_ctls);
 
+/* get the hda_spdif_out entry from the given NID
+ * call within spdif_mutex lock
+ */
 struct hda_spdif_out *snd_hda_spdif_out_of_nid(struct hda_codec *codec,
 					       hda_nid_t nid)
 {
@@ -3108,9 +3155,10 @@ EXPORT_SYMBOL_HDA(snd_hda_spdif_out_of_nid);
 
 void snd_hda_spdif_ctls_unassign(struct hda_codec *codec, int idx)
 {
-	struct hda_spdif_out *spdif = snd_array_elem(&codec->spdif_out, idx);
+	struct hda_spdif_out *spdif;
 
 	mutex_lock(&codec->spdif_mutex);
+	spdif = snd_array_elem(&codec->spdif_out, idx);
 	spdif->nid = (u16)-1;
 	mutex_unlock(&codec->spdif_mutex);
 }
@@ -3118,10 +3166,11 @@ EXPORT_SYMBOL_HDA(snd_hda_spdif_ctls_unassign);
 
 void snd_hda_spdif_ctls_assign(struct hda_codec *codec, int idx, hda_nid_t nid)
 {
-	struct hda_spdif_out *spdif = snd_array_elem(&codec->spdif_out, idx);
+	struct hda_spdif_out *spdif;
 	unsigned short val;
 
 	mutex_lock(&codec->spdif_mutex);
+	spdif = snd_array_elem(&codec->spdif_out, idx);
 	if (spdif->nid != nid) {
 		spdif->nid = nid;
 		val = spdif->ctls;
@@ -3486,11 +3535,14 @@ static void hda_call_codec_suspend(struct hda_codec *codec)
 			    codec->afg ? codec->afg : codec->mfg,
 			    AC_PWRST_D3);
 #ifdef CONFIG_SND_HDA_POWER_SAVE
-	snd_hda_update_power_acct(codec);
 	cancel_delayed_work(&codec->power_work);
+	spin_lock(&codec->power_lock);
+	snd_hda_update_power_acct(codec);
+	trace_hda_power_down(codec);
 	codec->power_on = 0;
 	codec->power_transition = 0;
 	codec->power_jiffies = jiffies;
+	spin_unlock(&codec->power_lock);
 #endif
 }
 
@@ -3499,6 +3551,10 @@ static void hda_call_codec_suspend(struct hda_codec *codec)
  */
 static void hda_call_codec_resume(struct hda_codec *codec)
 {
+	/* set as if powered on for avoiding re-entering the resume
+	 * in the resume / power-save sequence
+	 */
+	hda_keep_power_on(codec);
 	hda_set_power_state(codec,
 			    codec->afg ? codec->afg : codec->mfg,
 			    AC_PWRST_D0);
@@ -3514,6 +3570,7 @@ static void hda_call_codec_resume(struct hda_codec *codec)
 		snd_hda_codec_resume_amp(codec);
 		snd_hda_codec_resume_cache(codec);
 	}
+	snd_hda_power_down(codec); /* flag down before returning */
 }
 #endif /* CONFIG_PM */
 
@@ -3665,7 +3722,8 @@ unsigned int snd_hda_calc_stream_format(unsigned int rate,
 }
 EXPORT_SYMBOL_HDA(snd_hda_calc_stream_format);
 
-static unsigned int get_pcm_param(struct hda_codec *codec, hda_nid_t nid)
+static unsigned int get_pcm_param(struct hda_codec *codec, hda_nid_t nid,
+				  int dir)
 {
 	unsigned int val = 0;
 	if (nid != codec->afg &&
@@ -3680,11 +3738,12 @@ static unsigned int get_pcm_param(struct hda_codec *codec, hda_nid_t nid)
 
 static unsigned int query_pcm_param(struct hda_codec *codec, hda_nid_t nid)
 {
-	return query_caps_hash(codec, nid, HDA_HASH_PARPCM_KEY(nid),
+	return query_caps_hash(codec, nid, 0, HDA_HASH_PARPCM_KEY(nid),
 			       get_pcm_param);
 }
 
-static unsigned int get_stream_param(struct hda_codec *codec, hda_nid_t nid)
+static unsigned int get_stream_param(struct hda_codec *codec, hda_nid_t nid,
+				     int dir)
 {
 	unsigned int streams = snd_hda_param_read(codec, nid, AC_PAR_STREAM);
 	if (!streams || streams == -1)
@@ -3696,7 +3755,7 @@ static unsigned int get_stream_param(struct hda_codec *codec, hda_nid_t nid)
 
 static unsigned int query_stream_param(struct hda_codec *codec, hda_nid_t nid)
 {
-	return query_caps_hash(codec, nid, HDA_HASH_PARSTR_KEY(nid),
+	return query_caps_hash(codec, nid, 0, HDA_HASH_PARSTR_KEY(nid),
 			       get_stream_param);
 }
 
@@ -3775,11 +3834,13 @@ int snd_hda_query_supported_pcm(struct hda_codec *codec, hda_nid_t nid,
 					bps = 20;
 			}
 		}
+#if 0 /* FIXME: CS4206 doesn't work, which is the only codec supporting float */
 		if (streams & AC_SUPFMT_FLOAT32) {
 			formats |= SNDRV_PCM_FMTBIT_FLOAT_LE;
 			if (!bps)
 				bps = 32;
 		}
+#endif
 		if (streams == AC_SUPFMT_AC3) {
 			/* should be exclusive */
 			/* temporary hack: we have still no proper support
@@ -4283,12 +4344,18 @@ static void hda_power_work(struct work_struct *work)
 		container_of(work, struct hda_codec, power_work.work);
 	struct hda_bus *bus = codec->bus;
 
-	if (!codec->power_on || codec->power_count) {
-		codec->power_transition = 0;
+	spin_lock(&codec->power_lock);
+	if (codec->power_transition > 0) { /* during power-up sequence? */
+		spin_unlock(&codec->power_lock);
 		return;
 	}
+	if (!codec->power_on || codec->power_count) {
+		codec->power_transition = 0;
+		spin_unlock(&codec->power_lock);
+		return;
+	}
+	spin_unlock(&codec->power_lock);
 
-	trace_hda_power_down(codec);
 	hda_call_codec_suspend(codec);
 	if (bus->ops.pm_notify)
 		bus->ops.pm_notify(bus);
@@ -4296,9 +4363,11 @@ static void hda_power_work(struct work_struct *work)
 
 static void hda_keep_power_on(struct hda_codec *codec)
 {
+	spin_lock(&codec->power_lock);
 	codec->power_count++;
 	codec->power_on = 1;
 	codec->power_jiffies = jiffies;
+	spin_unlock(&codec->power_lock);
 }
 
 /* update the power on/off account with the current jiffies */
@@ -4323,19 +4392,31 @@ void snd_hda_power_up(struct hda_codec *codec)
 {
 	struct hda_bus *bus = codec->bus;
 
+	spin_lock(&codec->power_lock);
 	codec->power_count++;
-	if (codec->power_on || codec->power_transition)
+	if (codec->power_on || codec->power_transition > 0) {
+		spin_unlock(&codec->power_lock);
 		return;
+	}
+	spin_unlock(&codec->power_lock);
 
+	cancel_delayed_work_sync(&codec->power_work);
+
+	spin_lock(&codec->power_lock);
 	trace_hda_power_up(codec);
 	snd_hda_update_power_acct(codec);
 	codec->power_on = 1;
 	codec->power_jiffies = jiffies;
+	codec->power_transition = 1; /* avoid reentrance */
+	spin_unlock(&codec->power_lock);
+
 	if (bus->ops.pm_notify)
 		bus->ops.pm_notify(bus);
 	hda_call_codec_resume(codec);
-	cancel_delayed_work(&codec->power_work);
+
+	spin_lock(&codec->power_lock);
 	codec->power_transition = 0;
+	spin_unlock(&codec->power_lock);
 }
 EXPORT_SYMBOL_HDA(snd_hda_power_up);
 
@@ -4351,14 +4432,18 @@ EXPORT_SYMBOL_HDA(snd_hda_power_up);
  */
 void snd_hda_power_down(struct hda_codec *codec)
 {
+	spin_lock(&codec->power_lock);
 	--codec->power_count;
-	if (!codec->power_on || codec->power_count || codec->power_transition)
+	if (!codec->power_on || codec->power_count || codec->power_transition) {
+		spin_unlock(&codec->power_lock);
 		return;
+	}
 	if (power_save(codec)) {
-		codec->power_transition = 1; /* avoid reentrance */
+		codec->power_transition = -1; /* avoid reentrance */
 		queue_delayed_work(codec->bus->workq, &codec->power_work,
 				msecs_to_jiffies(power_save(codec) * 1000));
 	}
+	spin_unlock(&codec->power_lock);
 }
 EXPORT_SYMBOL_HDA(snd_hda_power_down);
 
@@ -4710,11 +4795,11 @@ int snd_hda_multi_out_analog_prepare(struct hda_codec *codec,
 {
 	const hda_nid_t *nids = mout->dac_nids;
 	int chs = substream->runtime->channels;
-	struct hda_spdif_out *spdif =
-			snd_hda_spdif_out_of_nid(codec, mout->dig_out_nid);
+	struct hda_spdif_out *spdif;
 	int i;
 
 	mutex_lock(&codec->spdif_mutex);
+	spdif = snd_hda_spdif_out_of_nid(codec, mout->dig_out_nid);
 	if (mout->dig_out_nid && mout->share_spdif &&
 	    mout->dig_out_used != HDA_DIG_EXCLUSIVE) {
 		if (chs == 2 &&
@@ -4795,601 +4880,58 @@ int snd_hda_multi_out_analog_cleanup(struct hda_codec *codec,
 }
 EXPORT_SYMBOL_HDA(snd_hda_multi_out_analog_cleanup);
 
-/*
- * Helper for automatic pin configuration
- */
-
-static int is_in_nid_list(hda_nid_t nid, const hda_nid_t *list)
-{
-	for (; *list; list++)
-		if (*list == nid)
-			return 1;
-	return 0;
-}
-
-
-/*
- * Sort an associated group of pins according to their sequence numbers.
- */
-static void sort_pins_by_sequence(hda_nid_t *pins, short *sequences,
-				  int num_pins)
-{
-	int i, j;
-	short seq;
-	hda_nid_t nid;
-
-	for (i = 0; i < num_pins; i++) {
-		for (j = i + 1; j < num_pins; j++) {
-			if (sequences[i] > sequences[j]) {
-				seq = sequences[i];
-				sequences[i] = sequences[j];
-				sequences[j] = seq;
-				nid = pins[i];
-				pins[i] = pins[j];
-				pins[j] = nid;
-			}
-		}
-	}
-}
-
-
-/* add the found input-pin to the cfg->inputs[] table */
-static void add_auto_cfg_input_pin(struct auto_pin_cfg *cfg, hda_nid_t nid,
-				   int type)
-{
-	if (cfg->num_inputs < AUTO_CFG_MAX_INS) {
-		cfg->inputs[cfg->num_inputs].pin = nid;
-		cfg->inputs[cfg->num_inputs].type = type;
-		cfg->num_inputs++;
-	}
-}
-
-/* sort inputs in the order of AUTO_PIN_* type */
-static void sort_autocfg_input_pins(struct auto_pin_cfg *cfg)
-{
-	int i, j;
-
-	for (i = 0; i < cfg->num_inputs; i++) {
-		for (j = i + 1; j < cfg->num_inputs; j++) {
-			if (cfg->inputs[i].type > cfg->inputs[j].type) {
-				struct auto_pin_cfg_item tmp;
-				tmp = cfg->inputs[i];
-				cfg->inputs[i] = cfg->inputs[j];
-				cfg->inputs[j] = tmp;
-			}
-		}
-	}
-}
-
-/* Reorder the surround channels
- * ALSA sequence is front/surr/clfe/side
- * HDA sequence is:
- *    4-ch: front/surr  =>  OK as it is
- *    6-ch: front/clfe/surr
- *    8-ch: front/clfe/rear/side|fc
- */
-static void reorder_outputs(unsigned int nums, hda_nid_t *pins)
-{
-	hda_nid_t nid;
-
-	switch (nums) {
-	case 3:
-	case 4:
-		nid = pins[1];
-		pins[1] = pins[2];
-		pins[2] = nid;
-		break;
-	}
-}
-
-/*
- * Parse all pin widgets and store the useful pin nids to cfg
- *
- * The number of line-outs or any primary output is stored in line_outs,
- * and the corresponding output pins are assigned to line_out_pins[],
- * in the order of front, rear, CLFE, side, ...
- *
- * If more extra outputs (speaker and headphone) are found, the pins are
- * assisnged to hp_pins[] and speaker_pins[], respectively.  If no line-out jack
- * is detected, one of speaker of HP pins is assigned as the primary
- * output, i.e. to line_out_pins[0].  So, line_outs is always positive
- * if any analog output exists.
- *
- * The analog input pins are assigned to inputs array.
- * The digital input/output pins are assigned to dig_in_pin and dig_out_pin,
- * respectively.
- */
-int snd_hda_parse_pin_defcfg(struct hda_codec *codec,
-			     struct auto_pin_cfg *cfg,
-			     const hda_nid_t *ignore_nids,
-			     unsigned int cond_flags)
-{
-	hda_nid_t nid, end_nid;
-	short seq, assoc_line_out;
-	short sequences_line_out[ARRAY_SIZE(cfg->line_out_pins)];
-	short sequences_speaker[ARRAY_SIZE(cfg->speaker_pins)];
-	short sequences_hp[ARRAY_SIZE(cfg->hp_pins)];
-	int i;
-
-	memset(cfg, 0, sizeof(*cfg));
-
-	memset(sequences_line_out, 0, sizeof(sequences_line_out));
-	memset(sequences_speaker, 0, sizeof(sequences_speaker));
-	memset(sequences_hp, 0, sizeof(sequences_hp));
-	assoc_line_out = 0;
-
-	codec->ignore_misc_bit = true;
-	end_nid = codec->start_nid + codec->num_nodes;
-	for (nid = codec->start_nid; nid < end_nid; nid++) {
-		unsigned int wid_caps = get_wcaps(codec, nid);
-		unsigned int wid_type = get_wcaps_type(wid_caps);
-		unsigned int def_conf;
-		short assoc, loc, conn, dev;
-
-		/* read all default configuration for pin complex */
-		if (wid_type != AC_WID_PIN)
-			continue;
-		/* ignore the given nids (e.g. pc-beep returns error) */
-		if (ignore_nids && is_in_nid_list(nid, ignore_nids))
-			continue;
-
-		def_conf = snd_hda_codec_get_pincfg(codec, nid);
-		if (!(get_defcfg_misc(snd_hda_codec_get_pincfg(codec, nid)) &
-		      AC_DEFCFG_MISC_NO_PRESENCE))
-			codec->ignore_misc_bit = false;
-		conn = get_defcfg_connect(def_conf);
-		if (conn == AC_JACK_PORT_NONE)
-			continue;
-		loc = get_defcfg_location(def_conf);
-		dev = get_defcfg_device(def_conf);
-
-		/* workaround for buggy BIOS setups */
-		if (dev == AC_JACK_LINE_OUT) {
-			if (conn == AC_JACK_PORT_FIXED)
-				dev = AC_JACK_SPEAKER;
-		}
-
-		switch (dev) {
-		case AC_JACK_LINE_OUT:
-			seq = get_defcfg_sequence(def_conf);
-			assoc = get_defcfg_association(def_conf);
-
-			if (!(wid_caps & AC_WCAP_STEREO))
-				if (!cfg->mono_out_pin)
-					cfg->mono_out_pin = nid;
-			if (!assoc)
-				continue;
-			if (!assoc_line_out)
-				assoc_line_out = assoc;
-			else if (assoc_line_out != assoc)
-				continue;
-			if (cfg->line_outs >= ARRAY_SIZE(cfg->line_out_pins))
-				continue;
-			cfg->line_out_pins[cfg->line_outs] = nid;
-			sequences_line_out[cfg->line_outs] = seq;
-			cfg->line_outs++;
-			break;
-		case AC_JACK_SPEAKER:
-			seq = get_defcfg_sequence(def_conf);
-			assoc = get_defcfg_association(def_conf);
-			if (cfg->speaker_outs >= ARRAY_SIZE(cfg->speaker_pins))
-				continue;
-			cfg->speaker_pins[cfg->speaker_outs] = nid;
-			sequences_speaker[cfg->speaker_outs] = (assoc << 4) | seq;
-			cfg->speaker_outs++;
-			break;
-		case AC_JACK_HP_OUT:
-			seq = get_defcfg_sequence(def_conf);
-			assoc = get_defcfg_association(def_conf);
-			if (cfg->hp_outs >= ARRAY_SIZE(cfg->hp_pins))
-				continue;
-			cfg->hp_pins[cfg->hp_outs] = nid;
-			sequences_hp[cfg->hp_outs] = (assoc << 4) | seq;
-			cfg->hp_outs++;
-			break;
-		case AC_JACK_MIC_IN:
-			add_auto_cfg_input_pin(cfg, nid, AUTO_PIN_MIC);
-			break;
-		case AC_JACK_LINE_IN:
-			add_auto_cfg_input_pin(cfg, nid, AUTO_PIN_LINE_IN);
-			break;
-		case AC_JACK_CD:
-			add_auto_cfg_input_pin(cfg, nid, AUTO_PIN_CD);
-			break;
-		case AC_JACK_AUX:
-			add_auto_cfg_input_pin(cfg, nid, AUTO_PIN_AUX);
-			break;
-		case AC_JACK_SPDIF_OUT:
-		case AC_JACK_DIG_OTHER_OUT:
-			if (cfg->dig_outs >= ARRAY_SIZE(cfg->dig_out_pins))
-				continue;
-			cfg->dig_out_pins[cfg->dig_outs] = nid;
-			cfg->dig_out_type[cfg->dig_outs] =
-				(loc == AC_JACK_LOC_HDMI) ?
-				HDA_PCM_TYPE_HDMI : HDA_PCM_TYPE_SPDIF;
-			cfg->dig_outs++;
-			break;
-		case AC_JACK_SPDIF_IN:
-		case AC_JACK_DIG_OTHER_IN:
-			cfg->dig_in_pin = nid;
-			if (loc == AC_JACK_LOC_HDMI)
-				cfg->dig_in_type = HDA_PCM_TYPE_HDMI;
-			else
-				cfg->dig_in_type = HDA_PCM_TYPE_SPDIF;
-			break;
-		}
-	}
-
-	/* FIX-UP:
-	 * If no line-out is defined but multiple HPs are found,
-	 * some of them might be the real line-outs.
-	 */
-	if (!cfg->line_outs && cfg->hp_outs > 1 &&
-	    !(cond_flags & HDA_PINCFG_NO_HP_FIXUP)) {
-		int i = 0;
-		while (i < cfg->hp_outs) {
-			/* The real HPs should have the sequence 0x0f */
-			if ((sequences_hp[i] & 0x0f) == 0x0f) {
-				i++;
-				continue;
-			}
-			/* Move it to the line-out table */
-			cfg->line_out_pins[cfg->line_outs] = cfg->hp_pins[i];
-			sequences_line_out[cfg->line_outs] = sequences_hp[i];
-			cfg->line_outs++;
-			cfg->hp_outs--;
-			memmove(cfg->hp_pins + i, cfg->hp_pins + i + 1,
-				sizeof(cfg->hp_pins[0]) * (cfg->hp_outs - i));
-			memmove(sequences_hp + i, sequences_hp + i + 1,
-				sizeof(sequences_hp[0]) * (cfg->hp_outs - i));
-		}
-		memset(cfg->hp_pins + cfg->hp_outs, 0,
-		       sizeof(hda_nid_t) * (AUTO_CFG_MAX_OUTS - cfg->hp_outs));
-		if (!cfg->hp_outs)
-			cfg->line_out_type = AUTO_PIN_HP_OUT;
-
-	}
-
-	/* sort by sequence */
-	sort_pins_by_sequence(cfg->line_out_pins, sequences_line_out,
-			      cfg->line_outs);
-	sort_pins_by_sequence(cfg->speaker_pins, sequences_speaker,
-			      cfg->speaker_outs);
-	sort_pins_by_sequence(cfg->hp_pins, sequences_hp,
-			      cfg->hp_outs);
-
-	/*
-	 * FIX-UP: if no line-outs are detected, try to use speaker or HP pin
-	 * as a primary output
-	 */
-	if (!cfg->line_outs &&
-	    !(cond_flags & HDA_PINCFG_NO_LO_FIXUP)) {
-		if (cfg->speaker_outs) {
-			cfg->line_outs = cfg->speaker_outs;
-			memcpy(cfg->line_out_pins, cfg->speaker_pins,
-			       sizeof(cfg->speaker_pins));
-			cfg->speaker_outs = 0;
-			memset(cfg->speaker_pins, 0, sizeof(cfg->speaker_pins));
-			cfg->line_out_type = AUTO_PIN_SPEAKER_OUT;
-		} else if (cfg->hp_outs) {
-			cfg->line_outs = cfg->hp_outs;
-			memcpy(cfg->line_out_pins, cfg->hp_pins,
-			       sizeof(cfg->hp_pins));
-			cfg->hp_outs = 0;
-			memset(cfg->hp_pins, 0, sizeof(cfg->hp_pins));
-			cfg->line_out_type = AUTO_PIN_HP_OUT;
-		}
-	}
-
-	reorder_outputs(cfg->line_outs, cfg->line_out_pins);
-	reorder_outputs(cfg->hp_outs, cfg->hp_pins);
-	reorder_outputs(cfg->speaker_outs, cfg->speaker_pins);
-
-	sort_autocfg_input_pins(cfg);
-
-	/*
-	 * debug prints of the parsed results
-	 */
-	snd_printd("autoconfig: line_outs=%d (0x%x/0x%x/0x%x/0x%x/0x%x) type:%s\n",
-		   cfg->line_outs, cfg->line_out_pins[0], cfg->line_out_pins[1],
-		   cfg->line_out_pins[2], cfg->line_out_pins[3],
-		   cfg->line_out_pins[4],
-		   cfg->line_out_type == AUTO_PIN_HP_OUT ? "hp" :
-		   (cfg->line_out_type == AUTO_PIN_SPEAKER_OUT ?
-		    "speaker" : "line"));
-	snd_printd("   speaker_outs=%d (0x%x/0x%x/0x%x/0x%x/0x%x)\n",
-		   cfg->speaker_outs, cfg->speaker_pins[0],
-		   cfg->speaker_pins[1], cfg->speaker_pins[2],
-		   cfg->speaker_pins[3], cfg->speaker_pins[4]);
-	snd_printd("   hp_outs=%d (0x%x/0x%x/0x%x/0x%x/0x%x)\n",
-		   cfg->hp_outs, cfg->hp_pins[0],
-		   cfg->hp_pins[1], cfg->hp_pins[2],
-		   cfg->hp_pins[3], cfg->hp_pins[4]);
-	snd_printd("   mono: mono_out=0x%x\n", cfg->mono_out_pin);
-	if (cfg->dig_outs)
-		snd_printd("   dig-out=0x%x/0x%x\n",
-			   cfg->dig_out_pins[0], cfg->dig_out_pins[1]);
-	snd_printd("   inputs:");
-	for (i = 0; i < cfg->num_inputs; i++) {
-		snd_printd(" %s=0x%x",
-			    hda_get_autocfg_input_label(codec, cfg, i),
-			    cfg->inputs[i].pin);
-	}
-	snd_printd("\n");
-	if (cfg->dig_in_pin)
-		snd_printd("   dig-in=0x%x\n", cfg->dig_in_pin);
-
-	return 0;
-}
-EXPORT_SYMBOL_HDA(snd_hda_parse_pin_defcfg);
-
-int snd_hda_get_input_pin_attr(unsigned int def_conf)
-{
-	unsigned int loc = get_defcfg_location(def_conf);
-	unsigned int conn = get_defcfg_connect(def_conf);
-	if (conn == AC_JACK_PORT_NONE)
-		return INPUT_PIN_ATTR_UNUSED;
-	/* Windows may claim the internal mic to be BOTH, too */
-	if (conn == AC_JACK_PORT_FIXED || conn == AC_JACK_PORT_BOTH)
-		return INPUT_PIN_ATTR_INT;
-	if ((loc & 0x30) == AC_JACK_LOC_INTERNAL)
-		return INPUT_PIN_ATTR_INT;
-	if ((loc & 0x30) == AC_JACK_LOC_SEPARATE)
-		return INPUT_PIN_ATTR_DOCK;
-	if (loc == AC_JACK_LOC_REAR)
-		return INPUT_PIN_ATTR_REAR;
-	if (loc == AC_JACK_LOC_FRONT)
-		return INPUT_PIN_ATTR_FRONT;
-	return INPUT_PIN_ATTR_NORMAL;
-}
-EXPORT_SYMBOL_HDA(snd_hda_get_input_pin_attr);
-
 /**
- * hda_get_input_pin_label - Give a label for the given input pin
+ * snd_hda_get_default_vref - Get the default (mic) VREF pin bits
  *
- * When check_location is true, the function checks the pin location
- * for mic and line-in pins, and set an appropriate prefix like "Front",
- * "Rear", "Internal".
+ * Guess the suitable VREF pin bits to be set as the pin-control value.
+ * Note: the function doesn't set the AC_PINCTL_IN_EN bit.
  */
-
-static const char *hda_get_input_pin_label(struct hda_codec *codec,
-					   hda_nid_t pin, bool check_location)
+unsigned int snd_hda_get_default_vref(struct hda_codec *codec, hda_nid_t pin)
 {
-	unsigned int def_conf;
-	static const char * const mic_names[] = {
-		"Internal Mic", "Dock Mic", "Mic", "Front Mic", "Rear Mic",
-	};
-	int attr;
-
-	def_conf = snd_hda_codec_get_pincfg(codec, pin);
-
-	switch (get_defcfg_device(def_conf)) {
-	case AC_JACK_MIC_IN:
-		if (!check_location)
-			return "Mic";
-		attr = snd_hda_get_input_pin_attr(def_conf);
-		if (!attr)
-			return "None";
-		return mic_names[attr - 1];
-	case AC_JACK_LINE_IN:
-		if (!check_location)
-			return "Line";
-		attr = snd_hda_get_input_pin_attr(def_conf);
-		if (!attr)
-			return "None";
-		if (attr == INPUT_PIN_ATTR_DOCK)
-			return "Dock Line";
-		return "Line";
-	case AC_JACK_AUX:
-		return "Aux";
-	case AC_JACK_CD:
-		return "CD";
-	case AC_JACK_SPDIF_IN:
-		return "SPDIF In";
-	case AC_JACK_DIG_OTHER_IN:
-		return "Digital In";
-	default:
-		return "Misc";
-	}
+	unsigned int pincap;
+	unsigned int oldval;
+	oldval = snd_hda_codec_read(codec, pin, 0,
+				    AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
+	pincap = snd_hda_query_pin_caps(codec, pin);
+	pincap = (pincap & AC_PINCAP_VREF) >> AC_PINCAP_VREF_SHIFT;
+	/* Exception: if the default pin setup is vref50, we give it priority */
+	if ((pincap & AC_PINCAP_VREF_80) && oldval != PIN_VREF50)
+		return AC_PINCTL_VREF_80;
+	else if (pincap & AC_PINCAP_VREF_50)
+		return AC_PINCTL_VREF_50;
+	else if (pincap & AC_PINCAP_VREF_100)
+		return AC_PINCTL_VREF_100;
+	else if (pincap & AC_PINCAP_VREF_GRD)
+		return AC_PINCTL_VREF_GRD;
+	return AC_PINCTL_VREF_HIZ;
 }
+EXPORT_SYMBOL_HDA(snd_hda_get_default_vref);
 
-/* Check whether the location prefix needs to be added to the label.
- * If all mic-jacks are in the same location (e.g. rear panel), we don't
- * have to put "Front" prefix to each label.  In such a case, returns false.
- */
-static int check_mic_location_need(struct hda_codec *codec,
-				   const struct auto_pin_cfg *cfg,
-				   int input)
+int _snd_hda_set_pin_ctl(struct hda_codec *codec, hda_nid_t pin,
+			 unsigned int val, bool cached)
 {
-	unsigned int defc;
-	int i, attr, attr2;
-
-	defc = snd_hda_codec_get_pincfg(codec, cfg->inputs[input].pin);
-	attr = snd_hda_get_input_pin_attr(defc);
-	/* for internal or docking mics, we need locations */
-	if (attr <= INPUT_PIN_ATTR_NORMAL)
-		return 1;
-
-	attr = 0;
-	for (i = 0; i < cfg->num_inputs; i++) {
-		defc = snd_hda_codec_get_pincfg(codec, cfg->inputs[i].pin);
-		attr2 = snd_hda_get_input_pin_attr(defc);
-		if (attr2 >= INPUT_PIN_ATTR_NORMAL) {
-			if (attr && attr != attr2)
-				return 1; /* different locations found */
-			attr = attr2;
+	if (val) {
+		unsigned int cap = snd_hda_query_pin_caps(codec, pin);
+		if (cap && (val & AC_PINCTL_OUT_EN)) {
+			if (!(cap & AC_PINCAP_OUT))
+				val &= ~(AC_PINCTL_OUT_EN | AC_PINCTL_HP_EN);
+			else if ((val & AC_PINCTL_HP_EN) &&
+				 !(cap & AC_PINCAP_HP_DRV))
+				val &= ~AC_PINCTL_HP_EN;
+		}
+		if (cap && (val & AC_PINCTL_IN_EN)) {
+			if (!(cap & AC_PINCAP_IN))
+				val &= ~(AC_PINCTL_IN_EN | AC_PINCTL_VREFEN);
 		}
 	}
-	return 0;
+	if (cached)
+		return snd_hda_codec_update_cache(codec, pin, 0,
+				AC_VERB_SET_PIN_WIDGET_CONTROL, val);
+	else
+		return snd_hda_codec_write(codec, pin, 0,
+					   AC_VERB_SET_PIN_WIDGET_CONTROL, val);
 }
-
-/**
- * hda_get_autocfg_input_label - Get a label for the given input
- *
- * Get a label for the given input pin defined by the autocfg item.
- * Unlike hda_get_input_pin_label(), this function checks all inputs
- * defined in autocfg and avoids the redundant mic/line prefix as much as
- * possible.
- */
-const char *hda_get_autocfg_input_label(struct hda_codec *codec,
-					const struct auto_pin_cfg *cfg,
-					int input)
-{
-	int type = cfg->inputs[input].type;
-	int has_multiple_pins = 0;
-
-	if ((input > 0 && cfg->inputs[input - 1].type == type) ||
-	    (input < cfg->num_inputs - 1 && cfg->inputs[input + 1].type == type))
-		has_multiple_pins = 1;
-	if (has_multiple_pins && type == AUTO_PIN_MIC)
-		has_multiple_pins &= check_mic_location_need(codec, cfg, input);
-	return hda_get_input_pin_label(codec, cfg->inputs[input].pin,
-				       has_multiple_pins);
-}
-EXPORT_SYMBOL_HDA(hda_get_autocfg_input_label);
-
-/* return the position of NID in the list, or -1 if not found */
-static int find_idx_in_nid_list(hda_nid_t nid, const hda_nid_t *list, int nums)
-{
-	int i;
-	for (i = 0; i < nums; i++)
-		if (list[i] == nid)
-			return i;
-	return -1;
-}
-
-/* get a unique suffix or an index number */
-static const char *check_output_sfx(hda_nid_t nid, const hda_nid_t *pins,
-				    int num_pins, int *indexp)
-{
-	static const char * const channel_sfx[] = {
-		" Front", " Surround", " CLFE", " Side"
-	};
-	int i;
-
-	i = find_idx_in_nid_list(nid, pins, num_pins);
-	if (i < 0)
-		return NULL;
-	if (num_pins == 1)
-		return "";
-	if (num_pins > ARRAY_SIZE(channel_sfx)) {
-		if (indexp)
-			*indexp = i;
-		return "";
-	}
-	return channel_sfx[i];
-}
-
-static int fill_audio_out_name(struct hda_codec *codec, hda_nid_t nid,
-			       const struct auto_pin_cfg *cfg,
-			       const char *name, char *label, int maxlen,
-			       int *indexp)
-{
-	unsigned int def_conf = snd_hda_codec_get_pincfg(codec, nid);
-	int attr = snd_hda_get_input_pin_attr(def_conf);
-	const char *pfx = "", *sfx = "";
-
-	/* handle as a speaker if it's a fixed line-out */
-	if (!strcmp(name, "Line Out") && attr == INPUT_PIN_ATTR_INT)
-		name = "Speaker";
-	/* check the location */
-	switch (attr) {
-	case INPUT_PIN_ATTR_DOCK:
-		pfx = "Dock ";
-		break;
-	case INPUT_PIN_ATTR_FRONT:
-		pfx = "Front ";
-		break;
-	}
-	if (cfg) {
-		/* try to give a unique suffix if needed */
-		sfx = check_output_sfx(nid, cfg->line_out_pins, cfg->line_outs,
-				       indexp);
-		if (!sfx)
-			sfx = check_output_sfx(nid, cfg->speaker_pins, cfg->speaker_outs,
-					       indexp);
-		if (!sfx) {
-			/* don't add channel suffix for Headphone controls */
-			int idx = find_idx_in_nid_list(nid, cfg->hp_pins,
-						       cfg->hp_outs);
-			if (idx >= 0)
-				*indexp = idx;
-			sfx = "";
-		}
-	}
-	snprintf(label, maxlen, "%s%s%s", pfx, name, sfx);
-	return 1;
-}
-
-/**
- * snd_hda_get_pin_label - Get a label for the given I/O pin
- *
- * Get a label for the given pin.  This function works for both input and
- * output pins.  When @cfg is given as non-NULL, the function tries to get
- * an optimized label using hda_get_autocfg_input_label().
- *
- * This function tries to give a unique label string for the pin as much as
- * possible.  For example, when the multiple line-outs are present, it adds
- * the channel suffix like "Front", "Surround", etc (only when @cfg is given).
- * If no unique name with a suffix is available and @indexp is non-NULL, the
- * index number is stored in the pointer.
- */
-int snd_hda_get_pin_label(struct hda_codec *codec, hda_nid_t nid,
-			  const struct auto_pin_cfg *cfg,
-			  char *label, int maxlen, int *indexp)
-{
-	unsigned int def_conf = snd_hda_codec_get_pincfg(codec, nid);
-	const char *name = NULL;
-	int i;
-
-	if (indexp)
-		*indexp = 0;
-	if (get_defcfg_connect(def_conf) == AC_JACK_PORT_NONE)
-		return 0;
-
-	switch (get_defcfg_device(def_conf)) {
-	case AC_JACK_LINE_OUT:
-		return fill_audio_out_name(codec, nid, cfg, "Line Out",
-					   label, maxlen, indexp);
-	case AC_JACK_SPEAKER:
-		return fill_audio_out_name(codec, nid, cfg, "Speaker",
-					   label, maxlen, indexp);
-	case AC_JACK_HP_OUT:
-		return fill_audio_out_name(codec, nid, cfg, "Headphone",
-					   label, maxlen, indexp);
-	case AC_JACK_SPDIF_OUT:
-	case AC_JACK_DIG_OTHER_OUT:
-		if (get_defcfg_location(def_conf) == AC_JACK_LOC_HDMI)
-			name = "HDMI";
-		else
-			name = "SPDIF";
-		if (cfg && indexp) {
-			i = find_idx_in_nid_list(nid, cfg->dig_out_pins,
-						 cfg->dig_outs);
-			if (i >= 0)
-				*indexp = i;
-		}
-		break;
-	default:
-		if (cfg) {
-			for (i = 0; i < cfg->num_inputs; i++) {
-				if (cfg->inputs[i].pin != nid)
-					continue;
-				name = hda_get_autocfg_input_label(codec, cfg, i);
-				if (name)
-					break;
-			}
-		}
-		if (!name)
-			name = hda_get_input_pin_label(codec, nid, true);
-		break;
-	}
-	if (!name)
-		return 0;
-	strlcpy(label, name, maxlen);
-	return 1;
-}
-EXPORT_SYMBOL_HDA(snd_hda_get_pin_label);
+EXPORT_SYMBOL_HDA(_snd_hda_set_pin_ctl);
 
 /**
  * snd_hda_add_imux_item - Add an item to input_mux
@@ -5444,8 +4986,6 @@ int snd_hda_suspend(struct hda_bus *bus)
 	list_for_each_entry(codec, &bus->codec_list, list) {
 		if (hda_codec_is_power_on(codec))
 			hda_call_codec_suspend(codec);
-		if (codec->patch_ops.post_suspend)
-			codec->patch_ops.post_suspend(codec);
 	}
 	return 0;
 }
@@ -5465,10 +5005,7 @@ int snd_hda_resume(struct hda_bus *bus)
 	struct hda_codec *codec;
 
 	list_for_each_entry(codec, &bus->codec_list, list) {
-		if (codec->patch_ops.pre_resume)
-			codec->patch_ops.pre_resume(codec);
-		if (snd_hda_codec_needs_resume(codec))
-			hda_call_codec_resume(codec);
+		hda_call_codec_resume(codec);
 	}
 	return 0;
 }
diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h
index 56b4f74c0b13..54b52819fb47 100644
--- a/sound/pci/hda/hda_codec.h
+++ b/sound/pci/hda/hda_codec.h
@@ -704,8 +704,6 @@ struct hda_codec_ops {
 				unsigned int power_state);
 #ifdef CONFIG_PM
 	int (*suspend)(struct hda_codec *codec, pm_message_t state);
-	int (*post_suspend)(struct hda_codec *codec);
-	int (*pre_resume)(struct hda_codec *codec);
 	int (*resume)(struct hda_codec *codec);
 #endif
 #ifdef CONFIG_SND_HDA_POWER_SAVE
@@ -829,6 +827,7 @@ struct hda_codec {
 
 	struct mutex spdif_mutex;
 	struct mutex control_mutex;
+	struct mutex hash_mutex;
 	struct snd_array spdif_out;
 	unsigned int spdif_in_enable;	/* SPDIF input enable? */
 	const hda_nid_t *slave_dig_outs; /* optional digital out slave widgets */
@@ -861,12 +860,13 @@ struct hda_codec {
 	unsigned int no_jack_detect:1;	/* Machine has no jack-detection */
 #ifdef CONFIG_SND_HDA_POWER_SAVE
 	unsigned int power_on :1;	/* current (global) power-state */
-	unsigned int power_transition :1; /* power-state in transition */
+	int power_transition;	/* power-state in transition */
 	int power_count;	/* current (global) power refcount */
 	struct delayed_work power_work; /* delayed task for powerdown */
 	unsigned long power_on_acct;
 	unsigned long power_off_acct;
 	unsigned long power_jiffies;
+	spinlock_t power_lock;
 #endif
 
 	/* codec-specific additional proc output */
@@ -911,10 +911,13 @@ int snd_hda_get_sub_nodes(struct hda_codec *codec, hda_nid_t nid,
 			  hda_nid_t *start_id);
 int snd_hda_get_connections(struct hda_codec *codec, hda_nid_t nid,
 			    hda_nid_t *conn_list, int max_conns);
+static inline int
+snd_hda_get_num_conns(struct hda_codec *codec, hda_nid_t nid)
+{
+	return snd_hda_get_connections(codec, nid, NULL, 0);
+}
 int snd_hda_get_raw_connections(struct hda_codec *codec, hda_nid_t nid,
 			    hda_nid_t *conn_list, int max_conns);
-int snd_hda_get_conn_list(struct hda_codec *codec, hda_nid_t nid,
-			  const hda_nid_t **listp);
 int snd_hda_override_conn_list(struct hda_codec *codec, hda_nid_t nid, int nums,
 			  const hda_nid_t *list);
 int snd_hda_get_conn_index(struct hda_codec *codec, hda_nid_t mux,
@@ -1051,12 +1054,10 @@ const char *snd_hda_get_jack_location(u32 cfg);
 #ifdef CONFIG_SND_HDA_POWER_SAVE
 void snd_hda_power_up(struct hda_codec *codec);
 void snd_hda_power_down(struct hda_codec *codec);
-#define snd_hda_codec_needs_resume(codec) codec->power_count
 void snd_hda_update_power_acct(struct hda_codec *codec);
 #else
 static inline void snd_hda_power_up(struct hda_codec *codec) {}
 static inline void snd_hda_power_down(struct hda_codec *codec) {}
-#define snd_hda_codec_needs_resume(codec) 1
 #endif
 
 #ifdef CONFIG_SND_HDA_PATCH_LOADER
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 1f350522bed4..4ab8102f87ea 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -497,6 +497,7 @@ enum {
 	AZX_DRIVER_NVIDIA,
 	AZX_DRIVER_TERA,
 	AZX_DRIVER_CTX,
+	AZX_DRIVER_CTHDA,
 	AZX_DRIVER_GENERIC,
 	AZX_NUM_DRIVERS, /* keep this as last entry */
 };
@@ -518,6 +519,7 @@ enum {
 #define AZX_DCAPS_OLD_SSYNC	(1 << 20)	/* Old SSYNC reg for ICH */
 #define AZX_DCAPS_BUFSIZE	(1 << 21)	/* no buffer size alignment */
 #define AZX_DCAPS_ALIGN_BUFSIZE	(1 << 22)	/* buffer size alignment */
+#define AZX_DCAPS_4K_BDLE_BOUNDARY (1 << 23)	/* BDLE in 4k boundary */
 
 /* quirks for ATI SB / AMD Hudson */
 #define AZX_DCAPS_PRESET_ATI_SB \
@@ -533,6 +535,9 @@ enum {
 	(AZX_DCAPS_NVIDIA_SNOOP | AZX_DCAPS_RIRB_DELAY | AZX_DCAPS_NO_MSI |\
 	 AZX_DCAPS_ALIGN_BUFSIZE)
 
+#define AZX_DCAPS_PRESET_CTHDA \
+	(AZX_DCAPS_NO_MSI | AZX_DCAPS_POSFIX_LPIB | AZX_DCAPS_4K_BDLE_BOUNDARY)
+
 static char *driver_short_names[] __devinitdata = {
 	[AZX_DRIVER_ICH] = "HDA Intel",
 	[AZX_DRIVER_PCH] = "HDA Intel PCH",
@@ -546,6 +551,7 @@ static char *driver_short_names[] __devinitdata = {
 	[AZX_DRIVER_NVIDIA] = "HDA NVidia",
 	[AZX_DRIVER_TERA] = "HDA Teradici", 
 	[AZX_DRIVER_CTX] = "HDA Creative", 
+	[AZX_DRIVER_CTHDA] = "HDA Creative",
 	[AZX_DRIVER_GENERIC] = "HD-Audio Generic",
 };
 
@@ -1285,7 +1291,8 @@ static irqreturn_t azx_interrupt(int irq, void *dev_id)
 /*
  * set up a BDL entry
  */
-static int setup_bdle(struct snd_pcm_substream *substream,
+static int setup_bdle(struct azx *chip,
+		      struct snd_pcm_substream *substream,
 		      struct azx_dev *azx_dev, u32 **bdlp,
 		      int ofs, int size, int with_ioc)
 {
@@ -1304,6 +1311,12 @@ static int setup_bdle(struct snd_pcm_substream *substream,
 		bdl[1] = cpu_to_le32(upper_32_bits(addr));
 		/* program the size field of the BDL entry */
 		chunk = snd_pcm_sgbuf_get_chunk_size(substream, ofs, size);
+		/* one BDLE cannot cross 4K boundary on CTHDA chips */
+		if (chip->driver_caps & AZX_DCAPS_4K_BDLE_BOUNDARY) {
+			u32 remain = 0x1000 - (ofs & 0xfff);
+			if (chunk > remain)
+				chunk = remain;
+		}
 		bdl[2] = cpu_to_le32(chunk);
 		/* program the IOC to enable interrupt
 		 * only when the whole fragment is processed
@@ -1356,7 +1369,7 @@ static int azx_setup_periods(struct azx *chip,
 				   bdl_pos_adj[chip->dev_index]);
 			pos_adj = 0;
 		} else {
-			ofs = setup_bdle(substream, azx_dev,
+			ofs = setup_bdle(chip, substream, azx_dev,
 					 &bdl, ofs, pos_adj,
 					 !substream->runtime->no_period_wakeup);
 			if (ofs < 0)
@@ -1366,10 +1379,10 @@ static int azx_setup_periods(struct azx *chip,
 		pos_adj = 0;
 	for (i = 0; i < periods; i++) {
 		if (i == periods - 1 && pos_adj)
-			ofs = setup_bdle(substream, azx_dev, &bdl, ofs,
+			ofs = setup_bdle(chip, substream, azx_dev, &bdl, ofs,
 					 period_bytes - pos_adj, 0);
 		else
-			ofs = setup_bdle(substream, azx_dev, &bdl, ofs,
+			ofs = setup_bdle(chip, substream, azx_dev, &bdl, ofs,
 					 period_bytes,
 					 !substream->runtime->no_period_wakeup);
 		if (ofs < 0)
@@ -2353,17 +2366,6 @@ static void azx_power_notify(struct hda_bus *bus)
  * power management
  */
 
-static int snd_hda_codecs_inuse(struct hda_bus *bus)
-{
-	struct hda_codec *codec;
-
-	list_for_each_entry(codec, &bus->codec_list, list) {
-		if (snd_hda_codec_needs_resume(codec))
-			return 1;
-	}
-	return 0;
-}
-
 static int azx_suspend(struct pci_dev *pci, pm_message_t state)
 {
 	struct snd_card *card = pci_get_drvdata(pci);
@@ -2410,8 +2412,7 @@ static int azx_resume(struct pci_dev *pci)
 		return -EIO;
 	azx_init_pci(chip);
 
-	if (snd_hda_codecs_inuse(chip->bus))
-		azx_init_chip(chip, 1);
+	azx_init_chip(chip, 1);
 
 	snd_hda_resume(chip->bus);
 	snd_power_change_state(card, SNDRV_CTL_POWER_D0);
@@ -2565,6 +2566,8 @@ static struct snd_pci_quirk probe_mask_list[] __devinitdata = {
 	/* forced codec slots */
 	SND_PCI_QUIRK(0x1043, 0x1262, "ASUS W5Fm", 0x103),
 	SND_PCI_QUIRK(0x1046, 0x1262, "ASUS W5F", 0x103),
+	/* WinFast VP200 H (Teradici) user reported broken communication */
+	SND_PCI_QUIRK(0x3a21, 0x040d, "WinFast VP200 H", 0x101),
 	{}
 };
 
@@ -3130,6 +3133,11 @@ static DEFINE_PCI_DEVICE_TABLE(azx_ids) = {
 	  .driver_data = AZX_DRIVER_CTX | AZX_DCAPS_CTX_WORKAROUND |
 	  AZX_DCAPS_RIRB_PRE_DELAY | AZX_DCAPS_POSFIX_LPIB },
 #endif
+	/* CTHDA chips */
+	{ PCI_DEVICE(0x1102, 0x0010),
+	  .driver_data = AZX_DRIVER_CTHDA | AZX_DCAPS_PRESET_CTHDA },
+	{ PCI_DEVICE(0x1102, 0x0012),
+	  .driver_data = AZX_DRIVER_CTHDA | AZX_DCAPS_PRESET_CTHDA },
 	/* Vortex86MX */
 	{ PCI_DEVICE(0x17f3, 0x3010), .driver_data = AZX_DRIVER_GENERIC },
 	/* VMware HDAudio */
@@ -3148,7 +3156,7 @@ static DEFINE_PCI_DEVICE_TABLE(azx_ids) = {
 MODULE_DEVICE_TABLE(pci, azx_ids);
 
 /* pci_driver definition */
-static struct pci_driver driver = {
+static struct pci_driver azx_driver = {
 	.name = KBUILD_MODNAME,
 	.id_table = azx_ids,
 	.probe = azx_probe,
@@ -3159,15 +3167,4 @@ static struct pci_driver driver = {
 #endif
 };
 
-static int __init alsa_card_azx_init(void)
-{
-	return pci_register_driver(&driver);
-}
-
-static void __exit alsa_card_azx_exit(void)
-{
-	pci_unregister_driver(&driver);
-}
-
-module_init(alsa_card_azx_init)
-module_exit(alsa_card_azx_exit)
+module_pci_driver(azx_driver);
diff --git a/sound/pci/hda/hda_jack.c b/sound/pci/hda/hda_jack.c
index d68948499fbc..2dd1c113a4c1 100644
--- a/sound/pci/hda/hda_jack.c
+++ b/sound/pci/hda/hda_jack.c
@@ -17,6 +17,7 @@
 #include <sound/jack.h>
 #include "hda_codec.h"
 #include "hda_local.h"
+#include "hda_auto_parser.h"
 #include "hda_jack.h"
 
 bool is_jack_detectable(struct hda_codec *codec, hda_nid_t nid)
diff --git a/sound/pci/hda/hda_jack.h b/sound/pci/hda/hda_jack.h
index c66655cf413a..8ae52465ec5d 100644
--- a/sound/pci/hda/hda_jack.h
+++ b/sound/pci/hda/hda_jack.h
@@ -12,6 +12,8 @@
 #ifndef __SOUND_HDA_JACK_H
 #define __SOUND_HDA_JACK_H
 
+struct auto_pin_cfg;
+
 struct hda_jack_tbl {
 	hda_nid_t nid;
 	unsigned char action;		/* event action (0 = none) */
diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h
index 0ec9248165bc..9a096a8e0fc5 100644
--- a/sound/pci/hda/hda_local.h
+++ b/sound/pci/hda/hda_local.h
@@ -262,6 +262,8 @@ int snd_hda_input_mux_put(struct hda_codec *codec,
 			  const struct hda_input_mux *imux,
 			  struct snd_ctl_elem_value *ucontrol, hda_nid_t nid,
 			  unsigned int *cur_val);
+int snd_hda_add_imux_item(struct hda_input_mux *imux, const char *label,
+			  int index, int *type_index_ret);
 
 /*
  * Channel mode helper
@@ -393,72 +395,7 @@ struct hda_bus_unsolicited {
 	struct hda_bus *bus;
 };
 
-/*
- * Helper for automatic pin configuration
- */
-
-enum {
-	AUTO_PIN_MIC,
-	AUTO_PIN_LINE_IN,
-	AUTO_PIN_CD,
-	AUTO_PIN_AUX,
-	AUTO_PIN_LAST
-};
-
-enum {
-	AUTO_PIN_LINE_OUT,
-	AUTO_PIN_SPEAKER_OUT,
-	AUTO_PIN_HP_OUT
-};
-
-#define AUTO_CFG_MAX_OUTS	HDA_MAX_OUTS
-#define AUTO_CFG_MAX_INS	8
-
-struct auto_pin_cfg_item {
-	hda_nid_t pin;
-	int type;
-};
-
-struct auto_pin_cfg;
-const char *hda_get_autocfg_input_label(struct hda_codec *codec,
-					const struct auto_pin_cfg *cfg,
-					int input);
-int snd_hda_get_pin_label(struct hda_codec *codec, hda_nid_t nid,
-			  const struct auto_pin_cfg *cfg,
-			  char *label, int maxlen, int *indexp);
-int snd_hda_add_imux_item(struct hda_input_mux *imux, const char *label,
-			  int index, int *type_index_ret);
-
-enum {
-	INPUT_PIN_ATTR_UNUSED,	/* pin not connected */
-	INPUT_PIN_ATTR_INT,	/* internal mic/line-in */
-	INPUT_PIN_ATTR_DOCK,	/* docking mic/line-in */
-	INPUT_PIN_ATTR_NORMAL,	/* mic/line-in jack */
-	INPUT_PIN_ATTR_FRONT,	/* mic/line-in jack in front */
-	INPUT_PIN_ATTR_REAR,	/* mic/line-in jack in rear */
-};
-
-int snd_hda_get_input_pin_attr(unsigned int def_conf);
-
-struct auto_pin_cfg {
-	int line_outs;
-	/* sorted in the order of Front/Surr/CLFE/Side */
-	hda_nid_t line_out_pins[AUTO_CFG_MAX_OUTS];
-	int speaker_outs;
-	hda_nid_t speaker_pins[AUTO_CFG_MAX_OUTS];
-	int hp_outs;
-	int line_out_type;	/* AUTO_PIN_XXX_OUT */
-	hda_nid_t hp_pins[AUTO_CFG_MAX_OUTS];
-	int num_inputs;
-	struct auto_pin_cfg_item inputs[AUTO_CFG_MAX_INS];
-	int dig_outs;
-	hda_nid_t dig_out_pins[2];
-	hda_nid_t dig_in_pin;
-	hda_nid_t mono_out_pin;
-	int dig_out_type[2]; /* HDA_PCM_TYPE_XXX */
-	int dig_in_type; /* HDA_PCM_TYPE_XXX */
-};
-
+/* helper macros to retrieve pin default-config values */
 #define get_defcfg_connect(cfg) \
 	((cfg & AC_DEFCFG_PORT_CONN) >> AC_DEFCFG_PORT_CONN_SHIFT)
 #define get_defcfg_association(cfg) \
@@ -472,19 +409,6 @@ struct auto_pin_cfg {
 #define get_defcfg_misc(cfg) \
 	((cfg & AC_DEFCFG_MISC) >> AC_DEFCFG_MISC_SHIFT)
 
-/* bit-flags for snd_hda_parse_pin_def_config() behavior */
-#define HDA_PINCFG_NO_HP_FIXUP	(1 << 0) /* no HP-split */
-#define HDA_PINCFG_NO_LO_FIXUP	(1 << 1) /* don't take other outs as LO */
-
-int snd_hda_parse_pin_defcfg(struct hda_codec *codec,
-			     struct auto_pin_cfg *cfg,
-			     const hda_nid_t *ignore_nids,
-			     unsigned int cond_flags);
-
-/* older function */
-#define snd_hda_parse_pin_def_config(codec, cfg, ignore) \
-	snd_hda_parse_pin_defcfg(codec, cfg, ignore, 0)
-
 /* amp values */
 #define AMP_IN_MUTE(idx)	(0x7080 | ((idx)<<8))
 #define AMP_IN_UNMUTE(idx)	(0x7000 | ((idx)<<8))
@@ -502,6 +426,46 @@ int snd_hda_parse_pin_defcfg(struct hda_codec *codec,
 #define PIN_HP			(AC_PINCTL_OUT_EN | AC_PINCTL_HP_EN)
 #define PIN_HP_AMP		(AC_PINCTL_HP_EN)
 
+unsigned int snd_hda_get_default_vref(struct hda_codec *codec, hda_nid_t pin);
+int _snd_hda_set_pin_ctl(struct hda_codec *codec, hda_nid_t pin,
+			 unsigned int val, bool cached);
+
+/**
+ * _snd_hda_set_pin_ctl - Set a pin-control value safely
+ * @codec: the codec instance
+ * @pin: the pin NID to set the control
+ * @val: the pin-control value (AC_PINCTL_* bits)
+ *
+ * This function sets the pin-control value to the given pin, but
+ * filters out the invalid pin-control bits when the pin has no such
+ * capabilities.  For example, when PIN_HP is passed but the pin has no
+ * HP-drive capability, the HP bit is omitted.
+ *
+ * The function doesn't check the input VREF capability bits, though.
+ * Use snd_hda_get_default_vref() to guess the right value.
+ * Also, this function is only for analog pins, not for HDMI pins.
+ */
+static inline int
+snd_hda_set_pin_ctl(struct hda_codec *codec, hda_nid_t pin, unsigned int val)
+{
+	return _snd_hda_set_pin_ctl(codec, pin, val, false);
+}
+
+/**
+ * snd_hda_set_pin_ctl_cache - Set a pin-control value safely
+ * @codec: the codec instance
+ * @pin: the pin NID to set the control
+ * @val: the pin-control value (AC_PINCTL_* bits)
+ *
+ * Just like snd_hda_set_pin_ctl() but write to cache as well.
+ */
+static inline int
+snd_hda_set_pin_ctl_cache(struct hda_codec *codec, hda_nid_t pin,
+			  unsigned int val)
+{
+	return _snd_hda_set_pin_ctl(codec, pin, val, true);
+}
+
 /*
  * get widget capabilities
  */
diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index 7143393927da..d8b2d6dee986 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -28,6 +28,7 @@
 #include <sound/core.h>
 #include "hda_codec.h"
 #include "hda_local.h"
+#include "hda_auto_parser.h"
 #include "hda_beep.h"
 #include "hda_jack.h"
 
@@ -1742,9 +1743,7 @@ static int ad1981_hp_master_sw_put(struct snd_kcontrol *kcontrol,
 	if (! ad198x_eapd_put(kcontrol, ucontrol))
 		return 0;
 	/* change speaker pin appropriately */
-	snd_hda_codec_write(codec, 0x05, 0,
-			    AC_VERB_SET_PIN_WIDGET_CONTROL,
-			    spec->cur_eapd ? PIN_OUT : 0);
+	snd_hda_set_pin_ctl(codec, 0x05, spec->cur_eapd ? PIN_OUT : 0);
 	/* toggle HP mute appropriately */
 	snd_hda_codec_amp_stereo(codec, 0x06, HDA_OUTPUT, 0,
 				 HDA_AMP_MUTE,
@@ -3103,7 +3102,7 @@ static void ad1988_auto_set_output_and_unmute(struct hda_codec *codec,
 					      int dac_idx)
 {
 	/* set as output */
-	snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, pin_type);
+	snd_hda_set_pin_ctl(codec, nid, pin_type);
 	snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE);
 	switch (nid) {
 	case 0x11: /* port-A - DAC 03 */
@@ -3157,6 +3156,7 @@ static void ad1988_auto_init_analog_input(struct hda_codec *codec)
 	for (i = 0; i < cfg->num_inputs; i++) {
 		hda_nid_t nid = cfg->inputs[i].pin;
 		int type = cfg->inputs[i].type;
+		int val;
 		switch (nid) {
 		case 0x15: /* port-C */
 			snd_hda_codec_write(codec, 0x33, 0, AC_VERB_SET_CONNECT_SEL, 0x0);
@@ -3165,8 +3165,10 @@ static void ad1988_auto_init_analog_input(struct hda_codec *codec)
 			snd_hda_codec_write(codec, 0x34, 0, AC_VERB_SET_CONNECT_SEL, 0x0);
 			break;
 		}
-		snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
-				    type == AUTO_PIN_MIC ? PIN_VREF80 : PIN_IN);
+		val = PIN_IN;
+		if (type == AUTO_PIN_MIC)
+			val |= snd_hda_get_default_vref(codec, nid);
+		snd_hda_set_pin_ctl(codec, nid, val);
 		if (nid != AD1988_PIN_CD_NID)
 			snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE,
 					    AMP_OUT_MUTE);
diff --git a/sound/pci/hda/patch_ca0110.c b/sound/pci/hda/patch_ca0110.c
index 09ccfabb4a17..19ae14f739cb 100644
--- a/sound/pci/hda/patch_ca0110.c
+++ b/sound/pci/hda/patch_ca0110.c
@@ -26,6 +26,7 @@
 #include <sound/core.h>
 #include "hda_codec.h"
 #include "hda_local.h"
+#include "hda_auto_parser.h"
 
 /*
  */
@@ -341,8 +342,7 @@ static int ca0110_build_pcms(struct hda_codec *codec)
 static void init_output(struct hda_codec *codec, hda_nid_t pin, hda_nid_t dac)
 {
 	if (pin) {
-		snd_hda_codec_write(codec, pin, 0,
-				    AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP);
+		snd_hda_set_pin_ctl(codec, pin, PIN_HP);
 		if (get_wcaps(codec, pin) & AC_WCAP_OUT_AMP)
 			snd_hda_codec_write(codec, pin, 0,
 					    AC_VERB_SET_AMP_GAIN_MUTE,
@@ -356,8 +356,8 @@ static void init_output(struct hda_codec *codec, hda_nid_t pin, hda_nid_t dac)
 static void init_input(struct hda_codec *codec, hda_nid_t pin, hda_nid_t adc)
 {
 	if (pin) {
-		snd_hda_codec_write(codec, pin, 0,
-				    AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80);
+		snd_hda_set_pin_ctl(codec, pin, PIN_IN |
+				    snd_hda_get_default_vref(codec, pin));
 		if (get_wcaps(codec, pin) & AC_WCAP_IN_AMP)
 			snd_hda_codec_write(codec, pin, 0,
 					    AC_VERB_SET_AMP_GAIN_MUTE,
diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c
index 21d91d580da8..d0d3540e39e7 100644
--- a/sound/pci/hda/patch_ca0132.c
+++ b/sound/pci/hda/patch_ca0132.c
@@ -30,6 +30,7 @@
 #include <sound/core.h>
 #include "hda_codec.h"
 #include "hda_local.h"
+#include "hda_auto_parser.h"
 
 #define WIDGET_CHIP_CTRL      0x15
 #define WIDGET_DSP_CTRL       0x16
@@ -239,8 +240,7 @@ enum get_set {
 static void init_output(struct hda_codec *codec, hda_nid_t pin, hda_nid_t dac)
 {
 	if (pin) {
-		snd_hda_codec_write(codec, pin, 0,
-				    AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP);
+		snd_hda_set_pin_ctl(codec, pin, PIN_HP);
 		if (get_wcaps(codec, pin) & AC_WCAP_OUT_AMP)
 			snd_hda_codec_write(codec, pin, 0,
 					    AC_VERB_SET_AMP_GAIN_MUTE,
@@ -254,9 +254,8 @@ static void init_output(struct hda_codec *codec, hda_nid_t pin, hda_nid_t dac)
 static void init_input(struct hda_codec *codec, hda_nid_t pin, hda_nid_t adc)
 {
 	if (pin) {
-		snd_hda_codec_write(codec, pin, 0,
-				    AC_VERB_SET_PIN_WIDGET_CONTROL,
-				    PIN_VREF80);
+		snd_hda_set_pin_ctl(codec, pin, PIN_IN |
+				    snd_hda_get_default_vref(codec, pin));
 		if (get_wcaps(codec, pin) & AC_WCAP_IN_AMP)
 			snd_hda_codec_write(codec, pin, 0,
 					    AC_VERB_SET_AMP_GAIN_MUTE,
diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c
index c83ccdba1e5a..9647ed4d7929 100644
--- a/sound/pci/hda/patch_cirrus.c
+++ b/sound/pci/hda/patch_cirrus.c
@@ -26,6 +26,7 @@
 #include <sound/core.h>
 #include "hda_codec.h"
 #include "hda_local.h"
+#include "hda_auto_parser.h"
 #include "hda_jack.h"
 #include <sound/tlv.h>
 
@@ -933,8 +934,7 @@ static void cs_automute(struct hda_codec *codec)
 			pin_ctl = 0;
 
 		nid = cfg->speaker_pins[i];
-		snd_hda_codec_write(codec, nid, 0,
-				    AC_VERB_SET_PIN_WIDGET_CONTROL, pin_ctl);
+		snd_hda_set_pin_ctl(codec, nid, pin_ctl);
 	}
 	if (spec->gpio_eapd_hp) {
 		unsigned int gpio = hp_present ?
@@ -948,16 +948,14 @@ static void cs_automute(struct hda_codec *codec)
 		/* mute HPs if spdif jack (SENSE_B) is present */
 		for (i = 0; i < cfg->hp_outs; i++) {
 			nid = cfg->hp_pins[i];
-			snd_hda_codec_write(codec, nid, 0,
-				AC_VERB_SET_PIN_WIDGET_CONTROL,
+			snd_hda_set_pin_ctl(codec, nid,
 				(spdif_present && spec->sense_b) ? 0 : PIN_HP);
 		}
 
 		/* SPDIF TX on/off */
 		if (cfg->dig_outs) {
 			nid = cfg->dig_out_pins[0];
-			snd_hda_codec_write(codec, nid, 0,
-				AC_VERB_SET_PIN_WIDGET_CONTROL,
+			snd_hda_set_pin_ctl(codec, nid,
 				spdif_present ? PIN_OUT : 0);
 
 		}
@@ -1024,13 +1022,11 @@ static void init_output(struct hda_codec *codec)
 
 	/* set appropriate pin controls */
 	for (i = 0; i < cfg->line_outs; i++)
-		snd_hda_codec_write(codec, cfg->line_out_pins[i], 0,
-				    AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT);
+		snd_hda_set_pin_ctl(codec, cfg->line_out_pins[i], PIN_OUT);
 	/* HP */
 	for (i = 0; i < cfg->hp_outs; i++) {
 		hda_nid_t nid = cfg->hp_pins[i];
-		snd_hda_codec_write(codec, nid, 0,
-				    AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP);
+		snd_hda_set_pin_ctl(codec, nid, PIN_HP);
 		if (!cfg->speaker_outs)
 			continue;
 		if (get_wcaps(codec, nid) & AC_WCAP_UNSOL_CAP) {
@@ -1041,8 +1037,7 @@ static void init_output(struct hda_codec *codec)
 
 	/* Speaker */
 	for (i = 0; i < cfg->speaker_outs; i++)
-		snd_hda_codec_write(codec, cfg->speaker_pins[i], 0,
-				    AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT);
+		snd_hda_set_pin_ctl(codec, cfg->speaker_pins[i], PIN_OUT);
 
 	/* SPDIF is enabled on presence detect for CS421x */
 	if (spec->hp_detect || spec->spdif_detect)
@@ -1063,14 +1058,9 @@ static void init_input(struct hda_codec *codec)
 			continue;
 		/* set appropriate pin control and mute first */
 		ctl = PIN_IN;
-		if (cfg->inputs[i].type == AUTO_PIN_MIC) {
-			unsigned int caps = snd_hda_query_pin_caps(codec, pin);
-			caps >>= AC_PINCAP_VREF_SHIFT;
-			if (caps & AC_PINCAP_VREF_80)
-				ctl = PIN_VREF80;
-		}
-		snd_hda_codec_write(codec, pin, 0,
-				    AC_VERB_SET_PIN_WIDGET_CONTROL, ctl);
+		if (cfg->inputs[i].type == AUTO_PIN_MIC)
+			ctl |= snd_hda_get_default_vref(codec, pin);
+		snd_hda_set_pin_ctl(codec, pin, ctl);
 		snd_hda_codec_write(codec, spec->adc_nid[i], 0,
 				    AC_VERB_SET_AMP_GAIN_MUTE,
 				    AMP_IN_MUTE(spec->adc_idx[i]));
diff --git a/sound/pci/hda/patch_cmedia.c b/sound/pci/hda/patch_cmedia.c
index b6767b4ced44..c8fdaaefe702 100644
--- a/sound/pci/hda/patch_cmedia.c
+++ b/sound/pci/hda/patch_cmedia.c
@@ -29,6 +29,7 @@
 #include <sound/core.h>
 #include "hda_codec.h"
 #include "hda_local.h"
+#include "hda_auto_parser.h"
 #define NUM_PINS	11
 
 
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index d906c5b74cf0..3acb5824ad39 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -30,6 +30,7 @@
 
 #include "hda_codec.h"
 #include "hda_local.h"
+#include "hda_auto_parser.h"
 #include "hda_beep.h"
 #include "hda_jack.h"
 
@@ -66,6 +67,7 @@ struct imux_info {
 };
 
 struct conexant_spec {
+	struct hda_gen_spec gen;
 
 	const struct snd_kcontrol_new *mixers[5];
 	int num_mixers;
@@ -141,6 +143,7 @@ struct conexant_spec {
 	unsigned int hp_laptop:1;
 	unsigned int asus:1;
 	unsigned int pin_eapd_ctrls:1;
+	unsigned int fixup_stereo_dmic:1;
 
 	unsigned int adc_switching:1;
 
@@ -1601,17 +1604,13 @@ static void cxt5051_update_speaker(struct hda_codec *codec)
 	unsigned int pinctl;
 	/* headphone pin */
 	pinctl = (spec->hp_present && spec->cur_eapd) ? PIN_HP : 0;
-	snd_hda_codec_write(codec, 0x16, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
-			    pinctl);
+	snd_hda_set_pin_ctl(codec, 0x16, pinctl);
 	/* speaker pin */
 	pinctl = (!spec->hp_present && spec->cur_eapd) ? PIN_OUT : 0;
-	snd_hda_codec_write(codec, 0x1a, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
-			    pinctl);
+	snd_hda_set_pin_ctl(codec, 0x1a, pinctl);
 	/* on ideapad there is an additional speaker (subwoofer) to mute */
 	if (spec->ideapad)
-		snd_hda_codec_write(codec, 0x1b, 0,
-				    AC_VERB_SET_PIN_WIDGET_CONTROL,
-				    pinctl);
+		snd_hda_set_pin_ctl(codec, 0x1b, pinctl);
 }
 
 /* turn on/off EAPD (+ mute HP) as a master switch */
@@ -1996,8 +1995,7 @@ static void cxt5066_update_speaker(struct hda_codec *codec)
 
 	/* Port A (HP) */
 	pinctl = (hp_port_a_present(spec) && spec->cur_eapd) ? PIN_HP : 0;
-	snd_hda_codec_write(codec, 0x19, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
-			pinctl);
+	snd_hda_set_pin_ctl(codec, 0x19, pinctl);
 
 	/* Port D (HP/LO) */
 	pinctl = spec->cur_eapd ? spec->port_d_mode : 0;
@@ -2010,13 +2008,11 @@ static void cxt5066_update_speaker(struct hda_codec *codec)
 		if (!hp_port_d_present(spec))
 			pinctl = 0;
 	}
-	snd_hda_codec_write(codec, 0x1c, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
-			pinctl);
+	snd_hda_set_pin_ctl(codec, 0x1c, pinctl);
 
 	/* CLASS_D AMP */
 	pinctl = (!spec->hp_present && spec->cur_eapd) ? PIN_OUT : 0;
-	snd_hda_codec_write(codec, 0x1f, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
-			pinctl);
+	snd_hda_set_pin_ctl(codec, 0x1f, pinctl);
 }
 
 /* turn on/off EAPD (+ mute HP) as a master switch */
@@ -2047,8 +2043,7 @@ static int cxt5066_set_olpc_dc_bias(struct hda_codec *codec)
 	/* Even though port F is the DC input, the bias is controlled on port B.
 	 * we also leave that port as an active input (but unselected) in DC mode
 	 * just in case that is necessary to make the bias setting take effect. */
-	return snd_hda_codec_write_cache(codec, 0x1a, 0,
-		AC_VERB_SET_PIN_WIDGET_CONTROL,
+	return snd_hda_set_pin_ctl_cache(codec, 0x1a,
 		cxt5066_olpc_dc_bias.items[spec->dc_input_bias].index);
 }
 
@@ -2081,14 +2076,14 @@ static void cxt5066_olpc_select_mic(struct hda_codec *codec)
 	}
 
 	/* disable DC (port F) */
-	snd_hda_codec_write(codec, 0x1e, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, 0);
+	snd_hda_set_pin_ctl(codec, 0x1e, 0);
 
 	/* external mic, port B */
-	snd_hda_codec_write(codec, 0x1a, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
+	snd_hda_set_pin_ctl(codec, 0x1a,
 		spec->ext_mic_present ? CXT5066_OLPC_EXT_MIC_BIAS : 0);
 
 	/* internal mic, port C */
-	snd_hda_codec_write(codec, 0x1b, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
+	snd_hda_set_pin_ctl(codec, 0x1b,
 		spec->ext_mic_present ? 0 : PIN_VREF80);
 }
 
@@ -3357,9 +3352,7 @@ static void do_automute(struct hda_codec *codec, int num_pins,
 	struct conexant_spec *spec = codec->spec;
 	int i;
 	for (i = 0; i < num_pins; i++)
-		snd_hda_codec_write(codec, pins[i], 0,
-				    AC_VERB_SET_PIN_WIDGET_CONTROL,
-				    on ? PIN_OUT : 0);
+		snd_hda_set_pin_ctl(codec, pins[i], on ? PIN_OUT : 0);
 	if (spec->pin_eapd_ctrls)
 		cx_auto_turn_eapd(codec, num_pins, pins, on);
 }
@@ -3976,8 +3969,7 @@ static void cx_auto_init_output(struct hda_codec *codec)
 		if (snd_hda_query_pin_caps(codec, cfg->hp_pins[i]) &
 		    AC_PINCAP_HP_DRV)
 			val |= AC_PINCTL_HP_EN;
-		snd_hda_codec_write(codec, cfg->hp_pins[i], 0,
-				    AC_VERB_SET_PIN_WIDGET_CONTROL, val);
+		snd_hda_set_pin_ctl(codec, cfg->hp_pins[i], val);
 	}
 	mute_outputs(codec, cfg->hp_outs, cfg->hp_pins);
 	mute_outputs(codec, cfg->line_outs, cfg->line_out_pins);
@@ -4030,13 +4022,11 @@ static void cx_auto_init_input(struct hda_codec *codec)
 	}
 
 	for (i = 0; i < cfg->num_inputs; i++) {
-		unsigned int type;
+		hda_nid_t pin = cfg->inputs[i].pin;
+		unsigned int type = PIN_IN;
 		if (cfg->inputs[i].type == AUTO_PIN_MIC)
-			type = PIN_VREF80;
-		else
-			type = PIN_IN;
-		snd_hda_codec_write(codec, cfg->inputs[i].pin, 0,
-				    AC_VERB_SET_PIN_WIDGET_CONTROL, type);
+			type |= snd_hda_get_default_vref(codec, pin);
+		snd_hda_set_pin_ctl(codec, pin, type);
 	}
 
 	if (spec->auto_mic) {
@@ -4063,11 +4053,9 @@ static void cx_auto_init_digital(struct hda_codec *codec)
 	struct auto_pin_cfg *cfg = &spec->autocfg;
 
 	if (spec->multiout.dig_out_nid)
-		snd_hda_codec_write(codec, cfg->dig_out_pins[0], 0,
-				    AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT);
+		snd_hda_set_pin_ctl(codec, cfg->dig_out_pins[0], PIN_OUT);
 	if (spec->dig_in_nid)
-		snd_hda_codec_write(codec, cfg->dig_in_pin, 0,
-				    AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN);
+		snd_hda_set_pin_ctl(codec, cfg->dig_in_pin, PIN_IN);
 }
 
 static int cx_auto_init(struct hda_codec *codec)
@@ -4084,9 +4072,9 @@ static int cx_auto_init(struct hda_codec *codec)
 
 static int cx_auto_add_volume_idx(struct hda_codec *codec, const char *basename,
 			      const char *dir, int cidx,
-			      hda_nid_t nid, int hda_dir, int amp_idx)
+			      hda_nid_t nid, int hda_dir, int amp_idx, int chs)
 {
-	static char name[32];
+	static char name[44];
 	static struct snd_kcontrol_new knew[] = {
 		HDA_CODEC_VOLUME(name, 0, 0, 0),
 		HDA_CODEC_MUTE(name, 0, 0, 0),
@@ -4096,7 +4084,7 @@ static int cx_auto_add_volume_idx(struct hda_codec *codec, const char *basename,
 
 	for (i = 0; i < 2; i++) {
 		struct snd_kcontrol *kctl;
-		knew[i].private_value = HDA_COMPOSE_AMP_VAL(nid, 3, amp_idx,
+		knew[i].private_value = HDA_COMPOSE_AMP_VAL(nid, chs, amp_idx,
 							    hda_dir);
 		knew[i].subdevice = HDA_SUBDEV_AMP_FLAG;
 		knew[i].index = cidx;
@@ -4115,7 +4103,7 @@ static int cx_auto_add_volume_idx(struct hda_codec *codec, const char *basename,
 }
 
 #define cx_auto_add_volume(codec, str, dir, cidx, nid, hda_dir)		\
-	cx_auto_add_volume_idx(codec, str, dir, cidx, nid, hda_dir, 0)
+	cx_auto_add_volume_idx(codec, str, dir, cidx, nid, hda_dir, 0, 3)
 
 #define cx_auto_add_pb_volume(codec, nid, str, idx)			\
 	cx_auto_add_volume(codec, str, " Playback", idx, nid, HDA_OUTPUT)
@@ -4185,6 +4173,36 @@ static int cx_auto_build_output_controls(struct hda_codec *codec)
 	return 0;
 }
 
+/* Returns zero if this is a normal stereo channel, and non-zero if it should
+   be split in two independent channels.
+   dest_label must be at least 44 characters. */
+static int cx_auto_get_rightch_label(struct hda_codec *codec, const char *label,
+				     char *dest_label, int nid)
+{
+	struct conexant_spec *spec = codec->spec;
+	int i;
+
+	if (!spec->fixup_stereo_dmic)
+		return 0;
+
+	for (i = 0; i < AUTO_CFG_MAX_INS; i++) {
+		int def_conf;
+		if (spec->autocfg.inputs[i].pin != nid)
+			continue;
+
+		if (spec->autocfg.inputs[i].type != AUTO_PIN_MIC)
+			return 0;
+		def_conf = snd_hda_codec_get_pincfg(codec, nid);
+		if (snd_hda_get_input_pin_attr(def_conf) != INPUT_PIN_ATTR_INT)
+			return 0;
+
+		/* Finally found the inverted internal mic! */
+		snprintf(dest_label, 44, "Inverted %s", label);
+		return 1;
+	}
+	return 0;
+}
+
 static int cx_auto_add_capture_volume(struct hda_codec *codec, hda_nid_t nid,
 				      const char *label, const char *pfx,
 				      int cidx)
@@ -4193,14 +4211,25 @@ static int cx_auto_add_capture_volume(struct hda_codec *codec, hda_nid_t nid,
 	int i;
 
 	for (i = 0; i < spec->num_adc_nids; i++) {
+		char rightch_label[44];
 		hda_nid_t adc_nid = spec->adc_nids[i];
 		int idx = get_input_connection(codec, adc_nid, nid);
 		if (idx < 0)
 			continue;
 		if (codec->single_adc_amp)
 			idx = 0;
+
+		if (cx_auto_get_rightch_label(codec, label, rightch_label, nid)) {
+			/* Make two independent kcontrols for left and right */
+			int err = cx_auto_add_volume_idx(codec, label, pfx,
+					      cidx, adc_nid, HDA_INPUT, idx, 1);
+			if (err < 0)
+				return err;
+			return cx_auto_add_volume_idx(codec, rightch_label, pfx,
+						      cidx, adc_nid, HDA_INPUT, idx, 2);
+		}
 		return cx_auto_add_volume_idx(codec, label, pfx,
-					      cidx, adc_nid, HDA_INPUT, idx);
+					      cidx, adc_nid, HDA_INPUT, idx, 3);
 	}
 	return 0;
 }
@@ -4213,9 +4242,19 @@ static int cx_auto_add_boost_volume(struct hda_codec *codec, int idx,
 	int i, con;
 
 	nid = spec->imux_info[idx].pin;
-	if (get_wcaps(codec, nid) & AC_WCAP_IN_AMP)
+	if (get_wcaps(codec, nid) & AC_WCAP_IN_AMP) {
+		char rightch_label[44];
+		if (cx_auto_get_rightch_label(codec, label, rightch_label, nid)) {
+			int err = cx_auto_add_volume_idx(codec, label, " Boost",
+							 cidx, nid, HDA_INPUT, 0, 1);
+			if (err < 0)
+				return err;
+			return cx_auto_add_volume_idx(codec, rightch_label, " Boost",
+						      cidx, nid, HDA_INPUT, 0, 2);
+		}
 		return cx_auto_add_volume(codec, label, " Boost", cidx,
 					  nid, HDA_INPUT);
+	}
 	con = __select_input_connection(codec, spec->imux_info[idx].adc, nid,
 					&mux, false, 0);
 	if (con < 0)
@@ -4370,37 +4409,21 @@ static const struct hda_codec_ops cx_auto_patch_ops = {
 /*
  * pin fix-up
  */
-struct cxt_pincfg {
-	hda_nid_t nid;
-	u32 val;
-};
-
-static void apply_pincfg(struct hda_codec *codec, const struct cxt_pincfg *cfg)
-{
-	for (; cfg->nid; cfg++)
-		snd_hda_codec_set_pincfg(codec, cfg->nid, cfg->val);
-
-}
-
-static void apply_pin_fixup(struct hda_codec *codec,
-			    const struct snd_pci_quirk *quirk,
-			    const struct cxt_pincfg **table)
-{
-	quirk = snd_pci_quirk_lookup(codec->bus->pci, quirk);
-	if (quirk) {
-		snd_printdd(KERN_INFO "hda_codec: applying pincfg for %s\n",
-			    quirk->name);
-		apply_pincfg(codec, table[quirk->value]);
-	}
-}
-
 enum {
 	CXT_PINCFG_LENOVO_X200,
 	CXT_PINCFG_LENOVO_TP410,
+	CXT_FIXUP_STEREO_DMIC,
 };
 
+static void cxt_fixup_stereo_dmic(struct hda_codec *codec,
+				  const struct hda_fixup *fix, int action)
+{
+	struct conexant_spec *spec = codec->spec;
+	spec->fixup_stereo_dmic = 1;
+}
+
 /* ThinkPad X200 & co with cxt5051 */
-static const struct cxt_pincfg cxt_pincfg_lenovo_x200[] = {
+static const struct hda_pintbl cxt_pincfg_lenovo_x200[] = {
 	{ 0x16, 0x042140ff }, /* HP (seq# overridden) */
 	{ 0x17, 0x21a11000 }, /* dock-mic */
 	{ 0x19, 0x2121103f }, /* dock-HP */
@@ -4409,16 +4432,26 @@ static const struct cxt_pincfg cxt_pincfg_lenovo_x200[] = {
 };
 
 /* ThinkPad 410/420/510/520, X201 & co with cxt5066 */
-static const struct cxt_pincfg cxt_pincfg_lenovo_tp410[] = {
+static const struct hda_pintbl cxt_pincfg_lenovo_tp410[] = {
 	{ 0x19, 0x042110ff }, /* HP (seq# overridden) */
 	{ 0x1a, 0x21a190f0 }, /* dock-mic */
 	{ 0x1c, 0x212140ff }, /* dock-HP */
 	{}
 };
 
-static const struct cxt_pincfg *cxt_pincfg_tbl[] = {
-	[CXT_PINCFG_LENOVO_X200] = cxt_pincfg_lenovo_x200,
-	[CXT_PINCFG_LENOVO_TP410] = cxt_pincfg_lenovo_tp410,
+static const struct hda_fixup cxt_fixups[] = {
+	[CXT_PINCFG_LENOVO_X200] = {
+		.type = HDA_FIXUP_PINS,
+		.v.pins = cxt_pincfg_lenovo_x200,
+	},
+	[CXT_PINCFG_LENOVO_TP410] = {
+		.type = HDA_FIXUP_PINS,
+		.v.pins = cxt_pincfg_lenovo_tp410,
+	},
+	[CXT_FIXUP_STEREO_DMIC] = {
+		.type = HDA_FIXUP_FUNC,
+		.v.func = cxt_fixup_stereo_dmic,
+	},
 };
 
 static const struct snd_pci_quirk cxt5051_fixups[] = {
@@ -4432,6 +4465,7 @@ static const struct snd_pci_quirk cxt5066_fixups[] = {
 	SND_PCI_QUIRK(0x17aa, 0x215f, "Lenovo T510", CXT_PINCFG_LENOVO_TP410),
 	SND_PCI_QUIRK(0x17aa, 0x21ce, "Lenovo T420", CXT_PINCFG_LENOVO_TP410),
 	SND_PCI_QUIRK(0x17aa, 0x21cf, "Lenovo T520", CXT_PINCFG_LENOVO_TP410),
+	SND_PCI_QUIRK(0x17aa, 0x3975, "Lenovo U300s", CXT_FIXUP_STEREO_DMIC),
 	{}
 };
 
@@ -4471,13 +4505,16 @@ static int patch_conexant_auto(struct hda_codec *codec)
 	case 0x14f15051:
 		add_cx5051_fake_mutes(codec);
 		codec->pin_amp_workaround = 1;
-		apply_pin_fixup(codec, cxt5051_fixups, cxt_pincfg_tbl);
+		snd_hda_pick_fixup(codec, NULL, cxt5051_fixups, cxt_fixups);
 		break;
 	default:
 		codec->pin_amp_workaround = 1;
-		apply_pin_fixup(codec, cxt5066_fixups, cxt_pincfg_tbl);
+		snd_hda_pick_fixup(codec, NULL, cxt5066_fixups, cxt_fixups);
+		break;
 	}
 
+	snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE);
+
 	/* Show mute-led control only on HP laptops
 	 * This is a sort of white-list: on HP laptops, EAPD corresponds
 	 * only to the mute-LED without actualy amp function.  Meanwhile,
@@ -4556,6 +4593,12 @@ static const struct hda_codec_preset snd_hda_preset_conexant[] = {
 	  .patch = patch_conexant_auto },
 	{ .id = 0x14f150b9, .name = "CX20665",
 	  .patch = patch_conexant_auto },
+	{ .id = 0x14f1510f, .name = "CX20751/2",
+	  .patch = patch_conexant_auto },
+	{ .id = 0x14f15110, .name = "CX20751/2",
+	  .patch = patch_conexant_auto },
+	{ .id = 0x14f15111, .name = "CX20753/4",
+	  .patch = patch_conexant_auto },
 	{} /* terminator */
 };
 
@@ -4576,6 +4619,9 @@ MODULE_ALIAS("snd-hda-codec-id:14f150ab");
 MODULE_ALIAS("snd-hda-codec-id:14f150ac");
 MODULE_ALIAS("snd-hda-codec-id:14f150b8");
 MODULE_ALIAS("snd-hda-codec-id:14f150b9");
+MODULE_ALIAS("snd-hda-codec-id:14f1510f");
+MODULE_ALIAS("snd-hda-codec-id:14f15110");
+MODULE_ALIAS("snd-hda-codec-id:14f15111");
 
 MODULE_LICENSE("GPL");
 MODULE_DESCRIPTION("Conexant HD-audio codec");
diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index 83f345f3c961..ad319d4dc32f 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -1592,10 +1592,10 @@ static int nvhdmi_8ch_7x_pcm_prepare(struct hda_pcm_stream *hinfo,
 	unsigned int dataDCC2, channel_id;
 	int i;
 	struct hdmi_spec *spec = codec->spec;
-	struct hda_spdif_out *spdif =
-		snd_hda_spdif_out_of_nid(codec, spec->cvts[0].cvt_nid);
+	struct hda_spdif_out *spdif;
 
 	mutex_lock(&codec->spdif_mutex);
+	spdif = snd_hda_spdif_out_of_nid(codec, spec->cvts[0].cvt_nid);
 
 	chs = substream->runtime->channels;
 
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 708d47c294ee..ff71dcef08ef 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -32,6 +32,7 @@
 #include <sound/jack.h>
 #include "hda_codec.h"
 #include "hda_local.h"
+#include "hda_auto_parser.h"
 #include "hda_beep.h"
 #include "hda_jack.h"
 
@@ -66,8 +67,6 @@ struct alc_customize_define {
 	unsigned int  fixup:1; /* Means that this sku is set by driver, not read from hw */
 };
 
-struct alc_fixup;
-
 struct alc_multi_io {
 	hda_nid_t pin;		/* multi-io widget pin NID */
 	hda_nid_t dac;		/* DAC to be connected */
@@ -82,19 +81,33 @@ enum {
 
 #define MAX_VOL_NIDS	0x40
 
+/* make compatible with old code */
+#define alc_apply_pincfgs	snd_hda_apply_pincfgs
+#define alc_apply_fixup		snd_hda_apply_fixup
+#define alc_pick_fixup		snd_hda_pick_fixup
+#define alc_fixup		hda_fixup
+#define alc_pincfg		hda_pintbl
+#define alc_model_fixup		hda_model_fixup
+
+#define ALC_FIXUP_PINS	HDA_FIXUP_PINS
+#define ALC_FIXUP_VERBS	HDA_FIXUP_VERBS
+#define ALC_FIXUP_FUNC	HDA_FIXUP_FUNC
+
+#define ALC_FIXUP_ACT_PRE_PROBE	HDA_FIXUP_ACT_PRE_PROBE
+#define ALC_FIXUP_ACT_PROBE	HDA_FIXUP_ACT_PROBE
+#define ALC_FIXUP_ACT_INIT	HDA_FIXUP_ACT_INIT
+#define ALC_FIXUP_ACT_BUILD	HDA_FIXUP_ACT_BUILD
+
+
 struct alc_spec {
+	struct hda_gen_spec gen;
+
 	/* codec parameterization */
 	const struct snd_kcontrol_new *mixers[5];	/* mixer arrays */
 	unsigned int num_mixers;
 	const struct snd_kcontrol_new *cap_mixer;	/* capture mixer */
 	unsigned int beep_amp;	/* beep amp value, set via set_beep_amp() */
 
-	const struct hda_verb *init_verbs[10];	/* initialization verbs
-						 * don't forget NULL
-						 * termination!
-						 */
-	unsigned int num_init_verbs;
-
 	char stream_name_analog[32];	/* analog PCM stream */
 	const struct hda_pcm_stream *stream_analog_playback;
 	const struct hda_pcm_stream *stream_analog_capture;
@@ -210,11 +223,6 @@ struct alc_spec {
 	unsigned int pll_coef_idx, pll_coef_bit;
 	unsigned int coef0;
 
-	/* fix-up list */
-	int fixup_id;
-	const struct alc_fixup *fixup_list;
-	const char *fixup_name;
-
 	/* multi-io */
 	int multi_ios;
 	struct alc_multi_io multi_io[4];
@@ -319,13 +327,16 @@ static int alc_mux_select(struct hda_codec *codec, unsigned int adc_idx,
 
 	/* for shared I/O, change the pin-control accordingly */
 	if (spec->shared_mic_hp) {
+		unsigned int val;
+		hda_nid_t pin = spec->autocfg.inputs[1].pin;
 		/* NOTE: this assumes that there are only two inputs, the
 		 * first is the real internal mic and the second is HP jack.
 		 */
-		snd_hda_codec_write(codec, spec->autocfg.inputs[1].pin, 0,
-				    AC_VERB_SET_PIN_WIDGET_CONTROL,
-				    spec->cur_mux[adc_idx] ?
-				    PIN_VREF80 : PIN_HP);
+		if (spec->cur_mux[adc_idx])
+			val = snd_hda_get_default_vref(codec, pin) | PIN_IN;
+		else
+			val = PIN_HP;
+		snd_hda_set_pin_ctl(codec, pin, val);
 		spec->automute_speaker = !spec->cur_mux[adc_idx];
 		call_update_outputs(codec);
 	}
@@ -338,7 +349,7 @@ static int alc_mux_select(struct hda_codec *codec, unsigned int adc_idx,
 	nid = get_capsrc(spec, adc_idx);
 
 	/* no selection? */
-	num_conns = snd_hda_get_conn_list(codec, nid, NULL);
+	num_conns = snd_hda_get_num_conns(codec, nid);
 	if (num_conns <= 1)
 		return 1;
 
@@ -376,25 +387,9 @@ static void alc_set_input_pin(struct hda_codec *codec, hda_nid_t nid,
 			      int auto_pin_type)
 {
 	unsigned int val = PIN_IN;
-
-	if (auto_pin_type == AUTO_PIN_MIC) {
-		unsigned int pincap;
-		unsigned int oldval;
-		oldval = snd_hda_codec_read(codec, nid, 0,
-					    AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
-		pincap = snd_hda_query_pin_caps(codec, nid);
-		pincap = (pincap & AC_PINCAP_VREF) >> AC_PINCAP_VREF_SHIFT;
-		/* if the default pin setup is vref50, we give it priority */
-		if ((pincap & AC_PINCAP_VREF_80) && oldval != PIN_VREF50)
-			val = PIN_VREF80;
-		else if (pincap & AC_PINCAP_VREF_50)
-			val = PIN_VREF50;
-		else if (pincap & AC_PINCAP_VREF_100)
-			val = PIN_VREF100;
-		else if (pincap & AC_PINCAP_VREF_GRD)
-			val = PIN_VREFGRD;
-	}
-	snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, val);
+	if (auto_pin_type == AUTO_PIN_MIC)
+		val |= snd_hda_get_default_vref(codec, nid);
+	snd_hda_set_pin_ctl(codec, nid, val);
 }
 
 /*
@@ -409,13 +404,6 @@ static void add_mixer(struct alc_spec *spec, const struct snd_kcontrol_new *mix)
 	spec->mixers[spec->num_mixers++] = mix;
 }
 
-static void add_verb(struct alc_spec *spec, const struct hda_verb *verb)
-{
-	if (snd_BUG_ON(spec->num_init_verbs >= ARRAY_SIZE(spec->init_verbs)))
-		return;
-	spec->init_verbs[spec->num_init_verbs++] = verb;
-}
-
 /*
  * GPIO setup tables, used in initialization
  */
@@ -517,9 +505,7 @@ static void do_automute(struct hda_codec *codec, int num_pins, hda_nid_t *pins,
 			} else
 				val = 0;
 			val |= pin_bits;
-			snd_hda_codec_write(codec, nid, 0,
-					    AC_VERB_SET_PIN_WIDGET_CONTROL,
-					    val);
+			snd_hda_set_pin_ctl(codec, nid, val);
 			break;
 		case ALC_AUTOMUTE_AMP:
 			snd_hda_codec_amp_stereo(codec, nid, HDA_OUTPUT, 0,
@@ -1200,6 +1186,16 @@ static void alc_auto_check_switches(struct hda_codec *codec)
  */
 #define ALC_FIXUP_SKU_IGNORE (2)
 
+static void alc_fixup_sku_ignore(struct hda_codec *codec,
+				 const struct hda_fixup *fix, int action)
+{
+	struct alc_spec *spec = codec->spec;
+	if (action == HDA_FIXUP_ACT_PRE_PROBE) {
+		spec->cdefine.fixup = 1;
+		spec->cdefine.sku_cfg = ALC_FIXUP_SKU_IGNORE;
+	}
+}
+
 static int alc_auto_parse_customize_define(struct hda_codec *codec)
 {
 	unsigned int ass, tmp, i;
@@ -1402,178 +1398,6 @@ static void alc_ssid_check(struct hda_codec *codec, const hda_nid_t *ports)
 	}
 }
 
-/*
- * Fix-up pin default configurations and add default verbs
- */
-
-struct alc_pincfg {
-	hda_nid_t nid;
-	u32 val;
-};
-
-struct alc_model_fixup {
-	const int id;
-	const char *name;
-};
-
-struct alc_fixup {
-	int type;
-	bool chained;
-	int chain_id;
-	union {
-		unsigned int sku;
-		const struct alc_pincfg *pins;
-		const struct hda_verb *verbs;
-		void (*func)(struct hda_codec *codec,
-			     const struct alc_fixup *fix,
-			     int action);
-	} v;
-};
-
-enum {
-	ALC_FIXUP_INVALID,
-	ALC_FIXUP_SKU,
-	ALC_FIXUP_PINS,
-	ALC_FIXUP_VERBS,
-	ALC_FIXUP_FUNC,
-};
-
-enum {
-	ALC_FIXUP_ACT_PRE_PROBE,
-	ALC_FIXUP_ACT_PROBE,
-	ALC_FIXUP_ACT_INIT,
-	ALC_FIXUP_ACT_BUILD,
-};
-
-static void alc_apply_pincfgs(struct hda_codec *codec,
-			      const struct alc_pincfg *cfg)
-{
-	for (; cfg->nid; cfg++)
-		snd_hda_codec_set_pincfg(codec, cfg->nid, cfg->val);
-}
-
-static void alc_apply_fixup(struct hda_codec *codec, int action)
-{
-	struct alc_spec *spec = codec->spec;
-	int id = spec->fixup_id;
-#ifdef CONFIG_SND_DEBUG_VERBOSE
-	const char *modelname = spec->fixup_name;
-#endif
-	int depth = 0;
-
-	if (!spec->fixup_list)
-		return;
-
-	while (id >= 0) {
-		const struct alc_fixup *fix = spec->fixup_list + id;
-		const struct alc_pincfg *cfg;
-
-		switch (fix->type) {
-		case ALC_FIXUP_SKU:
-			if (action != ALC_FIXUP_ACT_PRE_PROBE || !fix->v.sku)
-				break;
-			snd_printdd(KERN_INFO "hda_codec: %s: "
-				    "Apply sku override for %s\n",
-				    codec->chip_name, modelname);
-			spec->cdefine.sku_cfg = fix->v.sku;
-			spec->cdefine.fixup = 1;
-			break;
-		case ALC_FIXUP_PINS:
-			cfg = fix->v.pins;
-			if (action != ALC_FIXUP_ACT_PRE_PROBE || !cfg)
-				break;
-			snd_printdd(KERN_INFO "hda_codec: %s: "
-				    "Apply pincfg for %s\n",
-				    codec->chip_name, modelname);
-			alc_apply_pincfgs(codec, cfg);
-			break;
-		case ALC_FIXUP_VERBS:
-			if (action != ALC_FIXUP_ACT_PROBE || !fix->v.verbs)
-				break;
-			snd_printdd(KERN_INFO "hda_codec: %s: "
-				    "Apply fix-verbs for %s\n",
-				    codec->chip_name, modelname);
-			add_verb(codec->spec, fix->v.verbs);
-			break;
-		case ALC_FIXUP_FUNC:
-			if (!fix->v.func)
-				break;
-			snd_printdd(KERN_INFO "hda_codec: %s: "
-				    "Apply fix-func for %s\n",
-				    codec->chip_name, modelname);
-			fix->v.func(codec, fix, action);
-			break;
-		default:
-			snd_printk(KERN_ERR "hda_codec: %s: "
-				   "Invalid fixup type %d\n",
-				   codec->chip_name, fix->type);
-			break;
-		}
-		if (!fix->chained)
-			break;
-		if (++depth > 10)
-			break;
-		id = fix->chain_id;
-	}
-}
-
-static void alc_pick_fixup(struct hda_codec *codec,
-			   const struct alc_model_fixup *models,
-			   const struct snd_pci_quirk *quirk,
-			   const struct alc_fixup *fixlist)
-{
-	struct alc_spec *spec = codec->spec;
-	const struct snd_pci_quirk *q;
-	int id = -1;
-	const char *name = NULL;
-
-	/* when model=nofixup is given, don't pick up any fixups */
-	if (codec->modelname && !strcmp(codec->modelname, "nofixup")) {
-		spec->fixup_list = NULL;
-		spec->fixup_id = -1;
-		return;
-	}
-
-	if (codec->modelname && models) {
-		while (models->name) {
-			if (!strcmp(codec->modelname, models->name)) {
-				id = models->id;
-				name = models->name;
-				break;
-			}
-			models++;
-		}
-	}
-	if (id < 0) {
-		q = snd_pci_quirk_lookup(codec->bus->pci, quirk);
-		if (q) {
-			id = q->value;
-#ifdef CONFIG_SND_DEBUG_VERBOSE
-			name = q->name;
-#endif
-		}
-	}
-	if (id < 0) {
-		for (q = quirk; q->subvendor; q++) {
-			unsigned int vendorid =
-				q->subdevice | (q->subvendor << 16);
-			if (vendorid == codec->subsystem_id) {
-				id = q->value;
-#ifdef CONFIG_SND_DEBUG_VERBOSE
-				name = q->name;
-#endif
-				break;
-			}
-		}
-	}
-
-	spec->fixup_id = id;
-	if (id >= 0) {
-		spec->fixup_list = fixlist;
-		spec->fixup_name = name;
-	}
-}
-
 /*
  * COEF access helper functions
  */
@@ -1621,8 +1445,7 @@ static void alc_auto_init_digital(struct hda_codec *codec)
 		pin = spec->autocfg.dig_out_pins[i];
 		if (!pin)
 			continue;
-		snd_hda_codec_write(codec, pin, 0,
-				    AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT);
+		snd_hda_set_pin_ctl(codec, pin, PIN_OUT);
 		if (!i)
 			dac = spec->multiout.dig_out_nid;
 		else
@@ -1635,9 +1458,7 @@ static void alc_auto_init_digital(struct hda_codec *codec)
 	}
 	pin = spec->autocfg.dig_in_pin;
 	if (pin)
-		snd_hda_codec_write(codec, pin, 0,
-				    AC_VERB_SET_PIN_WIDGET_CONTROL,
-				    PIN_IN);
+		snd_hda_set_pin_ctl(codec, pin, PIN_IN);
 }
 
 /* parse digital I/Os and set up NIDs in BIOS auto-parse mode */
@@ -2068,7 +1889,6 @@ static void alc_auto_init_std(struct hda_codec *codec);
 static int alc_init(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
-	unsigned int i;
 
 	if (spec->init_hook)
 		spec->init_hook(codec);
@@ -2076,8 +1896,6 @@ static int alc_init(struct hda_codec *codec)
 	alc_fix_pll(codec);
 	alc_auto_init_amp(codec, spec->init_amp);
 
-	for (i = 0; i < spec->num_init_verbs; i++)
-		snd_hda_sequence_write(codec, spec->init_verbs[i]);
 	alc_init_special_input_src(codec);
 	alc_auto_init_std(codec);
 
@@ -2725,7 +2543,6 @@ static int alc_auto_fill_adc_caps(struct hda_codec *codec)
 	nid = codec->start_nid;
 	for (i = 0; i < codec->num_nodes; i++, nid++) {
 		hda_nid_t src;
-		const hda_nid_t *list;
 		unsigned int caps = get_wcaps(codec, nid);
 		int type = get_wcaps_type(caps);
 
@@ -2743,13 +2560,14 @@ static int alc_auto_fill_adc_caps(struct hda_codec *codec)
 				cap_nids[nums] = src;
 				break;
 			}
-			n = snd_hda_get_conn_list(codec, src, &list);
+			n = snd_hda_get_num_conns(codec, src);
 			if (n > 1) {
 				cap_nids[nums] = src;
 				break;
 			} else if (n != 1)
 				break;
-			src = *list;
+			if (snd_hda_get_connections(codec, src, &src, 1) != 1)
+				break;
 		}
 		if (++nums >= max_nums)
 			break;
@@ -2856,8 +2674,7 @@ static int alc_auto_create_shared_input(struct hda_codec *codec)
 static void alc_set_pin_output(struct hda_codec *codec, hda_nid_t nid,
 			       unsigned int pin_type)
 {
-	snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
-			    pin_type);
+	snd_hda_set_pin_ctl(codec, nid, pin_type);
 	/* unmute pin */
 	if (nid_has_mute(codec, nid, HDA_OUTPUT))
 		snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE,
@@ -2891,7 +2708,7 @@ static void alc_auto_init_analog_input(struct hda_codec *codec)
 
 	/* mute all loopback inputs */
 	if (spec->mixer_nid) {
-		int nums = snd_hda_get_conn_list(codec, spec->mixer_nid, NULL);
+		int nums = snd_hda_get_num_conns(codec, spec->mixer_nid);
 		for (i = 0; i < nums; i++)
 			snd_hda_codec_write(codec, spec->mixer_nid, 0,
 					    AC_VERB_SET_AMP_GAIN_MUTE,
@@ -3521,7 +3338,7 @@ static int alc_auto_add_sw_ctl(struct hda_codec *codec,
 	if (wid_type == AC_WID_PIN || wid_type == AC_WID_AUD_OUT) {
 		type = ALC_CTL_WIDGET_MUTE;
 		val = HDA_COMPOSE_AMP_VAL(nid, chs, 0, HDA_OUTPUT);
-	} else if (snd_hda_get_conn_list(codec, nid, NULL) == 1) {
+	} else if (snd_hda_get_num_conns(codec, nid) == 1) {
 		type = ALC_CTL_WIDGET_MUTE;
 		val = HDA_COMPOSE_AMP_VAL(nid, chs, 0, HDA_INPUT);
 	} else {
@@ -3998,9 +3815,7 @@ static int alc_set_multi_io(struct hda_codec *codec, int idx, bool output)
 			snd_hda_codec_read(codec, nid, 0,
 					   AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
 	if (output) {
-		snd_hda_codec_update_cache(codec, nid, 0,
-					   AC_VERB_SET_PIN_WIDGET_CONTROL,
-					   PIN_OUT);
+		snd_hda_set_pin_ctl_cache(codec, nid, PIN_OUT);
 		if (get_wcaps(codec, nid) & AC_WCAP_OUT_AMP)
 			snd_hda_codec_amp_stereo(codec, nid, HDA_OUTPUT, 0,
 						 HDA_AMP_MUTE, 0);
@@ -4009,9 +3824,8 @@ static int alc_set_multi_io(struct hda_codec *codec, int idx, bool output)
 		if (get_wcaps(codec, nid) & AC_WCAP_OUT_AMP)
 			snd_hda_codec_amp_stereo(codec, nid, HDA_OUTPUT, 0,
 						 HDA_AMP_MUTE, HDA_AMP_MUTE);
-		snd_hda_codec_update_cache(codec, nid, 0,
-					   AC_VERB_SET_PIN_WIDGET_CONTROL,
-					   spec->multi_io[idx].ctl_in);
+		snd_hda_set_pin_ctl_cache(codec, nid,
+					  spec->multi_io[idx].ctl_in);
 	}
 	return 0;
 }
@@ -4084,7 +3898,7 @@ static void alc_remove_invalid_adc_nids(struct hda_codec *codec)
 	nums = 0;
 	for (n = 0; n < spec->num_adc_nids; n++) {
 		hda_nid_t cap = spec->private_capsrc_nids[n];
-		int num_conns = snd_hda_get_conn_list(codec, cap, NULL);
+		int num_conns = snd_hda_get_num_conns(codec, cap);
 		for (i = 0; i < imux->num_items; i++) {
 			hda_nid_t pin = spec->imux_pins[i];
 			if (pin) {
@@ -4213,7 +4027,7 @@ static void select_or_unmute_capsrc(struct hda_codec *codec, hda_nid_t cap,
 	if (get_wcaps_type(get_wcaps(codec, cap)) == AC_WID_AUD_MIX) {
 		snd_hda_codec_amp_stereo(codec, cap, HDA_INPUT, idx,
 					 HDA_AMP_MUTE, 0);
-	} else if (snd_hda_get_conn_list(codec, cap, NULL) > 1) {
+	} else if (snd_hda_get_num_conns(codec, cap) > 1) {
 		snd_hda_codec_write_cache(codec, cap, 0,
 					  AC_VERB_SET_CONNECT_SEL, idx);
 	}
@@ -4427,6 +4241,25 @@ static int alc_parse_auto_config(struct hda_codec *codec,
 	return 1;
 }
 
+/* common preparation job for alc_spec */
+static int alc_alloc_spec(struct hda_codec *codec, hda_nid_t mixer_nid)
+{
+	struct alc_spec *spec = kzalloc(sizeof(*spec), GFP_KERNEL);
+	int err;
+
+	if (!spec)
+		return -ENOMEM;
+	codec->spec = spec;
+	spec->mixer_nid = mixer_nid;
+
+	err = alc_codec_rename_from_preset(codec);
+	if (err < 0) {
+		kfree(spec);
+		return err;
+	}
+	return 0;
+}
+
 static int alc880_parse_auto_config(struct hda_codec *codec)
 {
 	static const hda_nid_t alc880_ignore[] = { 0x1d, 0 };
@@ -4808,13 +4641,11 @@ static int patch_alc880(struct hda_codec *codec)
 	struct alc_spec *spec;
 	int err;
 
-	spec = kzalloc(sizeof(*spec), GFP_KERNEL);
-	if (spec == NULL)
-		return -ENOMEM;
+	err = alc_alloc_spec(codec, 0x0b);
+	if (err < 0)
+		return err;
 
-	codec->spec = spec;
-
-	spec->mixer_nid = 0x0b;
+	spec = codec->spec;
 	spec->need_dac_fix = 1;
 
 	alc_pick_fixup(codec, alc880_fixup_models, alc880_fixup_tbl,
@@ -4890,7 +4721,7 @@ static void alc260_fixup_gpio1_toggle(struct hda_codec *codec,
 		spec->autocfg.hp_pins[0] = 0x0f; /* copy it for automute */
 		snd_hda_jack_detect_enable(codec, 0x0f, ALC_HP_EVENT);
 		spec->unsol_event = alc_sku_unsol_event;
-		add_verb(codec->spec, alc_gpio1_init_verbs);
+		snd_hda_gen_add_verbs(&spec->gen, alc_gpio1_init_verbs);
 	}
 }
 
@@ -5001,13 +4832,11 @@ static int patch_alc260(struct hda_codec *codec)
 	struct alc_spec *spec;
 	int err;
 
-	spec = kzalloc(sizeof(*spec), GFP_KERNEL);
-	if (spec == NULL)
-		return -ENOMEM;
+	err = alc_alloc_spec(codec, 0x07);
+	if (err < 0)
+		return err;
 
-	codec->spec = spec;
-
-	spec->mixer_nid = 0x07;
+	spec = codec->spec;
 
 	alc_pick_fixup(codec, NULL, alc260_fixup_tbl, alc260_fixups);
 	alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE);
@@ -5171,8 +5000,7 @@ static void alc889_fixup_mbp_vref(struct hda_codec *codec,
 		val = snd_hda_codec_read(codec, nids[i], 0,
 					 AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
 		val |= AC_PINCTL_VREF_80;
-		snd_hda_codec_write(codec, nids[i], 0,
-				    AC_VERB_SET_PIN_WIDGET_CONTROL, val);
+		snd_hda_set_pin_ctl(codec, nids[i], val);
 		spec->keep_vref_in_automute = 1;
 		break;
 	}
@@ -5193,8 +5021,7 @@ static void alc889_fixup_imac91_vref(struct hda_codec *codec,
 		val = snd_hda_codec_read(codec, nids[i], 0,
 					 AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
 		val |= AC_PINCTL_VREF_50;
-		snd_hda_codec_write(codec, nids[i], 0,
-				    AC_VERB_SET_PIN_WIDGET_CONTROL, val);
+		snd_hda_set_pin_ctl(codec, nids[i], val);
 	}
 	spec->keep_vref_in_automute = 1;
 }
@@ -5225,8 +5052,8 @@ static const struct alc_fixup alc882_fixups[] = {
 		}
 	},
 	[ALC882_FIXUP_ACER_ASPIRE_7736] = {
-		.type = ALC_FIXUP_SKU,
-		.v.sku = ALC_FIXUP_SKU_IGNORE,
+		.type = ALC_FIXUP_FUNC,
+		.v.func = alc_fixup_sku_ignore,
 	},
 	[ALC882_FIXUP_ASUS_W90V] = {
 		.type = ALC_FIXUP_PINS,
@@ -5476,13 +5303,11 @@ static int patch_alc882(struct hda_codec *codec)
 	struct alc_spec *spec;
 	int err;
 
-	spec = kzalloc(sizeof(*spec), GFP_KERNEL);
-	if (spec == NULL)
-		return -ENOMEM;
+	err = alc_alloc_spec(codec, 0x0b);
+	if (err < 0)
+		return err;
 
-	codec->spec = spec;
-
-	spec->mixer_nid = 0x0b;
+	spec = codec->spec;
 
 	switch (codec->vendor_id) {
 	case 0x10ec0882:
@@ -5494,10 +5319,6 @@ static int patch_alc882(struct hda_codec *codec)
 		break;
 	}
 
-	err = alc_codec_rename_from_preset(codec);
-	if (err < 0)
-		goto error;
-
 	alc_pick_fixup(codec, alc882_fixup_models, alc882_fixup_tbl,
 		       alc882_fixups);
 	alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE);
@@ -5621,13 +5442,11 @@ static int patch_alc262(struct hda_codec *codec)
 	struct alc_spec *spec;
 	int err;
 
-	spec = kzalloc(sizeof(*spec), GFP_KERNEL);
-	if (spec == NULL)
-		return -ENOMEM;
+	err = alc_alloc_spec(codec, 0x0b);
+	if (err < 0)
+		return err;
 
-	codec->spec = spec;
-
-	spec->mixer_nid = 0x0b;
+	spec = codec->spec;
 
 #if 0
 	/* pshou 07/11/05  set a zero PCM sample to DAC when FIFO is
@@ -5710,7 +5529,7 @@ static int alc268_parse_auto_config(struct hda_codec *codec)
 	if (err > 0) {
 		if (!spec->no_analog && spec->autocfg.speaker_pins[0] != 0x1d) {
 			add_mixer(spec, alc268_beep_mixer);
-			add_verb(spec, alc268_beep_init_verbs);
+			snd_hda_gen_add_verbs(&spec->gen, alc268_beep_init_verbs);
 		}
 	}
 	return err;
@@ -5723,13 +5542,12 @@ static int patch_alc268(struct hda_codec *codec)
 	struct alc_spec *spec;
 	int i, has_beep, err;
 
-	spec = kzalloc(sizeof(*spec), GFP_KERNEL);
-	if (spec == NULL)
-		return -ENOMEM;
-
-	codec->spec = spec;
-
 	/* ALC268 has no aa-loopback mixer */
+	err = alc_alloc_spec(codec, 0);
+	if (err < 0)
+		return err;
+
+	spec = codec->spec;
 
 	/* automatic parse from the BIOS config */
 	err = alc268_parse_auto_config(codec);
@@ -5946,9 +5764,7 @@ static void alc269_fixup_mic2_mute_hook(void *private_data, int enabled)
 {
 	struct hda_codec *codec = private_data;
 	unsigned int pinval = enabled ? 0x20 : 0x24;
-	snd_hda_codec_update_cache(codec, 0x19, 0,
-				   AC_VERB_SET_PIN_WIDGET_CONTROL,
-				   pinval);
+	snd_hda_set_pin_ctl_cache(codec, 0x19, pinval);
 }
 
 static void alc269_fixup_mic2_mute(struct hda_codec *codec,
@@ -6015,8 +5831,8 @@ static const struct alc_fixup alc269_fixups[] = {
 		}
 	},
 	[ALC269_FIXUP_SKU_IGNORE] = {
-		.type = ALC_FIXUP_SKU,
-		.v.sku = ALC_FIXUP_SKU_IGNORE,
+		.type = ALC_FIXUP_FUNC,
+		.v.func = alc_fixup_sku_ignore,
 	},
 	[ALC269_FIXUP_ASUS_G73JW] = {
 		.type = ALC_FIXUP_PINS,
@@ -6242,19 +6058,13 @@ static void alc269_fill_coef(struct hda_codec *codec)
 static int patch_alc269(struct hda_codec *codec)
 {
 	struct alc_spec *spec;
-	int err = 0;
+	int err;
 
-	spec = kzalloc(sizeof(*spec), GFP_KERNEL);
-	if (spec == NULL)
-		return -ENOMEM;
-
-	codec->spec = spec;
-
-	spec->mixer_nid = 0x0b;
-
-	err = alc_codec_rename_from_preset(codec);
+	err = alc_alloc_spec(codec, 0x0b);
 	if (err < 0)
-		goto error;
+		return err;
+
+	spec = codec->spec;
 
 	if (codec->vendor_id == 0x10ec0269) {
 		spec->codec_variant = ALC269_TYPE_ALC269VA;
@@ -6346,8 +6156,7 @@ static void alc861_fixup_asus_amp_vref_0f(struct hda_codec *codec,
 	if (!(val & (AC_PINCTL_IN_EN | AC_PINCTL_OUT_EN)))
 		val |= AC_PINCTL_IN_EN;
 	val |= AC_PINCTL_VREF_50;
-	snd_hda_codec_write(codec, 0x0f, 0,
-			    AC_VERB_SET_PIN_WIDGET_CONTROL, val);
+	snd_hda_set_pin_ctl(codec, 0x0f, val);
 	spec->keep_vref_in_automute = 1;
 }
 
@@ -6401,13 +6210,11 @@ static int patch_alc861(struct hda_codec *codec)
 	struct alc_spec *spec;
 	int err;
 
-	spec = kzalloc(sizeof(*spec), GFP_KERNEL);
-	if (spec == NULL)
-		return -ENOMEM;
+	err = alc_alloc_spec(codec, 0x15);
+	if (err < 0)
+		return err;
 
-	codec->spec = spec;
-
-	spec->mixer_nid = 0x15;
+	spec = codec->spec;
 
 	alc_pick_fixup(codec, NULL, alc861_fixup_tbl, alc861_fixups);
 	alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE);
@@ -6504,13 +6311,11 @@ static int patch_alc861vd(struct hda_codec *codec)
 	struct alc_spec *spec;
 	int err;
 
-	spec = kzalloc(sizeof(*spec), GFP_KERNEL);
-	if (spec == NULL)
-		return -ENOMEM;
+	err = alc_alloc_spec(codec, 0x0b);
+	if (err < 0)
+		return err;
 
-	codec->spec = spec;
-
-	spec->mixer_nid = 0x0b;
+	spec = codec->spec;
 
 	alc_pick_fixup(codec, NULL, alc861vd_fixup_tbl, alc861vd_fixups);
 	alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE);
@@ -6522,7 +6327,7 @@ static int patch_alc861vd(struct hda_codec *codec)
 
 	if (codec->vendor_id == 0x10ec0660) {
 		/* always turn on EAPD */
-		add_verb(spec, alc660vd_eapd_verbs);
+		snd_hda_gen_add_verbs(&spec->gen, alc660vd_eapd_verbs);
 	}
 
 	if (!spec->no_analog) {
@@ -6635,8 +6440,8 @@ static const struct alc_fixup alc662_fixups[] = {
 		}
 	},
 	[ALC662_FIXUP_SKU_IGNORE] = {
-		.type = ALC_FIXUP_SKU,
-		.v.sku = ALC_FIXUP_SKU_IGNORE,
+		.type = ALC_FIXUP_FUNC,
+		.v.func = alc_fixup_sku_ignore,
 	},
 	[ALC662_FIXUP_HP_RP5800] = {
 		.type = ALC_FIXUP_PINS,
@@ -6849,25 +6654,19 @@ static const struct alc_model_fixup alc662_fixup_models[] = {
 static int patch_alc662(struct hda_codec *codec)
 {
 	struct alc_spec *spec;
-	int err = 0;
+	int err;
 
-	spec = kzalloc(sizeof(*spec), GFP_KERNEL);
-	if (!spec)
-		return -ENOMEM;
+	err = alc_alloc_spec(codec, 0x0b);
+	if (err < 0)
+		return err;
 
-	codec->spec = spec;
-
-	spec->mixer_nid = 0x0b;
+	spec = codec->spec;
 
 	/* handle multiple HPs as is */
 	spec->parse_flags = HDA_PINCFG_NO_HP_FIXUP;
 
 	alc_fix_pll_init(codec, 0x20, 0x04, 15);
 
-	err = alc_codec_rename_from_preset(codec);
-	if (err < 0)
-		goto error;
-
 	if ((alc_get_coef0(codec) & (1 << 14)) &&
 	    codec->bus->pci->subsystem_vendor == 0x1025 &&
 	    spec->cdefine.platform_type == 1) {
@@ -6930,16 +6729,12 @@ static int alc680_parse_auto_config(struct hda_codec *codec)
  */
 static int patch_alc680(struct hda_codec *codec)
 {
-	struct alc_spec *spec;
 	int err;
 
-	spec = kzalloc(sizeof(*spec), GFP_KERNEL);
-	if (spec == NULL)
-		return -ENOMEM;
-
-	codec->spec = spec;
-
 	/* ALC680 has no aa-loopback mixer */
+	err = alc_alloc_spec(codec, 0);
+	if (err < 0)
+		return err;
 
 	/* automatic parse from the BIOS config */
 	err = alc680_parse_auto_config(codec);
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 2cb1e08f962a..7db8228f1b88 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -36,6 +36,7 @@
 #include <sound/tlv.h>
 #include "hda_codec.h"
 #include "hda_local.h"
+#include "hda_auto_parser.h"
 #include "hda_beep.h"
 #include "hda_jack.h"
 
@@ -221,6 +222,7 @@ struct sigmatel_spec {
 	unsigned char aloopback_shift;
 
 	/* power management */
+	unsigned int power_map_bits;
 	unsigned int num_pwrs;
 	const hda_nid_t *pwr_nids;
 	const hda_nid_t *dac_list;
@@ -314,6 +316,9 @@ struct sigmatel_spec {
 	struct hda_vmaster_mute_hook vmaster_mute;
 };
 
+#define AC_VERB_IDT_SET_POWER_MAP	0x7ec
+#define AC_VERB_IDT_GET_POWER_MAP	0xfec
+
 static const hda_nid_t stac9200_adc_nids[1] = {
         0x03,
 };
@@ -681,8 +686,7 @@ static int stac_vrefout_set(struct hda_codec *codec,
 	pinctl &= ~AC_PINCTL_VREFEN;
 	pinctl |= (new_vref & AC_PINCTL_VREFEN);
 
-	error = snd_hda_codec_write_cache(codec, nid, 0,
-					AC_VERB_SET_PIN_WIDGET_CONTROL, pinctl);
+	error = snd_hda_set_pin_ctl_cache(codec, nid, pinctl);
 	if (error < 0)
 		return error;
 
@@ -706,8 +710,7 @@ static unsigned int stac92xx_vref_set(struct hda_codec *codec,
 	else
 		pincfg |= AC_PINCTL_IN_EN;
 
-	error = snd_hda_codec_write_cache(codec, nid, 0,
-					AC_VERB_SET_PIN_WIDGET_CONTROL, pincfg);
+	error = snd_hda_set_pin_ctl_cache(codec, nid, pincfg);
 	if (error < 0)
 		return error;
 	else
@@ -2505,27 +2508,10 @@ static int stac92xx_build_pcms(struct hda_codec *codec)
 	return 0;
 }
 
-static unsigned int stac92xx_get_default_vref(struct hda_codec *codec,
-					hda_nid_t nid)
-{
-	unsigned int pincap = snd_hda_query_pin_caps(codec, nid);
-	pincap = (pincap & AC_PINCAP_VREF) >> AC_PINCAP_VREF_SHIFT;
-	if (pincap & AC_PINCAP_VREF_100)
-		return AC_PINCTL_VREF_100;
-	if (pincap & AC_PINCAP_VREF_80)
-		return AC_PINCTL_VREF_80;
-	if (pincap & AC_PINCAP_VREF_50)
-		return AC_PINCTL_VREF_50;
-	if (pincap & AC_PINCAP_VREF_GRD)
-		return AC_PINCTL_VREF_GRD;
-	return 0;
-}
-
 static void stac92xx_auto_set_pinctl(struct hda_codec *codec, hda_nid_t nid, int pin_type)
 
 {
-	snd_hda_codec_write_cache(codec, nid, 0,
-				  AC_VERB_SET_PIN_WIDGET_CONTROL, pin_type);
+	snd_hda_set_pin_ctl_cache(codec, nid, pin_type);
 }
 
 #define stac92xx_hp_switch_info		snd_ctl_boolean_mono_info
@@ -2594,7 +2580,7 @@ static int stac92xx_dc_bias_get(struct snd_kcontrol *kcontrol,
 	hda_nid_t nid = kcontrol->private_value;
 	unsigned int vref = stac92xx_vref_get(codec, nid);
 
-	if (vref == stac92xx_get_default_vref(codec, nid))
+	if (vref == snd_hda_get_default_vref(codec, nid))
 		ucontrol->value.enumerated.item[0] = 0;
 	else if (vref == AC_PINCTL_VREF_GRD)
 		ucontrol->value.enumerated.item[0] = 1;
@@ -2613,7 +2599,7 @@ static int stac92xx_dc_bias_put(struct snd_kcontrol *kcontrol,
 	hda_nid_t nid = kcontrol->private_value;
 
 	if (ucontrol->value.enumerated.item[0] == 0)
-		new_vref = stac92xx_get_default_vref(codec, nid);
+		new_vref = snd_hda_get_default_vref(codec, nid);
 	else if (ucontrol->value.enumerated.item[0] == 1)
 		new_vref = AC_PINCTL_VREF_GRD;
 	else if (ucontrol->value.enumerated.item[0] == 2)
@@ -2679,7 +2665,7 @@ static int stac92xx_io_switch_put(struct snd_kcontrol *kcontrol, struct snd_ctl_
 	else {
 		unsigned int pinctl = AC_PINCTL_IN_EN;
 		if (io_idx) /* set VREF for mic */
-			pinctl |= stac92xx_get_default_vref(codec, nid);
+			pinctl |= snd_hda_get_default_vref(codec, nid);
 		stac92xx_auto_set_pinctl(codec, nid, pinctl);
 	}
 
@@ -2847,7 +2833,7 @@ static inline int stac92xx_add_jack_mode_control(struct hda_codec *codec,
 	char name[22];
 
 	if (snd_hda_get_input_pin_attr(def_conf) != INPUT_PIN_ATTR_INT) {
-		if (stac92xx_get_default_vref(codec, nid) == AC_PINCTL_VREF_GRD
+		if (snd_hda_get_default_vref(codec, nid) == AC_PINCTL_VREF_GRD
 			&& nid == spec->line_switch)
 			control = STAC_CTL_WIDGET_IO_SWITCH;
 		else if (snd_hda_query_pin_caps(codec, nid)
@@ -4250,13 +4236,6 @@ static void stac_store_hints(struct hda_codec *codec)
 	val = snd_hda_get_bool_hint(codec, "eapd_switch");
 	if (val >= 0)
 		spec->eapd_switch = val;
-	get_int_hint(codec, "gpio_led_polarity", &spec->gpio_led_polarity);
-	if (get_int_hint(codec, "gpio_led", &spec->gpio_led)) {
-		spec->gpio_mask |= spec->gpio_led;
-		spec->gpio_dir |= spec->gpio_led;
-		if (spec->gpio_led_polarity)
-			spec->gpio_data |= spec->gpio_led;
-	}
 }
 
 static void stac_issue_unsol_events(struct hda_codec *codec, int num_pins,
@@ -4354,7 +4333,7 @@ static int stac92xx_init(struct hda_codec *codec)
 		unsigned int pinctl, conf;
 		if (type == AUTO_PIN_MIC) {
 			/* for mic pins, force to initialize */
-			pinctl = stac92xx_get_default_vref(codec, nid);
+			pinctl = snd_hda_get_default_vref(codec, nid);
 			pinctl |= AC_PINCTL_IN_EN;
 			stac92xx_auto_set_pinctl(codec, nid, pinctl);
 		} else {
@@ -4390,10 +4369,18 @@ static int stac92xx_init(struct hda_codec *codec)
 		hda_nid_t nid = spec->pwr_nids[i];
 		int pinctl, def_conf;
 
+		def_conf = snd_hda_codec_get_pincfg(codec, nid);
+		def_conf = get_defcfg_connect(def_conf);
+		if (def_conf == AC_JACK_PORT_NONE) {
+			/* power off unused ports */
+			stac_toggle_power_map(codec, nid, 0);
+			continue;
+		}
 		/* power on when no jack detection is available */
 		/* or when the VREF is used for controlling LED */
 		if (!spec->hp_detect ||
-		    spec->vref_mute_led_nid == nid) {
+		    spec->vref_mute_led_nid == nid ||
+		    !is_jack_detectable(codec, nid)) {
 			stac_toggle_power_map(codec, nid, 1);
 			continue;
 		}
@@ -4411,15 +4398,6 @@ static int stac92xx_init(struct hda_codec *codec)
 			stac_toggle_power_map(codec, nid, 1);
 			continue;
 		}
-		def_conf = snd_hda_codec_get_pincfg(codec, nid);
-		def_conf = get_defcfg_connect(def_conf);
-		/* skip any ports that don't have jacks since presence
- 		 * detection is useless */
-		if (def_conf != AC_JACK_PORT_NONE &&
-		    !is_jack_detectable(codec, nid)) {
-			stac_toggle_power_map(codec, nid, 1);
-			continue;
-		}
 		if (enable_pin_detect(codec, nid, STAC_PWR_EVENT)) {
 			stac_issue_unsol_event(codec, nid);
 			continue;
@@ -4432,6 +4410,12 @@ static int stac92xx_init(struct hda_codec *codec)
 
 	/* sync mute LED */
 	snd_hda_sync_vmaster_hook(&spec->vmaster_mute);
+
+	/* sync the power-map */
+	if (spec->num_pwrs)
+		snd_hda_codec_write(codec, codec->afg, 0,
+				    AC_VERB_IDT_SET_POWER_MAP,
+				    spec->power_map_bits);
 	if (spec->dac_list)
 		stac92xx_power_down(codec);
 	return 0;
@@ -4460,8 +4444,7 @@ static void stac92xx_shutup_pins(struct hda_codec *codec)
 		struct hda_pincfg *pin = snd_array_elem(&codec->init_pins, i);
 		def_conf = snd_hda_codec_get_pincfg(codec, pin->nid);
 		if (get_defcfg_connect(def_conf) != AC_JACK_PORT_NONE)
-			snd_hda_codec_write(codec, pin->nid, 0,
-				    AC_VERB_SET_PIN_WIDGET_CONTROL, 0);
+			snd_hda_set_pin_ctl(codec, pin->nid, 0);
 	}
 }
 
@@ -4517,9 +4500,7 @@ static void stac92xx_set_pinctl(struct hda_codec *codec, hda_nid_t nid,
 	
 	pin_ctl |= flag;
 	if (old_ctl != pin_ctl)
-		snd_hda_codec_write_cache(codec, nid, 0,
-					  AC_VERB_SET_PIN_WIDGET_CONTROL,
-					  pin_ctl);
+		snd_hda_set_pin_ctl_cache(codec, nid, pin_ctl);
 }
 
 static void stac92xx_reset_pinctl(struct hda_codec *codec, hda_nid_t nid,
@@ -4528,9 +4509,7 @@ static void stac92xx_reset_pinctl(struct hda_codec *codec, hda_nid_t nid,
 	unsigned int pin_ctl = snd_hda_codec_read(codec, nid,
 			0, AC_VERB_GET_PIN_WIDGET_CONTROL, 0x00);
 	if (pin_ctl & flag)
-		snd_hda_codec_write_cache(codec, nid, 0,
-					  AC_VERB_SET_PIN_WIDGET_CONTROL,
-					  pin_ctl & ~flag);
+		snd_hda_set_pin_ctl_cache(codec, nid, pin_ctl & ~flag);
 }
 
 static inline int get_pin_presence(struct hda_codec *codec, hda_nid_t nid)
@@ -4682,14 +4661,18 @@ static void stac_toggle_power_map(struct hda_codec *codec, hda_nid_t nid,
 
 	idx = 1 << idx;
 
-	val = snd_hda_codec_read(codec, codec->afg, 0, 0x0fec, 0x0) & 0xff;
+	val = spec->power_map_bits;
 	if (enable)
 		val &= ~idx;
 	else
 		val |= idx;
 
 	/* power down unused output ports */
-	snd_hda_codec_write(codec, codec->afg, 0, 0x7ec, val);
+	if (val != spec->power_map_bits) {
+		spec->power_map_bits = val;
+		snd_hda_codec_write(codec, codec->afg, 0,
+				    AC_VERB_IDT_SET_POWER_MAP, val);
+	}
 }
 
 static void stac92xx_pin_sense(struct hda_codec *codec, hda_nid_t nid)
@@ -4866,6 +4849,11 @@ static int find_mute_led_cfg(struct hda_codec *codec, int default_polarity)
 	struct sigmatel_spec *spec = codec->spec;
 	const struct dmi_device *dev = NULL;
 
+	if (get_int_hint(codec, "gpio_led", &spec->gpio_led)) {
+		get_int_hint(codec, "gpio_led_polarity",
+			     &spec->gpio_led_polarity);
+		return 1;
+	}
 	if ((codec->subsystem_id >> 16) == PCI_VENDOR_ID_HP) {
 		while ((dev = dmi_find_device(DMI_DEV_TYPE_OEM_STRING,
 								NULL, dev))) {
@@ -4952,7 +4940,8 @@ static void stac92hd_proc_hook(struct snd_info_buffer *buffer,
 {
 	if (nid == codec->afg)
 		snd_iprintf(buffer, "Power-Map: 0x%02x\n", 
-			    snd_hda_codec_read(codec, nid, 0, 0x0fec, 0x0));
+			    snd_hda_codec_read(codec, nid, 0,
+					       AC_VERB_IDT_GET_POWER_MAP, 0));
 }
 
 static void analog_loop_proc_hook(struct snd_info_buffer *buffer,
@@ -5009,20 +4998,6 @@ static int stac92xx_suspend(struct hda_codec *codec, pm_message_t state)
 	return 0;
 }
 
-static int stac92xx_pre_resume(struct hda_codec *codec)
-{
-	struct sigmatel_spec *spec = codec->spec;
-
-	/* sync mute LED */
-	if (spec->vref_mute_led_nid)
-		stac_vrefout_set(codec, spec->vref_mute_led_nid,
-				 spec->vref_led);
-	else if (spec->gpio_led)
-		stac_gpio_set(codec, spec->gpio_mask,
-			      spec->gpio_dir, spec->gpio_data);
-	return 0;
-}
-
 static void stac92xx_set_power_state(struct hda_codec *codec, hda_nid_t fg,
 				unsigned int power_state)
 {
@@ -5046,7 +5021,6 @@ static void stac92xx_set_power_state(struct hda_codec *codec, hda_nid_t fg,
 #else
 #define stac92xx_suspend	NULL
 #define stac92xx_resume		NULL
-#define stac92xx_pre_resume	NULL
 #define stac92xx_set_power_state NULL
 #endif /* CONFIG_PM */
 
@@ -5592,9 +5566,6 @@ again:
 			codec->patch_ops.set_power_state =
 					stac92xx_set_power_state;
 		}
-#ifdef CONFIG_PM
-		codec->patch_ops.pre_resume = stac92xx_pre_resume;
-#endif
 	}
 
 	err = stac92xx_parse_auto_config(codec);
@@ -5901,9 +5872,6 @@ again:
 			codec->patch_ops.set_power_state =
 					stac92xx_set_power_state;
 		}
-#ifdef CONFIG_PM
-		codec->patch_ops.pre_resume = stac92xx_pre_resume;
-#endif
 	}
 
 	spec->multiout.dac_nids = spec->dac_nids;
diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c
index 06214fdc9486..82b368068e08 100644
--- a/sound/pci/hda/patch_via.c
+++ b/sound/pci/hda/patch_via.c
@@ -54,6 +54,7 @@
 #include <sound/asoundef.h>
 #include "hda_codec.h"
 #include "hda_local.h"
+#include "hda_auto_parser.h"
 #include "hda_jack.h"
 
 /* Pin Widget NID */
@@ -484,7 +485,7 @@ static void activate_output_mix(struct hda_codec *codec, struct nid_path *path,
 
 	if (!path)
 		return;
-	num = snd_hda_get_conn_list(codec, mix_nid, NULL);
+	num = snd_hda_get_num_conns(codec, mix_nid);
 	for (i = 0; i < num; i++) {
 		if (i == idx)
 			val = AMP_IN_UNMUTE(i);
@@ -532,8 +533,7 @@ static void init_output_pin(struct hda_codec *codec, hda_nid_t pin,
 {
 	if (!pin)
 		return;
-	snd_hda_codec_write(codec, pin, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
-			    pin_type);
+	snd_hda_set_pin_ctl(codec, pin, pin_type);
 	if (snd_hda_query_pin_caps(codec, pin) & AC_PINCAP_EAPD)
 		snd_hda_codec_write(codec, pin, 0,
 				    AC_VERB_SET_EAPD_BTLENABLE, 0x02);
@@ -662,12 +662,12 @@ static void via_auto_init_analog_input(struct hda_codec *codec)
 		hda_nid_t nid = cfg->inputs[i].pin;
 		if (spec->smart51_enabled && is_smart51_pins(codec, nid))
 			ctl = PIN_OUT;
-		else if (cfg->inputs[i].type == AUTO_PIN_MIC)
-			ctl = PIN_VREF50;
-		else
+		else {
 			ctl = PIN_IN;
-		snd_hda_codec_write(codec, nid, 0,
-				    AC_VERB_SET_PIN_WIDGET_CONTROL, ctl);
+			if (cfg->inputs[i].type == AUTO_PIN_MIC)
+				ctl |= snd_hda_get_default_vref(codec, nid);
+		}
+		snd_hda_set_pin_ctl(codec, nid, ctl);
 	}
 
 	/* init input-src */
@@ -1006,9 +1006,7 @@ static int via_smart51_put(struct snd_kcontrol *kcontrol,
 					  AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
 		parm &= ~(AC_PINCTL_IN_EN | AC_PINCTL_OUT_EN);
 		parm |= out_in;
-		snd_hda_codec_write(codec, nid, 0,
-				    AC_VERB_SET_PIN_WIDGET_CONTROL,
-				    parm);
+		snd_hda_set_pin_ctl(codec, nid, parm);
 		if (out_in == AC_PINCTL_OUT_EN) {
 			mute_aa_path(codec, 1);
 			notify_aa_path_ctls(codec);
@@ -1647,8 +1645,7 @@ static void toggle_output_mutes(struct hda_codec *codec, int num_pins,
 			parm &= ~AC_PINCTL_OUT_EN;
 		else
 			parm |= AC_PINCTL_OUT_EN;
-		snd_hda_codec_write(codec, pins[i], 0,
-				    AC_VERB_SET_PIN_WIDGET_CONTROL, parm);
+		snd_hda_set_pin_ctl(codec, pins[i], parm);
 	}
 }
 
@@ -1709,8 +1706,7 @@ static void via_gpio_control(struct hda_codec *codec)
 
 	if (gpio_data == 0x02) {
 		/* unmute line out */
-		snd_hda_codec_write(codec, spec->autocfg.line_out_pins[0], 0,
-				    AC_VERB_SET_PIN_WIDGET_CONTROL,
+		snd_hda_set_pin_ctl(codec, spec->autocfg.line_out_pins[0],
 				    PIN_OUT);
 		if (vol_counter & 0x20) {
 			/* decrease volume */
@@ -1728,9 +1724,7 @@ static void via_gpio_control(struct hda_codec *codec)
 		}
 	} else if (!(gpio_data & 0x02)) {
 		/* mute line out */
-		snd_hda_codec_write(codec, spec->autocfg.line_out_pins[0], 0,
-				    AC_VERB_SET_PIN_WIDGET_CONTROL,
-				    0);
+		snd_hda_set_pin_ctl(codec, spec->autocfg.line_out_pins[0], 0);
 	}
 }
 
@@ -2757,8 +2751,7 @@ static void via_auto_init_dig_in(struct hda_codec *codec)
 	struct via_spec *spec = codec->spec;
 	if (!spec->dig_in_nid)
 		return;
-	snd_hda_codec_write(codec, spec->autocfg.dig_in_pin, 0,
-			    AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN);
+	snd_hda_set_pin_ctl(codec, spec->autocfg.dig_in_pin, PIN_IN);
 }
 
 /* initialize the unsolicited events */
diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c
index 132a86e09d07..5be2e120a14e 100644
--- a/sound/pci/ice1712/ice1712.c
+++ b/sound/pci/ice1712/ice1712.c
@@ -2803,22 +2803,11 @@ static void __devexit snd_ice1712_remove(struct pci_dev *pci)
 	pci_set_drvdata(pci, NULL);
 }
 
-static struct pci_driver driver = {
+static struct pci_driver ice1712_driver = {
 	.name = KBUILD_MODNAME,
 	.id_table = snd_ice1712_ids,
 	.probe = snd_ice1712_probe,
 	.remove = __devexit_p(snd_ice1712_remove),
 };
 
-static int __init alsa_card_ice1712_init(void)
-{
-	return pci_register_driver(&driver);
-}
-
-static void __exit alsa_card_ice1712_exit(void)
-{
-	pci_unregister_driver(&driver);
-}
-
-module_init(alsa_card_ice1712_init)
-module_exit(alsa_card_ice1712_exit)
+module_pci_driver(ice1712_driver);
diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c
index 812d10e43ae0..a01a00d1cf4d 100644
--- a/sound/pci/ice1712/ice1724.c
+++ b/sound/pci/ice1712/ice1724.c
@@ -2873,7 +2873,7 @@ static int snd_vt1724_resume(struct pci_dev *pci)
 }
 #endif
 
-static struct pci_driver driver = {
+static struct pci_driver vt1724_driver = {
 	.name = KBUILD_MODNAME,
 	.id_table = snd_vt1724_ids,
 	.probe = snd_vt1724_probe,
@@ -2884,15 +2884,4 @@ static struct pci_driver driver = {
 #endif
 };
 
-static int __init alsa_card_ice1724_init(void)
-{
-	return pci_register_driver(&driver);
-}
-
-static void __exit alsa_card_ice1724_exit(void)
-{
-	pci_unregister_driver(&driver);
-}
-
-module_init(alsa_card_ice1724_init)
-module_exit(alsa_card_ice1724_exit)
+module_pci_driver(vt1724_driver);
diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c
index e0a4263baa20..f4e2dd4da8cf 100644
--- a/sound/pci/intel8x0.c
+++ b/sound/pci/intel8x0.c
@@ -3338,7 +3338,7 @@ static void __devexit snd_intel8x0_remove(struct pci_dev *pci)
 	pci_set_drvdata(pci, NULL);
 }
 
-static struct pci_driver driver = {
+static struct pci_driver intel8x0_driver = {
 	.name = KBUILD_MODNAME,
 	.id_table = snd_intel8x0_ids,
 	.probe = snd_intel8x0_probe,
@@ -3349,16 +3349,4 @@ static struct pci_driver driver = {
 #endif
 };
 
-
-static int __init alsa_card_intel8x0_init(void)
-{
-	return pci_register_driver(&driver);
-}
-
-static void __exit alsa_card_intel8x0_exit(void)
-{
-	pci_unregister_driver(&driver);
-}
-
-module_init(alsa_card_intel8x0_init)
-module_exit(alsa_card_intel8x0_exit)
+module_pci_driver(intel8x0_driver);
diff --git a/sound/pci/intel8x0m.c b/sound/pci/intel8x0m.c
index d689913a61be..fc27a6a69e77 100644
--- a/sound/pci/intel8x0m.c
+++ b/sound/pci/intel8x0m.c
@@ -1324,7 +1324,7 @@ static void __devexit snd_intel8x0m_remove(struct pci_dev *pci)
 	pci_set_drvdata(pci, NULL);
 }
 
-static struct pci_driver driver = {
+static struct pci_driver intel8x0m_driver = {
 	.name = KBUILD_MODNAME,
 	.id_table = snd_intel8x0m_ids,
 	.probe = snd_intel8x0m_probe,
@@ -1335,16 +1335,4 @@ static struct pci_driver driver = {
 #endif
 };
 
-
-static int __init alsa_card_intel8x0m_init(void)
-{
-	return pci_register_driver(&driver);
-}
-
-static void __exit alsa_card_intel8x0m_exit(void)
-{
-	pci_unregister_driver(&driver);
-}
-
-module_init(alsa_card_intel8x0m_init)
-module_exit(alsa_card_intel8x0m_exit)
+module_pci_driver(intel8x0m_driver);
diff --git a/sound/pci/korg1212/korg1212.c b/sound/pci/korg1212/korg1212.c
index 8fea45ab5882..e69ce5f9c31e 100644
--- a/sound/pci/korg1212/korg1212.c
+++ b/sound/pci/korg1212/korg1212.c
@@ -2476,22 +2476,11 @@ static void __devexit snd_korg1212_remove(struct pci_dev *pci)
 	pci_set_drvdata(pci, NULL);
 }
 
-static struct pci_driver driver = {
+static struct pci_driver korg1212_driver = {
 	.name = KBUILD_MODNAME,
 	.id_table = snd_korg1212_ids,
 	.probe = snd_korg1212_probe,
 	.remove = __devexit_p(snd_korg1212_remove),
 };
 
-static int __init alsa_card_korg1212_init(void)
-{
-	return pci_register_driver(&driver);
-}
-
-static void __exit alsa_card_korg1212_exit(void)
-{
-	pci_unregister_driver(&driver);
-}
-
-module_init(alsa_card_korg1212_init)
-module_exit(alsa_card_korg1212_exit)
+module_pci_driver(korg1212_driver);
diff --git a/sound/pci/lola/lola.c b/sound/pci/lola/lola.c
index 375982736858..ac15166bee68 100644
--- a/sound/pci/lola/lola.c
+++ b/sound/pci/lola/lola.c
@@ -770,22 +770,11 @@ static DEFINE_PCI_DEVICE_TABLE(lola_ids) = {
 MODULE_DEVICE_TABLE(pci, lola_ids);
 
 /* pci_driver definition */
-static struct pci_driver driver = {
+static struct pci_driver lola_driver = {
 	.name = KBUILD_MODNAME,
 	.id_table = lola_ids,
 	.probe = lola_probe,
 	.remove = __devexit_p(lola_remove),
 };
 
-static int __init alsa_card_lola_init(void)
-{
-	return pci_register_driver(&driver);
-}
-
-static void __exit alsa_card_lola_exit(void)
-{
-	pci_unregister_driver(&driver);
-}
-
-module_init(alsa_card_lola_init)
-module_exit(alsa_card_lola_exit)
+module_pci_driver(lola_driver);
diff --git a/sound/pci/lx6464es/lx6464es.c b/sound/pci/lx6464es/lx6464es.c
index d94c0c292bd0..d1ab43706735 100644
--- a/sound/pci/lx6464es/lx6464es.c
+++ b/sound/pci/lx6464es/lx6464es.c
@@ -1141,24 +1141,11 @@ static void __devexit snd_lx6464es_remove(struct pci_dev *pci)
 }
 
 
-static struct pci_driver driver = {
+static struct pci_driver lx6464es_driver = {
 	.name =     KBUILD_MODNAME,
 	.id_table = snd_lx6464es_ids,
 	.probe =    snd_lx6464es_probe,
 	.remove = __devexit_p(snd_lx6464es_remove),
 };
 
-
-/* module initialization */
-static int __init mod_init(void)
-{
-	return pci_register_driver(&driver);
-}
-
-static void __exit mod_exit(void)
-{
-	pci_unregister_driver(&driver);
-}
-
-module_init(mod_init);
-module_exit(mod_exit);
+module_pci_driver(lx6464es_driver);
diff --git a/sound/pci/maestro3.c b/sound/pci/maestro3.c
index 78229b0dad2b..deef21399586 100644
--- a/sound/pci/maestro3.c
+++ b/sound/pci/maestro3.c
@@ -2837,7 +2837,7 @@ static void __devexit snd_m3_remove(struct pci_dev *pci)
 	pci_set_drvdata(pci, NULL);
 }
 
-static struct pci_driver driver = {
+static struct pci_driver m3_driver = {
 	.name = KBUILD_MODNAME,
 	.id_table = snd_m3_ids,
 	.probe = snd_m3_probe,
@@ -2848,15 +2848,4 @@ static struct pci_driver driver = {
 #endif
 };
 	
-static int __init alsa_card_m3_init(void)
-{
-	return pci_register_driver(&driver);
-}
-
-static void __exit alsa_card_m3_exit(void)
-{
-	pci_unregister_driver(&driver);
-}
-
-module_init(alsa_card_m3_init)
-module_exit(alsa_card_m3_exit)
+module_pci_driver(m3_driver);
diff --git a/sound/pci/mixart/mixart.c b/sound/pci/mixart/mixart.c
index 487837c01c9f..0762610c99c0 100644
--- a/sound/pci/mixart/mixart.c
+++ b/sound/pci/mixart/mixart.c
@@ -1380,22 +1380,11 @@ static void __devexit snd_mixart_remove(struct pci_dev *pci)
 	pci_set_drvdata(pci, NULL);
 }
 
-static struct pci_driver driver = {
+static struct pci_driver mixart_driver = {
 	.name = KBUILD_MODNAME,
 	.id_table = snd_mixart_ids,
 	.probe = snd_mixart_probe,
 	.remove = __devexit_p(snd_mixart_remove),
 };
 
-static int __init alsa_card_mixart_init(void)
-{
-	return pci_register_driver(&driver);
-}
-
-static void __exit alsa_card_mixart_exit(void)
-{
-	pci_unregister_driver(&driver);
-}
-
-module_init(alsa_card_mixart_init)
-module_exit(alsa_card_mixart_exit)
+module_pci_driver(mixart_driver);
diff --git a/sound/pci/nm256/nm256.c b/sound/pci/nm256/nm256.c
index ade2c64bd606..8159b05ee94d 100644
--- a/sound/pci/nm256/nm256.c
+++ b/sound/pci/nm256/nm256.c
@@ -1742,7 +1742,7 @@ static void __devexit snd_nm256_remove(struct pci_dev *pci)
 }
 
 
-static struct pci_driver driver = {
+static struct pci_driver nm256_driver = {
 	.name = KBUILD_MODNAME,
 	.id_table = snd_nm256_ids,
 	.probe = snd_nm256_probe,
@@ -1753,16 +1753,4 @@ static struct pci_driver driver = {
 #endif
 };
 
-
-static int __init alsa_card_nm256_init(void)
-{
-	return pci_register_driver(&driver);
-}
-
-static void __exit alsa_card_nm256_exit(void)
-{
-	pci_unregister_driver(&driver);
-}
-
-module_init(alsa_card_nm256_init)
-module_exit(alsa_card_nm256_exit)
+module_pci_driver(nm256_driver);
diff --git a/sound/pci/oxygen/oxygen.c b/sound/pci/oxygen/oxygen.c
index eab663eef117..610275bfbaeb 100644
--- a/sound/pci/oxygen/oxygen.c
+++ b/sound/pci/oxygen/oxygen.c
@@ -94,6 +94,7 @@ enum {
 	MODEL_2CH_OUTPUT,
 	MODEL_HG2PCI,
 	MODEL_XONAR_DG,
+	MODEL_XONAR_DGX,
 };
 
 static DEFINE_PCI_DEVICE_TABLE(oxygen_ids) = {
@@ -109,6 +110,8 @@ static DEFINE_PCI_DEVICE_TABLE(oxygen_ids) = {
 	{ OXYGEN_PCI_SUBID(0x1a58, 0x0910), .driver_data = MODEL_CMEDIA_REF },
 	/* Asus Xonar DG */
 	{ OXYGEN_PCI_SUBID(0x1043, 0x8467), .driver_data = MODEL_XONAR_DG },
+	/* Asus Xonar DGX */
+	{ OXYGEN_PCI_SUBID(0x1043, 0x8521), .driver_data = MODEL_XONAR_DGX },
 	/* PCI 2.0 HD Audio */
 	{ OXYGEN_PCI_SUBID(0x13f6, 0x8782), .driver_data = MODEL_2CH_OUTPUT },
 	/* Kuroutoshikou CMI8787-HG2PCI */
@@ -827,6 +830,11 @@ static int __devinit get_oxygen_model(struct oxygen *chip,
 		break;
 	case MODEL_XONAR_DG:
 		chip->model = model_xonar_dg;
+		chip->model.shortname = "Xonar DG";
+		break;
+	case MODEL_XONAR_DGX:
+		chip->model = model_xonar_dg;
+		chip->model.shortname = "Xonar DGX";
 		break;
 	}
 	if (id->driver_data == MODEL_MERIDIAN ||
@@ -870,15 +878,4 @@ static struct pci_driver oxygen_driver = {
 #endif
 };
 
-static int __init alsa_card_oxygen_init(void)
-{
-	return pci_register_driver(&oxygen_driver);
-}
-
-static void __exit alsa_card_oxygen_exit(void)
-{
-	pci_unregister_driver(&oxygen_driver);
-}
-
-module_init(alsa_card_oxygen_init)
-module_exit(alsa_card_oxygen_exit)
+module_pci_driver(oxygen_driver);
diff --git a/sound/pci/oxygen/virtuoso.c b/sound/pci/oxygen/virtuoso.c
index 3fdee4950174..19962c6d38c3 100644
--- a/sound/pci/oxygen/virtuoso.c
+++ b/sound/pci/oxygen/virtuoso.c
@@ -100,15 +100,4 @@ static struct pci_driver xonar_driver = {
 	.shutdown = oxygen_pci_shutdown,
 };
 
-static int __init alsa_card_xonar_init(void)
-{
-	return pci_register_driver(&xonar_driver);
-}
-
-static void __exit alsa_card_xonar_exit(void)
-{
-	pci_unregister_driver(&xonar_driver);
-}
-
-module_init(alsa_card_xonar_init)
-module_exit(alsa_card_xonar_exit)
+module_pci_driver(xonar_driver);
diff --git a/sound/pci/oxygen/xonar_dg.c b/sound/pci/oxygen/xonar_dg.c
index 793bdf03d7e0..77acd790ea47 100644
--- a/sound/pci/oxygen/xonar_dg.c
+++ b/sound/pci/oxygen/xonar_dg.c
@@ -1,5 +1,5 @@
 /*
- * card driver for the Xonar DG
+ * card driver for the Xonar DG/DGX
  *
  * Copyright (c) Clemens Ladisch <clemens@ladisch.de>
  *
@@ -17,8 +17,8 @@
  */
 
 /*
- * Xonar DG
- * --------
+ * Xonar DG/DGX
+ * ------------
  *
  * CMI8788:
  *
@@ -581,7 +581,6 @@ static void dump_cs4245_registers(struct oxygen *chip,
 }
 
 struct oxygen_model model_xonar_dg = {
-	.shortname = "Xonar DG",
 	.longname = "C-Media Oxygen HD Audio",
 	.chip = "CMI8786",
 	.init = dg_init,
diff --git a/sound/pci/pcxhr/pcxhr.c b/sound/pci/pcxhr/pcxhr.c
index fd1809ab73b4..0435f45e9513 100644
--- a/sound/pci/pcxhr/pcxhr.c
+++ b/sound/pci/pcxhr/pcxhr.c
@@ -1607,22 +1607,11 @@ static void __devexit pcxhr_remove(struct pci_dev *pci)
 	pci_set_drvdata(pci, NULL);
 }
 
-static struct pci_driver driver = {
+static struct pci_driver pcxhr_driver = {
 	.name = KBUILD_MODNAME,
 	.id_table = pcxhr_ids,
 	.probe = pcxhr_probe,
 	.remove = __devexit_p(pcxhr_remove),
 };
 
-static int __init pcxhr_module_init(void)
-{
-	return pci_register_driver(&driver);
-}
-
-static void __exit pcxhr_module_exit(void)
-{
-	pci_unregister_driver(&driver);
-}
-
-module_init(pcxhr_module_init)
-module_exit(pcxhr_module_exit)
+module_pci_driver(pcxhr_driver);
diff --git a/sound/pci/riptide/riptide.c b/sound/pci/riptide/riptide.c
index 0481d94aac9b..cbeb3f77350c 100644
--- a/sound/pci/riptide/riptide.c
+++ b/sound/pci/riptide/riptide.c
@@ -1837,8 +1837,7 @@ static int snd_riptide_free(struct snd_riptide *chip)
 	}
 	if (chip->irq >= 0)
 		free_irq(chip->irq, chip);
-	if (chip->fw_entry)
-		release_firmware(chip->fw_entry);
+	release_firmware(chip->fw_entry);
 	release_and_free_resource(chip->res_port);
 	kfree(chip);
 	return 0;
diff --git a/sound/pci/rme32.c b/sound/pci/rme32.c
index b4819d5e41db..46b3629dda22 100644
--- a/sound/pci/rme32.c
+++ b/sound/pci/rme32.c
@@ -1984,22 +1984,11 @@ static void __devexit snd_rme32_remove(struct pci_dev *pci)
 	pci_set_drvdata(pci, NULL);
 }
 
-static struct pci_driver driver = {
+static struct pci_driver rme32_driver = {
 	.name =		KBUILD_MODNAME,
 	.id_table =	snd_rme32_ids,
 	.probe =	snd_rme32_probe,
 	.remove =	__devexit_p(snd_rme32_remove),
 };
 
-static int __init alsa_card_rme32_init(void)
-{
-	return pci_register_driver(&driver);
-}
-
-static void __exit alsa_card_rme32_exit(void)
-{
-	pci_unregister_driver(&driver);
-}
-
-module_init(alsa_card_rme32_init)
-module_exit(alsa_card_rme32_exit)
+module_pci_driver(rme32_driver);
diff --git a/sound/pci/rme96.c b/sound/pci/rme96.c
index ba894158e76c..9b98dc406988 100644
--- a/sound/pci/rme96.c
+++ b/sound/pci/rme96.c
@@ -2395,22 +2395,11 @@ static void __devexit snd_rme96_remove(struct pci_dev *pci)
 	pci_set_drvdata(pci, NULL);
 }
 
-static struct pci_driver driver = {
+static struct pci_driver rme96_driver = {
 	.name = KBUILD_MODNAME,
 	.id_table = snd_rme96_ids,
 	.probe = snd_rme96_probe,
 	.remove = __devexit_p(snd_rme96_remove),
 };
 
-static int __init alsa_card_rme96_init(void)
-{
-	return pci_register_driver(&driver);
-}
-
-static void __exit alsa_card_rme96_exit(void)
-{
-	pci_unregister_driver(&driver);
-}
-
-module_init(alsa_card_rme96_init)
-module_exit(alsa_card_rme96_exit)
+module_pci_driver(rme96_driver);
diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c
index 0b2aea2ce172..0d6930c4f4b7 100644
--- a/sound/pci/rme9652/hdsp.c
+++ b/sound/pci/rme9652/hdsp.c
@@ -5636,22 +5636,11 @@ static void __devexit snd_hdsp_remove(struct pci_dev *pci)
 	pci_set_drvdata(pci, NULL);
 }
 
-static struct pci_driver driver = {
+static struct pci_driver hdsp_driver = {
 	.name =     KBUILD_MODNAME,
 	.id_table = snd_hdsp_ids,
 	.probe =    snd_hdsp_probe,
 	.remove = __devexit_p(snd_hdsp_remove),
 };
 
-static int __init alsa_card_hdsp_init(void)
-{
-	return pci_register_driver(&driver);
-}
-
-static void __exit alsa_card_hdsp_exit(void)
-{
-	pci_unregister_driver(&driver);
-}
-
-module_init(alsa_card_hdsp_init)
-module_exit(alsa_card_hdsp_exit)
+module_pci_driver(hdsp_driver);
diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c
index bc030a2088da..0a5027b94714 100644
--- a/sound/pci/rme9652/hdspm.c
+++ b/sound/pci/rme9652/hdspm.c
@@ -6918,23 +6918,11 @@ static void __devexit snd_hdspm_remove(struct pci_dev *pci)
 	pci_set_drvdata(pci, NULL);
 }
 
-static struct pci_driver driver = {
+static struct pci_driver hdspm_driver = {
 	.name = KBUILD_MODNAME,
 	.id_table = snd_hdspm_ids,
 	.probe = snd_hdspm_probe,
 	.remove = __devexit_p(snd_hdspm_remove),
 };
 
-
-static int __init alsa_card_hdspm_init(void)
-{
-	return pci_register_driver(&driver);
-}
-
-static void __exit alsa_card_hdspm_exit(void)
-{
-	pci_unregister_driver(&driver);
-}
-
-module_init(alsa_card_hdspm_init)
-module_exit(alsa_card_hdspm_exit)
+module_pci_driver(hdspm_driver);
diff --git a/sound/pci/rme9652/rme9652.c b/sound/pci/rme9652/rme9652.c
index b737d1619cc7..a15fc100ab0c 100644
--- a/sound/pci/rme9652/rme9652.c
+++ b/sound/pci/rme9652/rme9652.c
@@ -2631,22 +2631,11 @@ static void __devexit snd_rme9652_remove(struct pci_dev *pci)
 	pci_set_drvdata(pci, NULL);
 }
 
-static struct pci_driver driver = {
+static struct pci_driver rme9652_driver = {
 	.name	  = KBUILD_MODNAME,
 	.id_table = snd_rme9652_ids,
 	.probe	  = snd_rme9652_probe,
 	.remove	  = __devexit_p(snd_rme9652_remove),
 };
 
-static int __init alsa_card_hammerfall_init(void)
-{
-	return pci_register_driver(&driver);
-}
-
-static void __exit alsa_card_hammerfall_exit(void)
-{
-	pci_unregister_driver(&driver);
-}
-
-module_init(alsa_card_hammerfall_init)
-module_exit(alsa_card_hammerfall_exit)
+module_pci_driver(rme9652_driver);
diff --git a/sound/pci/sis7019.c b/sound/pci/sis7019.c
index ff500a87f769..1552642765d6 100644
--- a/sound/pci/sis7019.c
+++ b/sound/pci/sis7019.c
@@ -1488,15 +1488,4 @@ static struct pci_driver sis7019_driver = {
 #endif
 };
 
-static int __init sis7019_init(void)
-{
-	return pci_register_driver(&sis7019_driver);
-}
-
-static void __exit sis7019_exit(void)
-{
-	pci_unregister_driver(&sis7019_driver);
-}
-
-module_init(sis7019_init);
-module_exit(sis7019_exit);
+module_pci_driver(sis7019_driver);
diff --git a/sound/pci/sonicvibes.c b/sound/pci/sonicvibes.c
index 54cc802050f7..baa9946bedf0 100644
--- a/sound/pci/sonicvibes.c
+++ b/sound/pci/sonicvibes.c
@@ -1530,22 +1530,11 @@ static void __devexit snd_sonic_remove(struct pci_dev *pci)
 	pci_set_drvdata(pci, NULL);
 }
 
-static struct pci_driver driver = {
+static struct pci_driver sonicvibes_driver = {
 	.name = KBUILD_MODNAME,
 	.id_table = snd_sonic_ids,
 	.probe = snd_sonic_probe,
 	.remove = __devexit_p(snd_sonic_remove),
 };
 
-static int __init alsa_card_sonicvibes_init(void)
-{
-	return pci_register_driver(&driver);
-}
-
-static void __exit alsa_card_sonicvibes_exit(void)
-{
-	pci_unregister_driver(&driver);
-}
-
-module_init(alsa_card_sonicvibes_init)
-module_exit(alsa_card_sonicvibes_exit)
+module_pci_driver(sonicvibes_driver);
diff --git a/sound/pci/trident/trident.c b/sound/pci/trident/trident.c
index 5f1def7f45e5..611983ec7321 100644
--- a/sound/pci/trident/trident.c
+++ b/sound/pci/trident/trident.c
@@ -172,7 +172,7 @@ static void __devexit snd_trident_remove(struct pci_dev *pci)
 	pci_set_drvdata(pci, NULL);
 }
 
-static struct pci_driver driver = {
+static struct pci_driver trident_driver = {
 	.name = KBUILD_MODNAME,
 	.id_table = snd_trident_ids,
 	.probe = snd_trident_probe,
@@ -183,15 +183,4 @@ static struct pci_driver driver = {
 #endif
 };
 
-static int __init alsa_card_trident_init(void)
-{
-	return pci_register_driver(&driver);
-}
-
-static void __exit alsa_card_trident_exit(void)
-{
-	pci_unregister_driver(&driver);
-}
-
-module_init(alsa_card_trident_init)
-module_exit(alsa_card_trident_exit)
+module_pci_driver(trident_driver);
diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c
index 75630408c6db..b5afab48943e 100644
--- a/sound/pci/via82xx.c
+++ b/sound/pci/via82xx.c
@@ -2619,7 +2619,7 @@ static void __devexit snd_via82xx_remove(struct pci_dev *pci)
 	pci_set_drvdata(pci, NULL);
 }
 
-static struct pci_driver driver = {
+static struct pci_driver via82xx_driver = {
 	.name = KBUILD_MODNAME,
 	.id_table = snd_via82xx_ids,
 	.probe = snd_via82xx_probe,
@@ -2630,15 +2630,4 @@ static struct pci_driver driver = {
 #endif
 };
 
-static int __init alsa_card_via82xx_init(void)
-{
-	return pci_register_driver(&driver);
-}
-
-static void __exit alsa_card_via82xx_exit(void)
-{
-	pci_unregister_driver(&driver);
-}
-
-module_init(alsa_card_via82xx_init)
-module_exit(alsa_card_via82xx_exit)
+module_pci_driver(via82xx_driver);
diff --git a/sound/pci/via82xx_modem.c b/sound/pci/via82xx_modem.c
index 5efcbcac506a..59fd47ed0a31 100644
--- a/sound/pci/via82xx_modem.c
+++ b/sound/pci/via82xx_modem.c
@@ -1223,7 +1223,7 @@ static void __devexit snd_via82xx_remove(struct pci_dev *pci)
 	pci_set_drvdata(pci, NULL);
 }
 
-static struct pci_driver driver = {
+static struct pci_driver via82xx_modem_driver = {
 	.name = KBUILD_MODNAME,
 	.id_table = snd_via82xx_modem_ids,
 	.probe = snd_via82xx_probe,
@@ -1234,15 +1234,4 @@ static struct pci_driver driver = {
 #endif
 };
 
-static int __init alsa_card_via82xx_init(void)
-{
-	return pci_register_driver(&driver);
-}
-
-static void __exit alsa_card_via82xx_exit(void)
-{
-	pci_unregister_driver(&driver);
-}
-
-module_init(alsa_card_via82xx_init)
-module_exit(alsa_card_via82xx_exit)
+module_pci_driver(via82xx_modem_driver);
diff --git a/sound/pci/vx222/vx222.c b/sound/pci/vx222/vx222.c
index 6a534bfe1274..1ea1f656a5dc 100644
--- a/sound/pci/vx222/vx222.c
+++ b/sound/pci/vx222/vx222.c
@@ -289,7 +289,7 @@ static int snd_vx222_resume(struct pci_dev *pci)
 }
 #endif
 
-static struct pci_driver driver = {
+static struct pci_driver vx222_driver = {
 	.name = KBUILD_MODNAME,
 	.id_table = snd_vx222_ids,
 	.probe = snd_vx222_probe,
@@ -300,15 +300,4 @@ static struct pci_driver driver = {
 #endif
 };
 
-static int __init alsa_card_vx222_init(void)
-{
-	return pci_register_driver(&driver);
-}
-
-static void __exit alsa_card_vx222_exit(void)
-{
-	pci_unregister_driver(&driver);
-}
-
-module_init(alsa_card_vx222_init)
-module_exit(alsa_card_vx222_exit)
+module_pci_driver(vx222_driver);
diff --git a/sound/pci/ymfpci/ymfpci.c b/sound/pci/ymfpci/ymfpci.c
index 94ab728f5ca8..9a1d01d653a7 100644
--- a/sound/pci/ymfpci/ymfpci.c
+++ b/sound/pci/ymfpci/ymfpci.c
@@ -350,7 +350,7 @@ static void __devexit snd_card_ymfpci_remove(struct pci_dev *pci)
 	pci_set_drvdata(pci, NULL);
 }
 
-static struct pci_driver driver = {
+static struct pci_driver ymfpci_driver = {
 	.name = KBUILD_MODNAME,
 	.id_table = snd_ymfpci_ids,
 	.probe = snd_card_ymfpci_probe,
@@ -361,15 +361,4 @@ static struct pci_driver driver = {
 #endif
 };
 
-static int __init alsa_card_ymfpci_init(void)
-{
-	return pci_register_driver(&driver);
-}
-
-static void __exit alsa_card_ymfpci_exit(void)
-{
-	pci_unregister_driver(&driver);
-}
-
-module_init(alsa_card_ymfpci_init)
-module_exit(alsa_card_ymfpci_exit)
+module_pci_driver(ymfpci_driver);
diff --git a/sound/sh/sh_dac_audio.c b/sound/sh/sh_dac_audio.c
index b11f82b5718f..f8b01c77b298 100644
--- a/sound/sh/sh_dac_audio.c
+++ b/sound/sh/sh_dac_audio.c
@@ -433,7 +433,7 @@ probe_error:
 /*
  * "driver" definition
  */
-static struct platform_driver driver = {
+static struct platform_driver sh_dac_driver = {
 	.probe	= snd_sh_dac_probe,
 	.remove = snd_sh_dac_remove,
 	.driver = {
@@ -441,4 +441,4 @@ static struct platform_driver driver = {
 	},
 };
 
-module_platform_driver(driver);
+module_platform_driver(sh_dac_driver);
diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig
index 91c985599d32..40b2ad1bb1cd 100644
--- a/sound/soc/Kconfig
+++ b/sound/soc/Kconfig
@@ -35,7 +35,6 @@ source "sound/soc/blackfin/Kconfig"
 source "sound/soc/davinci/Kconfig"
 source "sound/soc/ep93xx/Kconfig"
 source "sound/soc/fsl/Kconfig"
-source "sound/soc/imx/Kconfig"
 source "sound/soc/jz4740/Kconfig"
 source "sound/soc/nuc900/Kconfig"
 source "sound/soc/omap/Kconfig"
@@ -48,9 +47,13 @@ source "sound/soc/s6000/Kconfig"
 source "sound/soc/sh/Kconfig"
 source "sound/soc/tegra/Kconfig"
 source "sound/soc/txx9/Kconfig"
+source "sound/soc/ux500/Kconfig"
 
 # Supported codecs
 source "sound/soc/codecs/Kconfig"
 
+# generic frame-work
+source "sound/soc/generic/Kconfig"
+
 endif	# SND_SOC
 
diff --git a/sound/soc/Makefile b/sound/soc/Makefile
index 2feaf376e94b..70990f4017f4 100644
--- a/sound/soc/Makefile
+++ b/sound/soc/Makefile
@@ -6,13 +6,13 @@ obj-$(CONFIG_SND_SOC_DMAENGINE_PCM) += snd-soc-dmaengine-pcm.o
 
 obj-$(CONFIG_SND_SOC)	+= snd-soc-core.o
 obj-$(CONFIG_SND_SOC)	+= codecs/
+obj-$(CONFIG_SND_SOC)	+= generic/
 obj-$(CONFIG_SND_SOC)	+= atmel/
 obj-$(CONFIG_SND_SOC)	+= au1x/
 obj-$(CONFIG_SND_SOC)	+= blackfin/
 obj-$(CONFIG_SND_SOC)	+= davinci/
 obj-$(CONFIG_SND_SOC)	+= ep93xx/
 obj-$(CONFIG_SND_SOC)	+= fsl/
-obj-$(CONFIG_SND_SOC)   += imx/
 obj-$(CONFIG_SND_SOC)	+= jz4740/
 obj-$(CONFIG_SND_SOC)	+= mid-x86/
 obj-$(CONFIG_SND_SOC)	+= mxs/
@@ -25,3 +25,4 @@ obj-$(CONFIG_SND_SOC)	+= s6000/
 obj-$(CONFIG_SND_SOC)	+= sh/
 obj-$(CONFIG_SND_SOC)	+= tegra/
 obj-$(CONFIG_SND_SOC)	+= txx9/
+obj-$(CONFIG_SND_SOC)	+= ux500/
diff --git a/sound/soc/blackfin/bf5xx-ssm2602.c b/sound/soc/blackfin/bf5xx-ssm2602.c
index b39ad356b92b..7dbeef1099b4 100644
--- a/sound/soc/blackfin/bf5xx-ssm2602.c
+++ b/sound/soc/blackfin/bf5xx-ssm2602.c
@@ -44,16 +44,8 @@
 
 static struct snd_soc_card bf5xx_ssm2602;
 
-static int bf5xx_ssm2602_hw_params(struct snd_pcm_substream *substream,
-	struct snd_pcm_hw_params *params)
+static int bf5xx_ssm2602_dai_init(struct snd_soc_pcm_runtime *rtd)
 {
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
-	unsigned int clk = 0;
-	int ret = 0;
-
-	pr_debug("%s rate %d format %x\n", __func__, params_rate(params),
-		params_format(params));
 	/*
 	 * If you are using a crystal source which frequency is not 12MHz
 	 * then modify the below case statement with frequency of the crystal.
@@ -61,31 +53,10 @@ static int bf5xx_ssm2602_hw_params(struct snd_pcm_substream *substream,
 	 * If you are using the SPORT to generate clocking then this is
 	 * where to do it.
 	 */
-
-	switch (params_rate(params)) {
-	case 8000:
-	case 16000:
-	case 48000:
-	case 96000:
-	case 11025:
-	case 22050:
-	case 44100:
-		clk = 12000000;
-		break;
-	}
-
-	ret = snd_soc_dai_set_sysclk(codec_dai, SSM2602_SYSCLK, clk,
+	return snd_soc_dai_set_sysclk(rtd->codec_dai, SSM2602_SYSCLK, 12000000,
 		SND_SOC_CLOCK_IN);
-	if (ret < 0)
-		return ret;
-
-	return 0;
 }
 
-static struct snd_soc_ops bf5xx_ssm2602_ops = {
-	.hw_params = bf5xx_ssm2602_hw_params,
-};
-
 /* CODEC is master for BCLK and LRC in this configuration. */
 #define BF5XX_SSM2602_DAIFMT (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | \
 				SND_SOC_DAIFMT_CBM_CFM)
@@ -98,7 +69,7 @@ static struct snd_soc_dai_link bf5xx_ssm2602_dai[] = {
 		.codec_dai_name = "ssm2602-hifi",
 		.platform_name = "bfin-i2s-pcm-audio",
 		.codec_name = "ssm2602.0-001b",
-		.ops = &bf5xx_ssm2602_ops,
+		.init = bf5xx_ssm2602_dai_init,
 		.dai_fmt = BF5XX_SSM2602_DAIFMT,
 	},
 	{
@@ -108,7 +79,7 @@ static struct snd_soc_dai_link bf5xx_ssm2602_dai[] = {
 		.codec_dai_name = "ssm2602-hifi",
 		.platform_name = "bfin-i2s-pcm-audio",
 		.codec_name = "ssm2602.0-001b",
-		.ops = &bf5xx_ssm2602_ops,
+		.init = bf5xx_ssm2602_dai_init,
 		.dai_fmt = BF5XX_SSM2602_DAIFMT,
 	},
 };
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index 59d8efaa17e9..1e1613a438dd 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_CS42L52 if I2C
 	select SND_SOC_CS42L73 if I2C
 	select SND_SOC_CS4270 if I2C
 	select SND_SOC_CS4271 if SND_SOC_I2C_AND_SPI
@@ -37,11 +38,15 @@ config SND_SOC_ALL_CODECS
 	select SND_SOC_DFBMCS320
 	select SND_SOC_JZ4740_CODEC
 	select SND_SOC_LM4857 if I2C
+	select SND_SOC_LM49453 if I2C
 	select SND_SOC_MAX98088 if I2C
 	select SND_SOC_MAX98095 if I2C
 	select SND_SOC_MAX9850 if I2C
 	select SND_SOC_MAX9768 if I2C
 	select SND_SOC_MAX9877 if I2C
+	select SND_SOC_MC13783 if MFD_MC13XXX
+	select SND_SOC_ML26124 if I2C
+	select SND_SOC_OMAP_HDMI_CODEC if OMAP4_DSS_HDMI
 	select SND_SOC_PCM3008
 	select SND_SOC_RT5631 if I2C
 	select SND_SOC_SGTL5000 if I2C
@@ -181,6 +186,9 @@ config SND_SOC_CQ0093VC
 config SND_SOC_CS42L51
 	tristate
 
+config SND_SOC_CS42L52
+	tristate
+
 config SND_SOC_CS42L73
 	tristate
 
@@ -217,6 +225,9 @@ config SND_SOC_DFBMCS320
 config SND_SOC_DMIC
 	tristate
 
+config SND_SOC_LM49453
+	tristate
+
 config SND_SOC_MAX98088
        tristate
 
@@ -226,6 +237,9 @@ config SND_SOC_MAX98095
 config SND_SOC_MAX9850
 	tristate
 
+config SND_SOC_OMAP_HDMI_CODEC
+       tristate
+
 config SND_SOC_PCM3008
        tristate
 
@@ -435,5 +449,11 @@ config SND_SOC_MAX9768
 config SND_SOC_MAX9877
 	tristate
 
+config SND_SOC_MC13783
+	tristate
+
+config SND_SOC_ML26124
+	tristate
+
 config SND_SOC_TPA6130A2
 	tristate
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index 6662eb0cdcc0..fc27fec39487 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-cs42l52-objs := cs42l52.o
 snd-soc-cs42l73-objs := cs42l73.o
 snd-soc-cs4270-objs := cs4270.o
 snd-soc-cs4271-objs := cs4271.o
@@ -25,10 +26,14 @@ snd-soc-dmic-objs := dmic.o
 snd-soc-jz4740-codec-objs := jz4740.o
 snd-soc-l3-objs := l3.o
 snd-soc-lm4857-objs := lm4857.o
+snd-soc-lm49453-objs := lm49453.o
 snd-soc-max9768-objs := max9768.o
 snd-soc-max98088-objs := max98088.o
 snd-soc-max98095-objs := max98095.o
 snd-soc-max9850-objs := max9850.o
+snd-soc-mc13783-objs := mc13783.o
+snd-soc-ml26124-objs := ml26124.o
+snd-soc-omap-hdmi-codec-objs := omap-hdmi.o
 snd-soc-pcm3008-objs := pcm3008.o
 snd-soc-rt5631-objs := rt5631.o
 snd-soc-sgtl5000-objs := sgtl5000.o
@@ -121,6 +126,7 @@ 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-$(CONFIG_SND_SOC_CS42L52)	+= snd-soc-cs42l52.o
 obj-$(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
@@ -128,13 +134,17 @@ obj-$(CONFIG_SND_SOC_CX20442)	+= snd-soc-cx20442.o
 obj-$(CONFIG_SND_SOC_DA7210)	+= snd-soc-da7210.o
 obj-$(CONFIG_SND_SOC_DFBMCS320)	+= snd-soc-dfbmcs320.o
 obj-$(CONFIG_SND_SOC_DMIC)	+= snd-soc-dmic.o
+obj-$(CONFIG_SND_SOC_JZ4740_CODEC)	+= snd-soc-jz4740-codec.o
 obj-$(CONFIG_SND_SOC_L3)	+= snd-soc-l3.o
 obj-$(CONFIG_SND_SOC_LM4857)	+= snd-soc-lm4857.o
-obj-$(CONFIG_SND_SOC_JZ4740_CODEC)	+= snd-soc-jz4740-codec.o
+obj-$(CONFIG_SND_SOC_LM49453)   += snd-soc-lm49453.o
 obj-$(CONFIG_SND_SOC_MAX9768)	+= snd-soc-max9768.o
 obj-$(CONFIG_SND_SOC_MAX98088)	+= snd-soc-max98088.o
 obj-$(CONFIG_SND_SOC_MAX98095)	+= snd-soc-max98095.o
 obj-$(CONFIG_SND_SOC_MAX9850)	+= snd-soc-max9850.o
+obj-$(CONFIG_SND_SOC_MC13783)	+= snd-soc-mc13783.o
+obj-$(CONFIG_SND_SOC_ML26124)	+= snd-soc-ml26124.o
+obj-$(CONFIG_SND_SOC_OMAP_HDMI_CODEC) += snd-soc-omap-hdmi-codec.o
 obj-$(CONFIG_SND_SOC_PCM3008)	+= snd-soc-pcm3008.o
 obj-$(CONFIG_SND_SOC_RT5631)	+= snd-soc-rt5631.o
 obj-$(CONFIG_SND_SOC_SGTL5000)  += snd-soc-sgtl5000.o
diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c
index 1bbad4c16d28..2023c749f232 100644
--- a/sound/soc/codecs/ac97.c
+++ b/sound/soc/codecs/ac97.c
@@ -26,13 +26,11 @@
 static int ac97_prepare(struct snd_pcm_substream *substream,
 			struct snd_soc_dai *dai)
 {
-	struct snd_pcm_runtime *runtime = substream->runtime;
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_codec *codec = rtd->codec;
+	struct snd_soc_codec *codec = dai->codec;
 
 	int reg = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ?
 		  AC97_PCM_FRONT_DAC_RATE : AC97_PCM_LR_ADC_RATE;
-	return snd_ac97_set_rate(codec->ac97, reg, runtime->rate);
+	return snd_ac97_set_rate(codec->ac97, reg, substream->runtime->rate);
 }
 
 #define STD_AC97_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\
diff --git a/sound/soc/codecs/ad1836.c b/sound/soc/codecs/ad1836.c
index 12e3b4118557..c67b50d8b317 100644
--- a/sound/soc/codecs/ad1836.c
+++ b/sound/soc/codecs/ad1836.c
@@ -162,9 +162,7 @@ static int ad1836_hw_params(struct snd_pcm_substream *substream,
 		struct snd_soc_dai *dai)
 {
 	int word_len = 0;
-
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_codec *codec = rtd->codec;
+	struct snd_soc_codec *codec = dai->codec;
 
 	/* bit size */
 	switch (params_format(params)) {
diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c
index a4a6bef2c0bb..13e62be4f990 100644
--- a/sound/soc/codecs/ad193x.c
+++ b/sound/soc/codecs/ad193x.c
@@ -245,9 +245,7 @@ static int ad193x_hw_params(struct snd_pcm_substream *substream,
 		struct snd_soc_dai *dai)
 {
 	int word_len = 0, master_rate = 0;
-
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_codec *codec = rtd->codec;
+	struct snd_soc_codec *codec = dai->codec;
 	struct ad193x_priv *ad193x = snd_soc_codec_get_drvdata(codec);
 
 	/* bit size */
diff --git a/sound/soc/codecs/adau1701.c b/sound/soc/codecs/adau1701.c
index 78e9ce48bb99..3d50fc8646b6 100644
--- a/sound/soc/codecs/adau1701.c
+++ b/sound/soc/codecs/adau1701.c
@@ -258,8 +258,7 @@ static int adau1701_set_playback_pcm_format(struct snd_soc_codec *codec,
 static int adau1701_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 snd_soc_codec *codec = dai->codec;
 	snd_pcm_format_t format;
 	unsigned int val;
 
diff --git a/sound/soc/codecs/ak4104.c b/sound/soc/codecs/ak4104.c
index ceb96ecf5588..31d4483245d0 100644
--- a/sound/soc/codecs/ak4104.c
+++ b/sound/soc/codecs/ak4104.c
@@ -88,8 +88,7 @@ static int ak4104_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 snd_soc_codec *codec = dai->codec;
 	int val = 0;
 
 	/* set the IEC958 bits: consumer mode, no copyright bit */
diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c
index 838ae8b22b50..618fdc30f73e 100644
--- a/sound/soc/codecs/ak4535.c
+++ b/sound/soc/codecs/ak4535.c
@@ -262,8 +262,7 @@ static int ak4535_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 snd_soc_codec *codec = dai->codec;
 	struct ak4535_priv *ak4535 = snd_soc_codec_get_drvdata(codec);
 	u8 mode2 = snd_soc_read(codec, AK4535_MODE2) & ~(0x3 << 5);
 	int rate = params_rate(params), fs = 256;
diff --git a/sound/soc/codecs/ak4641.c b/sound/soc/codecs/ak4641.c
index c4d165a4bddf..543a12f471be 100644
--- a/sound/soc/codecs/ak4641.c
+++ b/sound/soc/codecs/ak4641.c
@@ -296,8 +296,7 @@ static int ak4641_i2s_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 snd_soc_codec *codec = dai->codec;
 	struct ak4641_priv *ak4641 = snd_soc_codec_get_drvdata(codec);
 	int rate = params_rate(params), fs = 256;
 	u8 mode2;
@@ -517,67 +516,24 @@ static int ak4641_resume(struct snd_soc_codec *codec)
 
 static int ak4641_probe(struct snd_soc_codec *codec)
 {
-	struct ak4641_platform_data *pdata = codec->dev->platform_data;
 	int ret;
 
-
-	if (pdata) {
-		if (gpio_is_valid(pdata->gpio_power)) {
-			ret = gpio_request_one(pdata->gpio_power,
-					GPIOF_OUT_INIT_LOW, "ak4641 power");
-			if (ret)
-				goto err_out;
-		}
-		if (gpio_is_valid(pdata->gpio_npdn)) {
-			ret = gpio_request_one(pdata->gpio_npdn,
-					GPIOF_OUT_INIT_LOW, "ak4641 npdn");
-			if (ret)
-				goto err_gpio;
-
-			udelay(1); /* > 150 ns */
-			gpio_set_value(pdata->gpio_npdn, 1);
-		}
-	}
-
 	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);
-		goto err_register;
+		return ret;
 	}
 
 	/* power on device */
 	ak4641_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 
 	return 0;
-
-err_register:
-	if (pdata) {
-		if (gpio_is_valid(pdata->gpio_power))
-			gpio_set_value(pdata->gpio_power, 0);
-		if (gpio_is_valid(pdata->gpio_npdn))
-			gpio_free(pdata->gpio_npdn);
-	}
-err_gpio:
-	if (pdata && gpio_is_valid(pdata->gpio_power))
-		gpio_free(pdata->gpio_power);
-err_out:
-	return ret;
 }
 
 static int ak4641_remove(struct snd_soc_codec *codec)
 {
-	struct ak4641_platform_data *pdata = codec->dev->platform_data;
-
 	ak4641_set_bias_level(codec, SND_SOC_BIAS_OFF);
 
-	if (pdata) {
-		if (gpio_is_valid(pdata->gpio_power)) {
-			gpio_set_value(pdata->gpio_power, 0);
-			gpio_free(pdata->gpio_power);
-		}
-		if (gpio_is_valid(pdata->gpio_npdn))
-			gpio_free(pdata->gpio_npdn);
-	}
 	return 0;
 }
 
@@ -604,6 +560,7 @@ static struct snd_soc_codec_driver soc_codec_dev_ak4641 = {
 static int __devinit ak4641_i2c_probe(struct i2c_client *i2c,
 				      const struct i2c_device_id *id)
 {
+	struct ak4641_platform_data *pdata = i2c->dev.platform_data;
 	struct ak4641_priv *ak4641;
 	int ret;
 
@@ -612,16 +569,62 @@ static int __devinit ak4641_i2c_probe(struct i2c_client *i2c,
 	if (!ak4641)
 		return -ENOMEM;
 
+	if (pdata) {
+		if (gpio_is_valid(pdata->gpio_power)) {
+			ret = gpio_request_one(pdata->gpio_power,
+					GPIOF_OUT_INIT_LOW, "ak4641 power");
+			if (ret)
+				goto err_out;
+		}
+		if (gpio_is_valid(pdata->gpio_npdn)) {
+			ret = gpio_request_one(pdata->gpio_npdn,
+					GPIOF_OUT_INIT_LOW, "ak4641 npdn");
+			if (ret)
+				goto err_gpio;
+
+			udelay(1); /* > 150 ns */
+			gpio_set_value(pdata->gpio_npdn, 1);
+		}
+	}
+
 	i2c_set_clientdata(i2c, ak4641);
 
 	ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_ak4641,
 				ak4641_dai, ARRAY_SIZE(ak4641_dai));
+	if (ret != 0)
+		goto err_gpio2;
+
+	return 0;
+
+err_gpio2:
+	if (pdata) {
+		if (gpio_is_valid(pdata->gpio_power))
+			gpio_set_value(pdata->gpio_power, 0);
+		if (gpio_is_valid(pdata->gpio_npdn))
+			gpio_free(pdata->gpio_npdn);
+	}
+err_gpio:
+	if (pdata && gpio_is_valid(pdata->gpio_power))
+		gpio_free(pdata->gpio_power);
+err_out:
 	return ret;
 }
 
 static int __devexit ak4641_i2c_remove(struct i2c_client *i2c)
 {
+	struct ak4641_platform_data *pdata = i2c->dev.platform_data;
+
 	snd_soc_unregister_codec(&i2c->dev);
+
+	if (pdata) {
+		if (gpio_is_valid(pdata->gpio_power)) {
+			gpio_set_value(pdata->gpio_power, 0);
+			gpio_free(pdata->gpio_power);
+		}
+		if (gpio_is_valid(pdata->gpio_npdn))
+			gpio_free(pdata->gpio_npdn);
+	}
+
 	return 0;
 }
 
@@ -641,23 +644,7 @@ static struct i2c_driver ak4641_i2c_driver = {
 	.id_table = ak4641_i2c_id,
 };
 
-static int __init ak4641_modinit(void)
-{
-	int ret;
-
-	ret = i2c_add_driver(&ak4641_i2c_driver);
-	if (ret != 0)
-		pr_err("Failed to register AK4641 I2C driver: %d\n", ret);
-
-	return ret;
-}
-module_init(ak4641_modinit);
-
-static void __exit ak4641_exit(void)
-{
-	i2c_del_driver(&ak4641_i2c_driver);
-}
-module_exit(ak4641_exit);
+module_i2c_driver(ak4641_i2c_driver);
 
 MODULE_DESCRIPTION("SoC AK4641 driver");
 MODULE_AUTHOR("Harald Welte <laforge@gnufiish.org>");
diff --git a/sound/soc/codecs/alc5623.c b/sound/soc/codecs/alc5623.c
index d47b62ddb210..1960478ce6bb 100644
--- a/sound/soc/codecs/alc5623.c
+++ b/sound/soc/codecs/alc5623.c
@@ -705,8 +705,7 @@ static int alc5623_set_dai_fmt(struct snd_soc_dai *codec_dai,
 static int alc5623_pcm_hw_params(struct snd_pcm_substream *substream,
 		struct snd_pcm_hw_params *params, struct snd_soc_dai *dai)
 {
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_codec *codec = rtd->codec;
+	struct snd_soc_codec *codec = dai->codec;
 	struct alc5623_priv *alc5623 = snd_soc_codec_get_drvdata(codec);
 	int coeff, rate;
 	u16 iface;
@@ -1084,25 +1083,7 @@ static struct i2c_driver alc5623_i2c_driver = {
 	.id_table = alc5623_i2c_table,
 };
 
-static int __init alc5623_modinit(void)
-{
-	int ret;
-
-	ret = i2c_add_driver(&alc5623_i2c_driver);
-	if (ret != 0) {
-		printk(KERN_ERR "%s: can't add i2c driver", __func__);
-		return ret;
-	}
-
-	return ret;
-}
-module_init(alc5623_modinit);
-
-static void __exit alc5623_modexit(void)
-{
-	i2c_del_driver(&alc5623_i2c_driver);
-}
-module_exit(alc5623_modexit);
+module_i2c_driver(alc5623_i2c_driver);
 
 MODULE_DESCRIPTION("ASoC alc5621/2/3 driver");
 MODULE_AUTHOR("Arnaud Patard <arnaud.patard@rtp-net.org>");
diff --git a/sound/soc/codecs/alc5632.c b/sound/soc/codecs/alc5632.c
index e2111e0ccad7..7dd02420b36d 100644
--- a/sound/soc/codecs/alc5632.c
+++ b/sound/soc/codecs/alc5632.c
@@ -861,8 +861,7 @@ static int alc5632_set_dai_fmt(struct snd_soc_dai *codec_dai,
 static int alc5632_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 snd_soc_codec *codec = dai->codec;
 	int coeff, rate;
 	u16 iface;
 
@@ -1131,7 +1130,7 @@ static __devinit int alc5632_i2c_probe(struct i2c_client *client,
 
 	i2c_set_clientdata(client, alc5632);
 
-	alc5632->regmap = regmap_init_i2c(client, &alc5632_regmap);
+	alc5632->regmap = devm_regmap_init_i2c(client, &alc5632_regmap);
 	if (IS_ERR(alc5632->regmap)) {
 		ret = PTR_ERR(alc5632->regmap);
 		dev_err(&client->dev, "regmap_init() failed: %d\n", ret);
@@ -1143,7 +1142,6 @@ static __devinit int alc5632_i2c_probe(struct i2c_client *client,
 	if (ret1 != 0 || ret2 != 0) {
 		dev_err(&client->dev,
 		"Failed to read chip ID: ret1=%d, ret2=%d\n", ret1, ret2);
-		regmap_exit(alc5632->regmap);
 		return -EIO;
 	}
 
@@ -1152,14 +1150,12 @@ static __devinit int alc5632_i2c_probe(struct i2c_client *client,
 	if ((vid1 != 0x10EC) || (vid2 != id->driver_data)) {
 		dev_err(&client->dev,
 		"Device is not a ALC5632: VID1=0x%x, VID2=0x%x\n", vid1, vid2);
-		regmap_exit(alc5632->regmap);
 		return -EINVAL;
 	}
 
 	ret = alc5632_reset(alc5632->regmap);
 	if (ret < 0) {
 		dev_err(&client->dev, "Failed to issue reset\n");
-		regmap_exit(alc5632->regmap);
 		return ret;
 	}
 
@@ -1177,7 +1173,6 @@ static __devinit int alc5632_i2c_probe(struct i2c_client *client,
 
 	if (ret < 0) {
 		dev_err(&client->dev, "Failed to register codec: %d\n", ret);
-		regmap_exit(alc5632->regmap);
 		return ret;
 	}
 
@@ -1186,9 +1181,7 @@ static __devinit int alc5632_i2c_probe(struct i2c_client *client,
 
 static __devexit int alc5632_i2c_remove(struct i2c_client *client)
 {
-	struct alc5632_priv *alc5632 = i2c_get_clientdata(client);
 	snd_soc_unregister_codec(&client->dev);
-	regmap_exit(alc5632->regmap);
 	return 0;
 }
 
@@ -1209,25 +1202,7 @@ static struct i2c_driver alc5632_i2c_driver = {
 	.id_table = alc5632_i2c_table,
 };
 
-static int __init alc5632_modinit(void)
-{
-	int ret;
-
-	ret = i2c_add_driver(&alc5632_i2c_driver);
-	if (ret != 0) {
-		printk(KERN_ERR "%s: can't add i2c driver", __func__);
-		return ret;
-	}
-
-	return ret;
-}
-module_init(alc5632_modinit);
-
-static void __exit alc5632_modexit(void)
-{
-	i2c_del_driver(&alc5632_i2c_driver);
-}
-module_exit(alc5632_modexit);
+module_i2c_driver(alc5632_i2c_driver);
 
 MODULE_DESCRIPTION("ASoC ALC5632 driver");
 MODULE_AUTHOR("Leon Romanovsky <leon@leon.nu>");
diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c
index 1d672f528662..047917f0b8ae 100644
--- a/sound/soc/codecs/cs4270.c
+++ b/sound/soc/codecs/cs4270.c
@@ -307,8 +307,7 @@ static int cs4270_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 snd_soc_codec *codec = dai->codec;
 	struct cs4270_private *cs4270 = snd_soc_codec_get_drvdata(codec);
 	int ret;
 	unsigned int i;
@@ -600,10 +599,12 @@ static int cs4270_soc_suspend(struct snd_soc_codec *codec)
 static int cs4270_soc_resume(struct snd_soc_codec *codec)
 {
 	struct cs4270_private *cs4270 = snd_soc_codec_get_drvdata(codec);
-	int reg;
+	int reg, ret;
 
-	regulator_bulk_enable(ARRAY_SIZE(cs4270->supplies),
-			      cs4270->supplies);
+	ret = regulator_bulk_enable(ARRAY_SIZE(cs4270->supplies),
+				    cs4270->supplies);
+	if (ret != 0)
+		return ret;
 
 	/* In case the device was put to hard reset during sleep, we need to
 	 * wait 500ns here before any I2C communication. */
diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c
index bf7141280a74..9eb01d7d58a3 100644
--- a/sound/soc/codecs/cs4271.c
+++ b/sound/soc/codecs/cs4271.c
@@ -318,8 +318,7 @@ static int cs4271_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 snd_soc_codec *codec = dai->codec;
 	struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec);
 	int i, ret;
 	unsigned int ratio, val;
diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c
index a8bf588e8740..091d0193f507 100644
--- a/sound/soc/codecs/cs42l51.c
+++ b/sound/soc/codecs/cs42l51.c
@@ -141,15 +141,15 @@ static const struct soc_enum cs42l51_chan_mix =
 static const struct snd_kcontrol_new cs42l51_snd_controls[] = {
 	SOC_DOUBLE_R_SX_TLV("PCM Playback Volume",
 			CS42L51_PCMA_VOL, CS42L51_PCMB_VOL,
-			7, 0xffffff99, 0x18, adc_pcm_tlv),
+			6, 0x19, 0x7F, adc_pcm_tlv),
 	SOC_DOUBLE_R("PCM Playback Switch",
 			CS42L51_PCMA_VOL, CS42L51_PCMB_VOL, 7, 1, 1),
 	SOC_DOUBLE_R_SX_TLV("Analog Playback Volume",
 			CS42L51_AOUTA_VOL, CS42L51_AOUTB_VOL,
-			8, 0xffffff19, 0x18, aout_tlv),
+			0, 0x34, 0xE4, aout_tlv),
 	SOC_DOUBLE_R_SX_TLV("ADC Mixer Volume",
 			CS42L51_ADCA_VOL, CS42L51_ADCB_VOL,
-			7, 0xffffff99, 0x18, adc_pcm_tlv),
+			6, 0x19, 0x7F, adc_pcm_tlv),
 	SOC_DOUBLE_R("ADC Mixer Switch",
 			CS42L51_ADCA_VOL, CS42L51_ADCB_VOL, 7, 1, 1),
 	SOC_SINGLE("Playback Deemphasis Switch", CS42L51_DAC_CTL, 3, 1, 0),
@@ -356,8 +356,7 @@ static int cs42l51_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 snd_soc_codec *codec = dai->codec;
 	struct cs42l51_private *cs42l51 = snd_soc_codec_get_drvdata(codec);
 	int ret;
 	unsigned int i;
diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c
new file mode 100644
index 000000000000..a7109413aef1
--- /dev/null
+++ b/sound/soc/codecs/cs42l52.c
@@ -0,0 +1,1295 @@
+/*
+ * cs42l52.c -- CS42L52 ALSA SoC audio driver
+ *
+ * Copyright 2012 CirrusLogic, Inc.
+ *
+ * Author: Georgi Vlaev <joe@nucleusys.com>
+ * Author: 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.
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/version.h>
+#include <linux/kernel.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/i2c.h>
+#include <linux/input.h>
+#include <linux/regmap.h>
+#include <linux/slab.h>
+#include <linux/workqueue.h>
+#include <linux/platform_device.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 <sound/cs42l52.h>
+#include "cs42l52.h"
+
+struct sp_config {
+	u8 spc, format, spfs;
+	u32 srate;
+};
+
+struct  cs42l52_private {
+	struct regmap *regmap;
+	struct snd_soc_codec *codec;
+	struct device *dev;
+	struct sp_config config;
+	struct cs42l52_platform_data pdata;
+	u32 sysclk;
+	u8 mclksel;
+	u32 mclk;
+	u8 flags;
+#if defined(CONFIG_INPUT) || defined(CONFIG_INPUT_MODULE)
+	struct input_dev *beep;
+	struct work_struct beep_work;
+	int beep_rate;
+#endif
+};
+
+static const struct reg_default cs42l52_reg_defaults[] = {
+	{ CS42L52_PWRCTL1, 0x9F },	/* r02 PWRCTL 1 */
+	{ CS42L52_PWRCTL2, 0x07 },	/* r03 PWRCTL 2 */
+	{ CS42L52_PWRCTL3, 0xFF },	/* r04 PWRCTL 3 */
+	{ CS42L52_CLK_CTL, 0xA0 },	/* r05 Clocking Ctl */
+	{ CS42L52_IFACE_CTL1, 0x00 },	/* r06 Interface Ctl 1 */
+	{ CS42L52_ADC_PGA_A, 0x80 },	/* r08 Input A Select */
+	{ CS42L52_ADC_PGA_B, 0x80 },	/* r09 Input B Select */
+	{ CS42L52_ANALOG_HPF_CTL, 0xA5 },	/* r0A Analog HPF Ctl */
+	{ CS42L52_ADC_HPF_FREQ, 0x00 },	/* r0B ADC HPF Corner Freq */
+	{ CS42L52_ADC_MISC_CTL, 0x00 },	/* r0C Misc. ADC Ctl */
+	{ CS42L52_PB_CTL1, 0x60 },	/* r0D Playback Ctl 1 */
+	{ CS42L52_MISC_CTL, 0x02 },	/* r0E Misc. Ctl */
+	{ CS42L52_PB_CTL2, 0x00 },	/* r0F Playback Ctl 2 */
+	{ CS42L52_MICA_CTL, 0x00 },	/* r10 MICA Amp Ctl */
+	{ CS42L52_MICB_CTL, 0x00 },	/* r11 MICB Amp Ctl */
+	{ CS42L52_PGAA_CTL, 0x00 },	/* r12 PGAA Vol, Misc. */
+	{ CS42L52_PGAB_CTL, 0x00 },	/* r13 PGAB Vol, Misc. */
+	{ CS42L52_PASSTHRUA_VOL, 0x00 },	/* r14 Bypass A Vol */
+	{ CS42L52_PASSTHRUB_VOL, 0x00 },	/* r15 Bypass B Vol */
+	{ CS42L52_ADCA_VOL, 0x00 },	/* r16 ADCA Volume */
+	{ CS42L52_ADCB_VOL, 0x00 },	/* r17 ADCB Volume */
+	{ CS42L52_ADCA_MIXER_VOL, 0x80 },	/* r18 ADCA Mixer Volume */
+	{ CS42L52_ADCB_MIXER_VOL, 0x80 },	/* r19 ADCB Mixer Volume */
+	{ CS42L52_PCMA_MIXER_VOL, 0x00 },	/* r1A PCMA Mixer Volume */
+	{ CS42L52_PCMB_MIXER_VOL, 0x00 },	/* r1B PCMB Mixer Volume */
+	{ CS42L52_BEEP_FREQ, 0x00 },	/* r1C Beep Freq on Time */
+	{ CS42L52_BEEP_VOL, 0x00 },	/* r1D Beep Volume off Time */
+	{ CS42L52_BEEP_TONE_CTL, 0x00 },	/* r1E Beep Tone Cfg. */
+	{ CS42L52_TONE_CTL, 0x00 },	/* r1F Tone Ctl */
+	{ CS42L52_MASTERA_VOL, 0x88 },	/* r20 Master A Volume */
+	{ CS42L52_MASTERB_VOL, 0x00 },	/* r21 Master B Volume */
+	{ CS42L52_HPA_VOL, 0x00 },	/* r22 Headphone A Volume */
+	{ CS42L52_HPB_VOL, 0x00 },	/* r23 Headphone B Volume */
+	{ CS42L52_SPKA_VOL, 0x00 },	/* r24 Speaker A Volume */
+	{ CS42L52_SPKB_VOL, 0x00 },	/* r25 Speaker B Volume */
+	{ CS42L52_ADC_PCM_MIXER, 0x00 },	/* r26 Channel Mixer and Swap */
+	{ CS42L52_LIMITER_CTL1, 0x00 },	/* r27 Limit Ctl 1 Thresholds */
+	{ CS42L52_LIMITER_CTL2, 0x7F },	/* r28 Limit Ctl 2 Release Rate */
+	{ CS42L52_LIMITER_AT_RATE, 0xC0 },	/* r29 Limiter Attack Rate */
+	{ CS42L52_ALC_CTL, 0x00 },	/* r2A ALC Ctl 1 Attack Rate */
+	{ CS42L52_ALC_RATE, 0x3F },	/* r2B ALC Release Rate */
+	{ CS42L52_ALC_THRESHOLD, 0x3f },	/* r2C ALC Thresholds */
+	{ CS42L52_NOISE_GATE_CTL, 0x00 },	/* r2D Noise Gate Ctl */
+	{ CS42L52_CLK_STATUS, 0x00 },	/* r2E Overflow and Clock Status */
+	{ CS42L52_BATT_COMPEN, 0x00 },	/* r2F battery Compensation */
+	{ CS42L52_BATT_LEVEL, 0x00 },	/* r30 VP Battery Level */
+	{ CS42L52_SPK_STATUS, 0x00 },	/* r31 Speaker Status */
+	{ CS42L52_TEM_CTL, 0x3B },	/* r32 Temp Ctl */
+	{ CS42L52_THE_FOLDBACK, 0x00 },	/* r33 Foldback */
+};
+
+static bool cs42l52_readable_register(struct device *dev, unsigned int reg)
+{
+	switch (reg) {
+	case CS42L52_CHIP:
+	case CS42L52_PWRCTL1:
+	case CS42L52_PWRCTL2:
+	case CS42L52_PWRCTL3:
+	case CS42L52_CLK_CTL:
+	case CS42L52_IFACE_CTL1:
+	case CS42L52_IFACE_CTL2:
+	case CS42L52_ADC_PGA_A:
+	case CS42L52_ADC_PGA_B:
+	case CS42L52_ANALOG_HPF_CTL:
+	case CS42L52_ADC_HPF_FREQ:
+	case CS42L52_ADC_MISC_CTL:
+	case CS42L52_PB_CTL1:
+	case CS42L52_MISC_CTL:
+	case CS42L52_PB_CTL2:
+	case CS42L52_MICA_CTL:
+	case CS42L52_MICB_CTL:
+	case CS42L52_PGAA_CTL:
+	case CS42L52_PGAB_CTL:
+	case CS42L52_PASSTHRUA_VOL:
+	case CS42L52_PASSTHRUB_VOL:
+	case CS42L52_ADCA_VOL:
+	case CS42L52_ADCB_VOL:
+	case CS42L52_ADCA_MIXER_VOL:
+	case CS42L52_ADCB_MIXER_VOL:
+	case CS42L52_PCMA_MIXER_VOL:
+	case CS42L52_PCMB_MIXER_VOL:
+	case CS42L52_BEEP_FREQ:
+	case CS42L52_BEEP_VOL:
+	case CS42L52_BEEP_TONE_CTL:
+	case CS42L52_TONE_CTL:
+	case CS42L52_MASTERA_VOL:
+	case CS42L52_MASTERB_VOL:
+	case CS42L52_HPA_VOL:
+	case CS42L52_HPB_VOL:
+	case CS42L52_SPKA_VOL:
+	case CS42L52_SPKB_VOL:
+	case CS42L52_ADC_PCM_MIXER:
+	case CS42L52_LIMITER_CTL1:
+	case CS42L52_LIMITER_CTL2:
+	case CS42L52_LIMITER_AT_RATE:
+	case CS42L52_ALC_CTL:
+	case CS42L52_ALC_RATE:
+	case CS42L52_ALC_THRESHOLD:
+	case CS42L52_NOISE_GATE_CTL:
+	case CS42L52_CLK_STATUS:
+	case CS42L52_BATT_COMPEN:
+	case CS42L52_BATT_LEVEL:
+	case CS42L52_SPK_STATUS:
+	case CS42L52_TEM_CTL:
+	case CS42L52_THE_FOLDBACK:
+	case CS42L52_CHARGE_PUMP:
+		return true;
+	default:
+		return false;
+	}
+}
+
+static bool cs42l52_volatile_register(struct device *dev, unsigned int reg)
+{
+	switch (reg) {
+	case CS42L52_IFACE_CTL2:
+	case CS42L52_CLK_STATUS:
+	case CS42L52_BATT_LEVEL:
+	case CS42L52_SPK_STATUS:
+	case CS42L52_CHARGE_PUMP:
+		return 1;
+	default:
+		return 0;
+	}
+}
+
+static DECLARE_TLV_DB_SCALE(hl_tlv, -10200, 50, 0);
+
+static DECLARE_TLV_DB_SCALE(hpd_tlv, -9600, 50, 1);
+
+static DECLARE_TLV_DB_SCALE(ipd_tlv, -9600, 100, 0);
+
+static DECLARE_TLV_DB_SCALE(mic_tlv, 1600, 100, 0);
+
+static DECLARE_TLV_DB_SCALE(pga_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 char * const cs42l52_adca_text[] = {
+	"Input1A", "Input2A", "Input3A", "Input4A", "PGA Input Left"};
+
+static const char * const cs42l52_adcb_text[] = {
+	"Input1B", "Input2B", "Input3B", "Input4B", "PGA Input Right"};
+
+static const struct soc_enum adca_enum =
+	SOC_ENUM_SINGLE(CS42L52_ADC_PGA_A, 5,
+		ARRAY_SIZE(cs42l52_adca_text), cs42l52_adca_text);
+
+static const struct soc_enum adcb_enum =
+	SOC_ENUM_SINGLE(CS42L52_ADC_PGA_B, 5,
+		ARRAY_SIZE(cs42l52_adcb_text), cs42l52_adcb_text);
+
+static const struct snd_kcontrol_new adca_mux =
+	SOC_DAPM_ENUM("Left ADC Input Capture Mux", adca_enum);
+
+static const struct snd_kcontrol_new adcb_mux =
+	SOC_DAPM_ENUM("Right ADC Input Capture Mux", adcb_enum);
+
+static const char * const mic_bias_level_text[] = {
+	"0.5 +VA", "0.6 +VA", "0.7 +VA",
+	"0.8 +VA", "0.83 +VA", "0.91 +VA"
+};
+
+static const struct soc_enum mic_bias_level_enum =
+	SOC_ENUM_SINGLE(CS42L52_IFACE_CTL1, 0,
+			ARRAY_SIZE(mic_bias_level_text), mic_bias_level_text);
+
+static const char * const cs42l52_mic_text[] = { "Single", "Differential" };
+
+static const struct soc_enum mica_enum =
+	SOC_ENUM_SINGLE(CS42L52_MICA_CTL, 5,
+			ARRAY_SIZE(cs42l52_mic_text), cs42l52_mic_text);
+
+static const struct soc_enum micb_enum =
+	SOC_ENUM_SINGLE(CS42L52_MICB_CTL, 5,
+			ARRAY_SIZE(cs42l52_mic_text), cs42l52_mic_text);
+
+static const struct snd_kcontrol_new mica_mux =
+	SOC_DAPM_ENUM("Left Mic Input Capture Mux", mica_enum);
+
+static const struct snd_kcontrol_new micb_mux =
+	SOC_DAPM_ENUM("Right Mic Input Capture Mux", micb_enum);
+
+static const char * const digital_output_mux_text[] = {"ADC", "DSP"};
+
+static const struct soc_enum digital_output_mux_enum =
+	SOC_ENUM_SINGLE(CS42L52_ADC_MISC_CTL, 6,
+			ARRAY_SIZE(digital_output_mux_text),
+			digital_output_mux_text);
+
+static const struct snd_kcontrol_new digital_output_mux =
+	SOC_DAPM_ENUM("Digital Output Mux", digital_output_mux_enum);
+
+static const char * const hp_gain_num_text[] = {
+	"0.3959", "0.4571", "0.5111", "0.6047",
+	"0.7099", "0.8399", "1.000", "1.1430"
+};
+
+static const struct soc_enum hp_gain_enum =
+	SOC_ENUM_SINGLE(CS42L52_PB_CTL1, 4,
+		ARRAY_SIZE(hp_gain_num_text), hp_gain_num_text);
+
+static const char * const beep_pitch_text[] = {
+	"C4", "C5", "D5", "E5", "F5", "G5", "A5", "B5",
+	"C6", "D6", "E6", "F6", "G6", "A6", "B6", "C7"
+};
+
+static const struct soc_enum beep_pitch_enum =
+	SOC_ENUM_SINGLE(CS42L52_BEEP_FREQ, 4,
+			ARRAY_SIZE(beep_pitch_text), beep_pitch_text);
+
+static const char * const beep_ontime_text[] = {
+	"86 ms", "430 ms", "780 ms", "1.20 s", "1.50 s",
+	"1.80 s", "2.20 s", "2.50 s", "2.80 s", "3.20 s",
+	"3.50 s", "3.80 s", "4.20 s", "4.50 s", "4.80 s", "5.20 s"
+};
+
+static const struct soc_enum beep_ontime_enum =
+	SOC_ENUM_SINGLE(CS42L52_BEEP_FREQ, 0,
+			ARRAY_SIZE(beep_ontime_text), beep_ontime_text);
+
+static const char * const beep_offtime_text[] = {
+	"1.23 s", "2.58 s", "3.90 s", "5.20 s",
+	"6.60 s", "8.05 s", "9.35 s", "10.80 s"
+};
+
+static const struct soc_enum beep_offtime_enum =
+	SOC_ENUM_SINGLE(CS42L52_BEEP_VOL, 5,
+			ARRAY_SIZE(beep_offtime_text), beep_offtime_text);
+
+static const char * const beep_config_text[] = {
+	"Off", "Single", "Multiple", "Continuous"
+};
+
+static const struct soc_enum beep_config_enum =
+	SOC_ENUM_SINGLE(CS42L52_BEEP_TONE_CTL, 6,
+			ARRAY_SIZE(beep_config_text), beep_config_text);
+
+static const char * const beep_bass_text[] = {
+	"50 Hz", "100 Hz", "200 Hz", "250 Hz"
+};
+
+static const struct soc_enum beep_bass_enum =
+	SOC_ENUM_SINGLE(CS42L52_BEEP_TONE_CTL, 1,
+			ARRAY_SIZE(beep_bass_text), beep_bass_text);
+
+static const char * const beep_treble_text[] = {
+	"5 kHz", "7 kHz", "10 kHz", " 15 kHz"
+};
+
+static const struct soc_enum beep_treble_enum =
+	SOC_ENUM_SINGLE(CS42L52_BEEP_TONE_CTL, 3,
+			ARRAY_SIZE(beep_treble_text), beep_treble_text);
+
+static const char * const ng_threshold_text[] = {
+	"-34dB", "-37dB", "-40dB", "-43dB",
+	"-46dB", "-52dB", "-58dB", "-64dB"
+};
+
+static const struct soc_enum ng_threshold_enum =
+	SOC_ENUM_SINGLE(CS42L52_NOISE_GATE_CTL, 2,
+		ARRAY_SIZE(ng_threshold_text), ng_threshold_text);
+
+static const char * const cs42l52_ng_delay_text[] = {
+	"50ms", "100ms", "150ms", "200ms"};
+
+static const struct soc_enum ng_delay_enum =
+	SOC_ENUM_SINGLE(CS42L52_NOISE_GATE_CTL, 0,
+		ARRAY_SIZE(cs42l52_ng_delay_text), cs42l52_ng_delay_text);
+
+static const char * const cs42l52_ng_type_text[] = {
+	"Apply Specific", "Apply All"
+};
+
+static const struct soc_enum ng_type_enum =
+	SOC_ENUM_SINGLE(CS42L52_NOISE_GATE_CTL, 6,
+		ARRAY_SIZE(cs42l52_ng_type_text), cs42l52_ng_type_text);
+
+static const char * const left_swap_text[] = {
+	"Left", "LR 2", "Right"};
+
+static const char * const right_swap_text[] = {
+	"Right", "LR 2", "Left"};
+
+static const unsigned int swap_values[] = { 0, 1, 3 };
+
+static const struct soc_enum adca_swap_enum =
+	SOC_VALUE_ENUM_SINGLE(CS42L52_ADC_PCM_MIXER, 2, 1,
+			      ARRAY_SIZE(left_swap_text),
+			      left_swap_text,
+			      swap_values);
+
+static const struct snd_kcontrol_new adca_mixer =
+	SOC_DAPM_ENUM("Route", adca_swap_enum);
+
+static const struct soc_enum pcma_swap_enum =
+	SOC_VALUE_ENUM_SINGLE(CS42L52_ADC_PCM_MIXER, 6, 1,
+			      ARRAY_SIZE(left_swap_text),
+			      left_swap_text,
+			      swap_values);
+
+static const struct snd_kcontrol_new pcma_mixer =
+	SOC_DAPM_ENUM("Route", pcma_swap_enum);
+
+static const struct soc_enum adcb_swap_enum =
+	SOC_VALUE_ENUM_SINGLE(CS42L52_ADC_PCM_MIXER, 0, 1,
+			      ARRAY_SIZE(right_swap_text),
+			      right_swap_text,
+			      swap_values);
+
+static const struct snd_kcontrol_new adcb_mixer =
+	SOC_DAPM_ENUM("Route", adcb_swap_enum);
+
+static const struct soc_enum pcmb_swap_enum =
+	SOC_VALUE_ENUM_SINGLE(CS42L52_ADC_PCM_MIXER, 4, 1,
+			      ARRAY_SIZE(right_swap_text),
+			      right_swap_text,
+			      swap_values);
+
+static const struct snd_kcontrol_new pcmb_mixer =
+	SOC_DAPM_ENUM("Route", pcmb_swap_enum);
+
+
+static const struct snd_kcontrol_new passthrul_ctl =
+	SOC_DAPM_SINGLE("Switch", CS42L52_MISC_CTL, 6, 1, 0);
+
+static const struct snd_kcontrol_new passthrur_ctl =
+	SOC_DAPM_SINGLE("Switch", CS42L52_MISC_CTL, 7, 1, 0);
+
+static const struct snd_kcontrol_new spkl_ctl =
+	SOC_DAPM_SINGLE("Switch", CS42L52_PWRCTL3, 0, 1, 1);
+
+static const struct snd_kcontrol_new spkr_ctl =
+	SOC_DAPM_SINGLE("Switch", CS42L52_PWRCTL3, 2, 1, 1);
+
+static const struct snd_kcontrol_new hpl_ctl =
+	SOC_DAPM_SINGLE("Switch", CS42L52_PWRCTL3, 4, 1, 1);
+
+static const struct snd_kcontrol_new hpr_ctl =
+	SOC_DAPM_SINGLE("Switch", CS42L52_PWRCTL3, 6, 1, 1);
+
+static const struct snd_kcontrol_new cs42l52_snd_controls[] = {
+
+	SOC_DOUBLE_R_SX_TLV("Master Volume", CS42L52_MASTERA_VOL,
+			      CS42L52_MASTERB_VOL, 0, 0x34, 0xE4, hl_tlv),
+
+	SOC_DOUBLE_R_SX_TLV("Headphone Volume", CS42L52_HPA_VOL,
+			      CS42L52_HPB_VOL, 0, 0x34, 0xCC, hpd_tlv),
+
+	SOC_ENUM("Headphone Analog Gain", hp_gain_enum),
+
+	SOC_DOUBLE_R_SX_TLV("Speaker Volume", CS42L52_SPKA_VOL,
+			      CS42L52_SPKB_VOL, 7, 0x1, 0xff, hl_tlv),
+
+	SOC_DOUBLE_R_SX_TLV("Bypass Volume", CS42L52_PASSTHRUA_VOL,
+			      CS42L52_PASSTHRUB_VOL, 6, 0x18, 0x90, pga_tlv),
+
+	SOC_DOUBLE("Bypass Mute", CS42L52_MISC_CTL, 4, 5, 1, 0),
+
+	SOC_DOUBLE_R_TLV("MIC Gain Volume", CS42L52_MICA_CTL,
+			      CS42L52_MICB_CTL, 0, 0x10, 0, mic_tlv),
+
+	SOC_ENUM("MIC Bias Level", mic_bias_level_enum),
+
+	SOC_DOUBLE_R_SX_TLV("ADC Volume", CS42L52_ADCA_VOL,
+			      CS42L52_ADCB_VOL, 7, 0x80, 0xA0, ipd_tlv),
+	SOC_DOUBLE_R_SX_TLV("ADC Mixer Volume",
+			     CS42L52_ADCA_MIXER_VOL, CS42L52_ADCB_MIXER_VOL,
+				6, 0x7f, 0x19, ipd_tlv),
+
+	SOC_DOUBLE("ADC Switch", CS42L52_ADC_MISC_CTL, 0, 1, 1, 0),
+
+	SOC_DOUBLE_R("ADC Mixer Switch", CS42L52_ADCA_MIXER_VOL,
+		     CS42L52_ADCB_MIXER_VOL, 7, 1, 1),
+
+	SOC_DOUBLE_R_SX_TLV("PGA Volume", CS42L52_PGAA_CTL,
+			    CS42L52_PGAB_CTL, 0, 0x28, 0x30, pga_tlv),
+
+	SOC_DOUBLE_R_SX_TLV("PCM Mixer Volume",
+			    CS42L52_PCMA_MIXER_VOL, CS42L52_PCMB_MIXER_VOL,
+				6, 0x7f, 0x19, hl_tlv),
+	SOC_DOUBLE_R("PCM Mixer Switch",
+		     CS42L52_PCMA_MIXER_VOL, CS42L52_PCMB_MIXER_VOL, 7, 1, 1),
+
+	SOC_ENUM("Beep Config", beep_config_enum),
+	SOC_ENUM("Beep Pitch", beep_pitch_enum),
+	SOC_ENUM("Beep on Time", beep_ontime_enum),
+	SOC_ENUM("Beep off Time", beep_offtime_enum),
+	SOC_SINGLE_TLV("Beep Volume", CS42L52_BEEP_VOL, 0, 0x1f, 0x07, hl_tlv),
+	SOC_SINGLE("Beep Mixer Switch", CS42L52_BEEP_TONE_CTL, 5, 1, 1),
+	SOC_ENUM("Beep Treble Corner Freq", beep_treble_enum),
+	SOC_ENUM("Beep Bass Corner Freq", beep_bass_enum),
+
+	SOC_SINGLE("Tone Control Switch", CS42L52_BEEP_TONE_CTL, 0, 1, 1),
+	SOC_SINGLE_TLV("Treble Gain Volume",
+			    CS42L52_TONE_CTL, 4, 15, 1, hl_tlv),
+	SOC_SINGLE_TLV("Bass Gain Volume",
+			    CS42L52_TONE_CTL, 0, 15, 1, hl_tlv),
+
+	/* Limiter */
+	SOC_SINGLE_TLV("Limiter Max Threshold Volume",
+		       CS42L52_LIMITER_CTL1, 5, 7, 0, limiter_tlv),
+	SOC_SINGLE_TLV("Limiter Cushion Threshold Volume",
+		       CS42L52_LIMITER_CTL1, 2, 7, 0, limiter_tlv),
+	SOC_SINGLE_TLV("Limiter Release Rate Volume",
+		       CS42L52_LIMITER_CTL2, 0, 63, 0, limiter_tlv),
+	SOC_SINGLE_TLV("Limiter Attack Rate Volume",
+		       CS42L52_LIMITER_AT_RATE, 0, 63, 0, limiter_tlv),
+
+	SOC_SINGLE("Limiter SR Switch", CS42L52_LIMITER_CTL1, 1, 1, 0),
+	SOC_SINGLE("Limiter ZC Switch", CS42L52_LIMITER_CTL1, 0, 1, 0),
+	SOC_SINGLE("Limiter Switch", CS42L52_LIMITER_CTL2, 7, 1, 0),
+
+	/* ALC */
+	SOC_SINGLE_TLV("ALC Attack Rate Volume", CS42L52_ALC_CTL,
+		       0, 63, 0, limiter_tlv),
+	SOC_SINGLE_TLV("ALC Release Rate Volume", CS42L52_ALC_RATE,
+		       0, 63, 0, limiter_tlv),
+	SOC_SINGLE_TLV("ALC Max Threshold Volume", CS42L52_ALC_THRESHOLD,
+		       5, 7, 0, limiter_tlv),
+	SOC_SINGLE_TLV("ALC Min Threshold Volume", CS42L52_ALC_THRESHOLD,
+		       2, 7, 0, limiter_tlv),
+
+	SOC_DOUBLE_R("ALC SR Capture Switch", CS42L52_PGAA_CTL,
+		     CS42L52_PGAB_CTL, 7, 1, 1),
+	SOC_DOUBLE_R("ALC ZC Capture Switch", CS42L52_PGAA_CTL,
+		     CS42L52_PGAB_CTL, 6, 1, 1),
+	SOC_DOUBLE("ALC Capture Switch", CS42L52_ALC_CTL, 6, 7, 1, 0),
+
+	/* Noise gate */
+	SOC_ENUM("NG Type Switch", ng_type_enum),
+	SOC_SINGLE("NG Enable Switch", CS42L52_NOISE_GATE_CTL, 6, 1, 0),
+	SOC_SINGLE("NG Boost Switch", CS42L52_NOISE_GATE_CTL, 5, 1, 1),
+	SOC_ENUM("NG Threshold", ng_threshold_enum),
+	SOC_ENUM("NG Delay", ng_delay_enum),
+
+	SOC_DOUBLE("HPF Switch", CS42L52_ANALOG_HPF_CTL, 5, 7, 1, 0),
+
+	SOC_DOUBLE("Analog SR Switch", CS42L52_ANALOG_HPF_CTL, 1, 3, 1, 1),
+	SOC_DOUBLE("Analog ZC Switch", CS42L52_ANALOG_HPF_CTL, 0, 2, 1, 1),
+	SOC_SINGLE("Digital SR Switch", CS42L52_MISC_CTL, 1, 1, 0),
+	SOC_SINGLE("Digital ZC Switch", CS42L52_MISC_CTL, 0, 1, 0),
+	SOC_SINGLE("Deemphasis Switch", CS42L52_MISC_CTL, 2, 1, 0),
+
+	SOC_SINGLE("Batt Compensation Switch", CS42L52_BATT_COMPEN, 7, 1, 0),
+	SOC_SINGLE("Batt VP Monitor Switch", CS42L52_BATT_COMPEN, 6, 1, 0),
+	SOC_SINGLE("Batt VP ref", CS42L52_BATT_COMPEN, 0, 0x0f, 0),
+
+	SOC_SINGLE("PGA AIN1L Switch", CS42L52_ADC_PGA_A, 0, 1, 0),
+	SOC_SINGLE("PGA AIN1R Switch", CS42L52_ADC_PGA_B, 0, 1, 0),
+	SOC_SINGLE("PGA AIN2L Switch", CS42L52_ADC_PGA_A, 1, 1, 0),
+	SOC_SINGLE("PGA AIN2R Switch", CS42L52_ADC_PGA_B, 1, 1, 0),
+
+	SOC_SINGLE("PGA AIN3L Switch", CS42L52_ADC_PGA_A, 2, 1, 0),
+	SOC_SINGLE("PGA AIN3R Switch", CS42L52_ADC_PGA_B, 2, 1, 0),
+
+	SOC_SINGLE("PGA AIN4L Switch", CS42L52_ADC_PGA_A, 3, 1, 0),
+	SOC_SINGLE("PGA AIN4R Switch", CS42L52_ADC_PGA_B, 3, 1, 0),
+
+	SOC_SINGLE("PGA MICA Switch", CS42L52_ADC_PGA_A, 4, 1, 0),
+	SOC_SINGLE("PGA MICB Switch", CS42L52_ADC_PGA_B, 4, 1, 0),
+
+};
+
+static const struct snd_soc_dapm_widget cs42l52_dapm_widgets[] = {
+
+	SND_SOC_DAPM_INPUT("AIN1L"),
+	SND_SOC_DAPM_INPUT("AIN1R"),
+	SND_SOC_DAPM_INPUT("AIN2L"),
+	SND_SOC_DAPM_INPUT("AIN2R"),
+	SND_SOC_DAPM_INPUT("AIN3L"),
+	SND_SOC_DAPM_INPUT("AIN3R"),
+	SND_SOC_DAPM_INPUT("AIN4L"),
+	SND_SOC_DAPM_INPUT("AIN4R"),
+	SND_SOC_DAPM_INPUT("MICA"),
+	SND_SOC_DAPM_INPUT("MICB"),
+	SND_SOC_DAPM_SIGGEN("Beep"),
+
+	SND_SOC_DAPM_AIF_OUT("AIFOUTL", NULL,  0,
+			SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_OUT("AIFOUTR", NULL,  0,
+			SND_SOC_NOPM, 0, 0),
+
+	SND_SOC_DAPM_MUX("MICA Mux", SND_SOC_NOPM, 0, 0, &mica_mux),
+	SND_SOC_DAPM_MUX("MICB Mux", SND_SOC_NOPM, 0, 0, &micb_mux),
+
+	SND_SOC_DAPM_ADC("ADC Left", NULL, CS42L52_PWRCTL1, 1, 1),
+	SND_SOC_DAPM_ADC("ADC Right", NULL, CS42L52_PWRCTL1, 2, 1),
+	SND_SOC_DAPM_PGA("PGA Left", CS42L52_PWRCTL1, 3, 1, NULL, 0),
+	SND_SOC_DAPM_PGA("PGA Right", CS42L52_PWRCTL1, 4, 1, NULL, 0),
+
+	SND_SOC_DAPM_MUX("ADC Left Mux", SND_SOC_NOPM, 0, 0, &adca_mux),
+	SND_SOC_DAPM_MUX("ADC Right Mux", SND_SOC_NOPM, 0, 0, &adcb_mux),
+
+	SND_SOC_DAPM_MUX("ADC Left Swap", SND_SOC_NOPM,
+			 0, 0, &adca_mixer),
+	SND_SOC_DAPM_MUX("ADC Right Swap", SND_SOC_NOPM,
+			 0, 0, &adcb_mixer),
+
+	SND_SOC_DAPM_MUX("Output Mux", SND_SOC_NOPM,
+			 0, 0, &digital_output_mux),
+
+	SND_SOC_DAPM_PGA("PGA MICA", CS42L52_PWRCTL2, 1, 1, NULL, 0),
+	SND_SOC_DAPM_PGA("PGA MICB", CS42L52_PWRCTL2, 2, 1, NULL, 0),
+
+	SND_SOC_DAPM_SUPPLY("Mic Bias", CS42L52_PWRCTL2, 0, 1, NULL, 0),
+	SND_SOC_DAPM_SUPPLY("Charge Pump", CS42L52_PWRCTL1, 7, 1, NULL, 0),
+
+	SND_SOC_DAPM_AIF_IN("AIFINL", NULL,  0,
+			SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_IN("AIFINR", NULL,  0,
+			SND_SOC_NOPM, 0, 0),
+
+	SND_SOC_DAPM_DAC("DAC Left", NULL, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_DAC("DAC Right", NULL, SND_SOC_NOPM, 0, 0),
+
+	SND_SOC_DAPM_SWITCH("Bypass Left", CS42L52_MISC_CTL,
+			    6, 0, &passthrul_ctl),
+	SND_SOC_DAPM_SWITCH("Bypass Right", CS42L52_MISC_CTL,
+			    7, 0, &passthrur_ctl),
+
+	SND_SOC_DAPM_MUX("PCM Left Swap", SND_SOC_NOPM,
+			 0, 0, &pcma_mixer),
+	SND_SOC_DAPM_MUX("PCM Right Swap", SND_SOC_NOPM,
+			 0, 0, &pcmb_mixer),
+
+	SND_SOC_DAPM_SWITCH("HP Left Amp", SND_SOC_NOPM, 0, 0, &hpl_ctl),
+	SND_SOC_DAPM_SWITCH("HP Right Amp", SND_SOC_NOPM, 0, 0, &hpr_ctl),
+
+	SND_SOC_DAPM_SWITCH("SPK Left Amp", SND_SOC_NOPM, 0, 0, &spkl_ctl),
+	SND_SOC_DAPM_SWITCH("SPK Right Amp", SND_SOC_NOPM, 0, 0, &spkr_ctl),
+
+	SND_SOC_DAPM_OUTPUT("HPOUTA"),
+	SND_SOC_DAPM_OUTPUT("HPOUTB"),
+	SND_SOC_DAPM_OUTPUT("SPKOUTA"),
+	SND_SOC_DAPM_OUTPUT("SPKOUTB"),
+
+};
+
+static const struct snd_soc_dapm_route cs42l52_audio_map[] = {
+
+	{"Capture", NULL, "AIFOUTL"},
+	{"Capture", NULL, "AIFOUTL"},
+
+	{"AIFOUTL", NULL, "Output Mux"},
+	{"AIFOUTR", NULL, "Output Mux"},
+
+	{"Output Mux", "ADC", "ADC Left"},
+	{"Output Mux", "ADC", "ADC Right"},
+
+	{"ADC Left", NULL, "Charge Pump"},
+	{"ADC Right", NULL, "Charge Pump"},
+
+	{"Charge Pump", NULL, "ADC Left Mux"},
+	{"Charge Pump", NULL, "ADC Right Mux"},
+
+	{"ADC Left Mux", "Input1A", "AIN1L"},
+	{"ADC Right Mux", "Input1B", "AIN1R"},
+	{"ADC Left Mux", "Input2A", "AIN2L"},
+	{"ADC Right Mux", "Input2B", "AIN2R"},
+	{"ADC Left Mux", "Input3A", "AIN3L"},
+	{"ADC Right Mux", "Input3B", "AIN3R"},
+	{"ADC Left Mux", "Input4A", "AIN4L"},
+	{"ADC Right Mux", "Input4B", "AIN4R"},
+	{"ADC Left Mux", "PGA Input Left", "PGA Left"},
+	{"ADC Right Mux", "PGA Input Right" , "PGA Right"},
+
+	{"PGA Left", "Switch", "AIN1L"},
+	{"PGA Right", "Switch", "AIN1R"},
+	{"PGA Left", "Switch", "AIN2L"},
+	{"PGA Right", "Switch", "AIN2R"},
+	{"PGA Left", "Switch", "AIN3L"},
+	{"PGA Right", "Switch", "AIN3R"},
+	{"PGA Left", "Switch", "AIN4L"},
+	{"PGA Right", "Switch", "AIN4R"},
+
+	{"PGA Left", "Switch", "PGA MICA"},
+	{"PGA MICA", NULL, "MICA"},
+
+	{"PGA Right", "Switch", "PGA MICB"},
+	{"PGA MICB", NULL, "MICB"},
+
+	{"HPOUTA", NULL, "HP Left Amp"},
+	{"HPOUTB", NULL, "HP Right Amp"},
+	{"HP Left Amp", NULL, "Bypass Left"},
+	{"HP Right Amp", NULL, "Bypass Right"},
+	{"Bypass Left", "Switch", "PGA Left"},
+	{"Bypass Right", "Switch", "PGA Right"},
+	{"HP Left Amp", "Switch", "DAC Left"},
+	{"HP Right Amp", "Switch", "DAC Right"},
+
+	{"SPKOUTA", NULL, "SPK Left Amp"},
+	{"SPKOUTB", NULL, "SPK Right Amp"},
+
+	{"SPK Left Amp", NULL, "Beep"},
+	{"SPK Right Amp", NULL, "Beep"},
+	{"SPK Left Amp", "Switch", "Playback"},
+	{"SPK Right Amp", "Switch", "Playback"},
+
+	{"DAC Left", NULL, "Beep"},
+	{"DAC Right", NULL, "Beep"},
+	{"DAC Left", NULL, "Playback"},
+	{"DAC Right", NULL, "Playback"},
+
+	{"Output Mux", "DSP", "Playback"},
+	{"Output Mux", "DSP", "Playback"},
+
+	{"AIFINL", NULL, "Playback"},
+	{"AIFINR", NULL, "Playback"},
+
+};
+
+struct cs42l52_clk_para {
+	u32 mclk;
+	u32 rate;
+	u8 speed;
+	u8 group;
+	u8 videoclk;
+	u8 ratio;
+	u8 mclkdiv2;
+};
+
+static const struct cs42l52_clk_para clk_map_table[] = {
+	/*8k*/
+	{12288000, 8000, CLK_QS_MODE, CLK_32K, CLK_NO_27M, CLK_R_128, 0},
+	{18432000, 8000, CLK_QS_MODE, CLK_32K, CLK_NO_27M, CLK_R_128, 0},
+	{12000000, 8000, CLK_QS_MODE, CLK_32K, CLK_NO_27M, CLK_R_125, 0},
+	{24000000, 8000, CLK_QS_MODE, CLK_32K, CLK_NO_27M, CLK_R_125, 1},
+	{27000000, 8000, CLK_QS_MODE, CLK_32K, CLK_27M_MCLK, CLK_R_125, 0},
+
+	/*11.025k*/
+	{11289600, 11025, CLK_QS_MODE, CLK_NO_32K, CLK_NO_27M, CLK_R_128, 0},
+	{16934400, 11025, CLK_QS_MODE, CLK_NO_32K, CLK_NO_27M, CLK_R_128, 0},
+
+	/*16k*/
+	{12288000, 16000, CLK_HS_MODE, CLK_32K, CLK_NO_27M, CLK_R_128, 0},
+	{18432000, 16000, CLK_HS_MODE, CLK_32K, CLK_NO_27M, CLK_R_128, 0},
+	{12000000, 16000, CLK_HS_MODE, CLK_32K, CLK_NO_27M, CLK_R_125, 0},
+	{24000000, 16000, CLK_HS_MODE, CLK_32K, CLK_NO_27M, CLK_R_125, 1},
+	{27000000, 16000, CLK_HS_MODE, CLK_32K, CLK_27M_MCLK, CLK_R_125, 1},
+
+	/*22.05k*/
+	{11289600, 22050, CLK_HS_MODE, CLK_NO_32K, CLK_NO_27M, CLK_R_128, 0},
+	{16934400, 22050, CLK_HS_MODE, CLK_NO_32K, CLK_NO_27M, CLK_R_128, 0},
+
+	/* 32k */
+	{12288000, 32000, CLK_SS_MODE, CLK_32K, CLK_NO_27M, CLK_R_128, 0},
+	{18432000, 32000, CLK_SS_MODE, CLK_32K, CLK_NO_27M, CLK_R_128, 0},
+	{12000000, 32000, CLK_SS_MODE, CLK_32K, CLK_NO_27M, CLK_R_125, 0},
+	{24000000, 32000, CLK_SS_MODE, CLK_32K, CLK_NO_27M, CLK_R_125, 1},
+	{27000000, 32000, CLK_SS_MODE, CLK_32K, CLK_27M_MCLK, CLK_R_125, 0},
+
+	/* 44.1k */
+	{11289600, 44100, CLK_SS_MODE, CLK_NO_32K, CLK_NO_27M, CLK_R_128, 0},
+	{16934400, 44100, CLK_SS_MODE, CLK_NO_32K, CLK_NO_27M, CLK_R_128, 0},
+
+	/* 48k */
+	{12288000, 48000, CLK_SS_MODE, CLK_NO_32K, CLK_NO_27M, CLK_R_128, 0},
+	{18432000, 48000, CLK_SS_MODE, CLK_NO_32K, CLK_NO_27M, CLK_R_128, 0},
+	{12000000, 48000, CLK_SS_MODE, CLK_NO_32K, CLK_NO_27M, CLK_R_125, 0},
+	{24000000, 48000, CLK_SS_MODE, CLK_NO_32K, CLK_NO_27M, CLK_R_125, 1},
+	{27000000, 48000, CLK_SS_MODE, CLK_NO_32K, CLK_27M_MCLK, CLK_R_125, 1},
+
+	/* 88.2k */
+	{11289600, 88200, CLK_DS_MODE, CLK_NO_32K, CLK_NO_27M, CLK_R_128, 0},
+	{16934400, 88200, CLK_DS_MODE, CLK_NO_32K, CLK_NO_27M, CLK_R_128, 0},
+
+	/* 96k */
+	{12288000, 96000, CLK_DS_MODE, CLK_NO_32K, CLK_NO_27M, CLK_R_128, 0},
+	{18432000, 96000, CLK_DS_MODE, CLK_NO_32K, CLK_NO_27M, CLK_R_128, 0},
+	{12000000, 96000, CLK_DS_MODE, CLK_NO_32K, CLK_NO_27M, CLK_R_125, 0},
+	{24000000, 96000, CLK_DS_MODE, CLK_NO_32K, CLK_NO_27M, CLK_R_125, 1},
+};
+
+static int cs42l52_get_clk(int mclk, int rate)
+{
+	int i, ret = 0;
+	u_int mclk1, mclk2 = 0;
+
+	for (i = 0; i < ARRAY_SIZE(clk_map_table); i++) {
+		if (clk_map_table[i].rate == rate) {
+			mclk1 = clk_map_table[i].mclk;
+			if (abs(mclk - mclk1) < abs(mclk - mclk2)) {
+				mclk2 = mclk1;
+				ret = i;
+			}
+		}
+	}
+	if (ret > ARRAY_SIZE(clk_map_table))
+		return -EINVAL;
+	return ret;
+}
+
+static int cs42l52_set_sysclk(struct snd_soc_dai *codec_dai,
+			int clk_id, unsigned int freq, int dir)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	struct cs42l52_private *cs42l52 = snd_soc_codec_get_drvdata(codec);
+
+	if ((freq >= CS42L52_MIN_CLK) && (freq <= CS42L52_MAX_CLK)) {
+		cs42l52->sysclk = freq;
+	} else {
+		dev_err(codec->dev, "Invalid freq paramter\n");
+		return -EINVAL;
+	}
+	return 0;
+}
+
+static int cs42l52_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	struct cs42l52_private *cs42l52 = snd_soc_codec_get_drvdata(codec);
+	int ret = 0;
+	u8 iface = 0;
+
+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBM_CFM:
+		iface = CS42L52_IFACE_CTL1_MASTER;
+		break;
+	case SND_SOC_DAIFMT_CBS_CFS:
+		iface = CS42L52_IFACE_CTL1_SLAVE;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	 /* interface format */
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+		iface |= CS42L52_IFACE_CTL1_ADC_FMT_I2S |
+				CS42L52_IFACE_CTL1_DAC_FMT_I2S;
+		break;
+	case SND_SOC_DAIFMT_RIGHT_J:
+		iface |= CS42L52_IFACE_CTL1_DAC_FMT_RIGHT_J;
+		break;
+	case SND_SOC_DAIFMT_LEFT_J:
+		iface |= CS42L52_IFACE_CTL1_ADC_FMT_LEFT_J |
+				CS42L52_IFACE_CTL1_DAC_FMT_LEFT_J;
+		break;
+	case SND_SOC_DAIFMT_DSP_A:
+		iface |= CS42L52_IFACE_CTL1_DSP_MODE_EN;
+		break;
+	case SND_SOC_DAIFMT_DSP_B:
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	/* clock inversion */
+	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+	case SND_SOC_DAIFMT_NB_NF:
+		break;
+	case SND_SOC_DAIFMT_IB_IF:
+		iface |= CS42L52_IFACE_CTL1_INV_SCLK;
+		break;
+	case SND_SOC_DAIFMT_IB_NF:
+		iface |= CS42L52_IFACE_CTL1_INV_SCLK;
+		break;
+	case SND_SOC_DAIFMT_NB_IF:
+		break;
+	default:
+		ret = -EINVAL;
+	}
+	cs42l52->config.format = iface;
+	snd_soc_write(codec, CS42L52_IFACE_CTL1, cs42l52->config.format);
+
+	return 0;
+}
+
+static int cs42l52_digital_mute(struct snd_soc_dai *dai, int mute)
+{
+	struct snd_soc_codec *codec = dai->codec;
+
+	if (mute)
+		snd_soc_update_bits(codec, CS42L52_PB_CTL1,
+				    CS42L52_PB_CTL1_MUTE_MASK,
+				CS42L52_PB_CTL1_MUTE);
+	else
+		snd_soc_update_bits(codec, CS42L52_PB_CTL1,
+				    CS42L52_PB_CTL1_MUTE_MASK,
+				CS42L52_PB_CTL1_UNMUTE);
+
+	return 0;
+}
+
+static int cs42l52_pcm_hw_params(struct snd_pcm_substream *substream,
+				     struct snd_pcm_hw_params *params,
+				     struct snd_soc_dai *dai)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	struct cs42l52_private *cs42l52 = snd_soc_codec_get_drvdata(codec);
+	u32 clk = 0;
+	int index;
+
+	index = cs42l52_get_clk(cs42l52->sysclk, params_rate(params));
+	if (index >= 0) {
+		cs42l52->sysclk = clk_map_table[index].mclk;
+
+		clk |= (clk_map_table[index].speed << CLK_SPEED_SHIFT) |
+		(clk_map_table[index].group << CLK_32K_SR_SHIFT) |
+		(clk_map_table[index].videoclk << CLK_27M_MCLK_SHIFT) |
+		(clk_map_table[index].ratio << CLK_RATIO_SHIFT) |
+		clk_map_table[index].mclkdiv2;
+
+		snd_soc_write(codec, CS42L52_CLK_CTL, clk);
+	} else {
+		dev_err(codec->dev, "can't get correct mclk\n");
+		return -EINVAL;
+	}
+
+	return 0;
+}
+
+static int cs42l52_set_bias_level(struct snd_soc_codec *codec,
+					enum snd_soc_bias_level level)
+{
+	struct cs42l52_private *cs42l52 = snd_soc_codec_get_drvdata(codec);
+
+	switch (level) {
+	case SND_SOC_BIAS_ON:
+		break;
+	case SND_SOC_BIAS_PREPARE:
+		snd_soc_update_bits(codec, CS42L52_PWRCTL1,
+				    CS42L52_PWRCTL1_PDN_CODEC, 0);
+		break;
+	case SND_SOC_BIAS_STANDBY:
+		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
+			regcache_cache_only(cs42l52->regmap, false);
+			regcache_sync(cs42l52->regmap);
+		}
+		snd_soc_write(codec, CS42L52_PWRCTL1, CS42L52_PWRCTL1_PDN_ALL);
+		break;
+	case SND_SOC_BIAS_OFF:
+		snd_soc_write(codec, CS42L52_PWRCTL1, CS42L52_PWRCTL1_PDN_ALL);
+		regcache_cache_only(cs42l52->regmap, true);
+		break;
+	}
+	codec->dapm.bias_level = level;
+
+	return 0;
+}
+
+#define CS42L52_RATES (SNDRV_PCM_RATE_8000_96000)
+
+#define CS42L52_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_U16_LE | \
+			SNDRV_PCM_FMTBIT_S18_3LE | SNDRV_PCM_FMTBIT_U18_3LE | \
+			SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_U20_3LE | \
+			SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_U24_LE)
+
+static struct snd_soc_dai_ops cs42l52_ops = {
+	.hw_params	= cs42l52_pcm_hw_params,
+	.digital_mute	= cs42l52_digital_mute,
+	.set_fmt	= cs42l52_set_fmt,
+	.set_sysclk	= cs42l52_set_sysclk,
+};
+
+static struct snd_soc_dai_driver cs42l52_dai = {
+		.name = "cs42l52",
+		.playback = {
+			.stream_name = "Playback",
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = CS42L52_RATES,
+			.formats = CS42L52_FORMATS,
+		},
+		.capture = {
+			.stream_name = "Capture",
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = CS42L52_RATES,
+			.formats = CS42L52_FORMATS,
+		},
+		.ops = &cs42l52_ops,
+};
+
+static int cs42l52_suspend(struct snd_soc_codec *codec)
+{
+	cs42l52_set_bias_level(codec, SND_SOC_BIAS_OFF);
+
+	return 0;
+}
+
+static int cs42l52_resume(struct snd_soc_codec *codec)
+{
+	cs42l52_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+	return 0;
+}
+
+#if defined(CONFIG_INPUT) || defined(CONFIG_INPUT_MODULE)
+static int beep_rates[] = {
+	261, 522, 585, 667, 706, 774, 889, 1000,
+	1043, 1200, 1333, 1412, 1600, 1714, 2000, 2182
+};
+
+static void cs42l52_beep_work(struct work_struct *work)
+{
+	struct cs42l52_private *cs42l52 =
+		container_of(work, struct cs42l52_private, beep_work);
+	struct snd_soc_codec *codec = cs42l52->codec;
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
+	int i;
+	int val = 0;
+	int best = 0;
+
+	if (cs42l52->beep_rate) {
+		for (i = 0; i < ARRAY_SIZE(beep_rates); i++) {
+			if (abs(cs42l52->beep_rate - beep_rates[i]) <
+			    abs(cs42l52->beep_rate - beep_rates[best]))
+				best = i;
+		}
+
+		dev_dbg(codec->dev, "Set beep rate %dHz for requested %dHz\n",
+			beep_rates[best], cs42l52->beep_rate);
+
+		val = (best << CS42L52_BEEP_RATE_SHIFT);
+
+		snd_soc_dapm_enable_pin(dapm, "Beep");
+	} else {
+		dev_dbg(codec->dev, "Disabling beep\n");
+		snd_soc_dapm_disable_pin(dapm, "Beep");
+	}
+
+	snd_soc_update_bits(codec, CS42L52_BEEP_FREQ,
+			    CS42L52_BEEP_RATE_MASK, val);
+
+	snd_soc_dapm_sync(dapm);
+}
+
+/* For usability define a way of injecting beep events for the device -
+ * many systems will not have a keyboard.
+ */
+static int cs42l52_beep_event(struct input_dev *dev, unsigned int type,
+			     unsigned int code, int hz)
+{
+	struct snd_soc_codec *codec = input_get_drvdata(dev);
+	struct cs42l52_private *cs42l52 = snd_soc_codec_get_drvdata(codec);
+
+	dev_dbg(codec->dev, "Beep event %x %x\n", code, hz);
+
+	switch (code) {
+	case SND_BELL:
+		if (hz)
+			hz = 261;
+	case SND_TONE:
+		break;
+	default:
+		return -1;
+	}
+
+	/* Kick the beep from a workqueue */
+	cs42l52->beep_rate = hz;
+	schedule_work(&cs42l52->beep_work);
+	return 0;
+}
+
+static ssize_t cs42l52_beep_set(struct device *dev,
+			       struct device_attribute *attr,
+			       const char *buf, size_t count)
+{
+	struct cs42l52_private *cs42l52 = dev_get_drvdata(dev);
+	long int time;
+	int ret;
+
+	ret = kstrtol(buf, 10, &time);
+	if (ret != 0)
+		return ret;
+
+	input_event(cs42l52->beep, EV_SND, SND_TONE, time);
+
+	return count;
+}
+
+static DEVICE_ATTR(beep, 0200, NULL, cs42l52_beep_set);
+
+static void cs42l52_init_beep(struct snd_soc_codec *codec)
+{
+	struct cs42l52_private *cs42l52 = snd_soc_codec_get_drvdata(codec);
+	int ret;
+
+	cs42l52->beep = input_allocate_device();
+	if (!cs42l52->beep) {
+		dev_err(codec->dev, "Failed to allocate beep device\n");
+		return;
+	}
+
+	INIT_WORK(&cs42l52->beep_work, cs42l52_beep_work);
+	cs42l52->beep_rate = 0;
+
+	cs42l52->beep->name = "CS42L52 Beep Generator";
+	cs42l52->beep->phys = dev_name(codec->dev);
+	cs42l52->beep->id.bustype = BUS_I2C;
+
+	cs42l52->beep->evbit[0] = BIT_MASK(EV_SND);
+	cs42l52->beep->sndbit[0] = BIT_MASK(SND_BELL) | BIT_MASK(SND_TONE);
+	cs42l52->beep->event = cs42l52_beep_event;
+	cs42l52->beep->dev.parent = codec->dev;
+	input_set_drvdata(cs42l52->beep, codec);
+
+	ret = input_register_device(cs42l52->beep);
+	if (ret != 0) {
+		input_free_device(cs42l52->beep);
+		cs42l52->beep = NULL;
+		dev_err(codec->dev, "Failed to register beep device\n");
+	}
+
+	ret = device_create_file(codec->dev, &dev_attr_beep);
+	if (ret != 0) {
+		dev_err(codec->dev, "Failed to create keyclick file: %d\n",
+			ret);
+	}
+}
+
+static void cs42l52_free_beep(struct snd_soc_codec *codec)
+{
+	struct cs42l52_private *cs42l52 = snd_soc_codec_get_drvdata(codec);
+
+	device_remove_file(codec->dev, &dev_attr_beep);
+	input_unregister_device(cs42l52->beep);
+	cancel_work_sync(&cs42l52->beep_work);
+	cs42l52->beep = NULL;
+
+	snd_soc_update_bits(codec, CS42L52_BEEP_TONE_CTL,
+			    CS42L52_BEEP_EN_MASK, 0);
+}
+#else
+static void cs42l52_init_beep(struct snd_soc_codec *codec)
+{
+}
+
+static void cs42l52_free_beep(struct snd_soc_codec *codec)
+{
+}
+#endif
+
+static int cs42l52_probe(struct snd_soc_codec *codec)
+{
+	struct cs42l52_private *cs42l52 = snd_soc_codec_get_drvdata(codec);
+	int ret;
+
+	codec->control_data = cs42l52->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(cs42l52->regmap, true);
+
+	cs42l52_init_beep(codec);
+
+	cs42l52_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+	cs42l52->sysclk = CS42L52_DEFAULT_CLK;
+	cs42l52->config.format = CS42L52_DEFAULT_FORMAT;
+
+	/* Set Platform MICx CFG */
+	snd_soc_update_bits(codec, CS42L52_MICA_CTL,
+			    CS42L52_MIC_CTL_TYPE_MASK,
+				cs42l52->pdata.mica_cfg <<
+				CS42L52_MIC_CTL_TYPE_SHIFT);
+
+	snd_soc_update_bits(codec, CS42L52_MICB_CTL,
+			    CS42L52_MIC_CTL_TYPE_MASK,
+				cs42l52->pdata.micb_cfg <<
+				CS42L52_MIC_CTL_TYPE_SHIFT);
+
+	/* if Single Ended, Get Mic_Select */
+	if (cs42l52->pdata.mica_cfg)
+		snd_soc_update_bits(codec, CS42L52_MICA_CTL,
+				    CS42L52_MIC_CTL_MIC_SEL_MASK,
+				cs42l52->pdata.mica_sel <<
+				CS42L52_MIC_CTL_MIC_SEL_SHIFT);
+	if (cs42l52->pdata.micb_cfg)
+		snd_soc_update_bits(codec, CS42L52_MICB_CTL,
+				    CS42L52_MIC_CTL_MIC_SEL_MASK,
+				cs42l52->pdata.micb_sel <<
+				CS42L52_MIC_CTL_MIC_SEL_SHIFT);
+
+	/* Set Platform Charge Pump Freq */
+	snd_soc_update_bits(codec, CS42L52_CHARGE_PUMP,
+			    CS42L52_CHARGE_PUMP_MASK,
+				cs42l52->pdata.chgfreq <<
+				CS42L52_CHARGE_PUMP_SHIFT);
+
+	/* Set Platform Bias Level */
+	snd_soc_update_bits(codec, CS42L52_IFACE_CTL2,
+			    CS42L52_IFACE_CTL2_BIAS_LVL,
+				cs42l52->pdata.micbias_lvl);
+
+	return ret;
+}
+
+static int cs42l52_remove(struct snd_soc_codec *codec)
+{
+	cs42l52_free_beep(codec);
+	cs42l52_set_bias_level(codec, SND_SOC_BIAS_OFF);
+
+	return 0;
+}
+
+static struct snd_soc_codec_driver soc_codec_dev_cs42l52 = {
+	.probe = cs42l52_probe,
+	.remove = cs42l52_remove,
+	.suspend = cs42l52_suspend,
+	.resume = cs42l52_resume,
+	.set_bias_level = cs42l52_set_bias_level,
+
+	.dapm_widgets = cs42l52_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(cs42l52_dapm_widgets),
+	.dapm_routes = cs42l52_audio_map,
+	.num_dapm_routes = ARRAY_SIZE(cs42l52_audio_map),
+
+	.controls = cs42l52_snd_controls,
+	.num_controls = ARRAY_SIZE(cs42l52_snd_controls),
+};
+
+/* Current and threshold powerup sequence Pg37 */
+static const struct reg_default cs42l52_threshold_patch[] = {
+
+	{ 0x00, 0x99 },
+	{ 0x3E, 0xBA },
+	{ 0x47, 0x80 },
+	{ 0x32, 0xBB },
+	{ 0x32, 0x3B },
+	{ 0x00, 0x00 },
+
+};
+
+static struct regmap_config cs42l52_regmap = {
+	.reg_bits = 8,
+	.val_bits = 8,
+
+	.max_register = CS42L52_MAX_REGISTER,
+	.reg_defaults = cs42l52_reg_defaults,
+	.num_reg_defaults = ARRAY_SIZE(cs42l52_reg_defaults),
+	.readable_reg = cs42l52_readable_register,
+	.volatile_reg = cs42l52_volatile_register,
+	.cache_type = REGCACHE_RBTREE,
+};
+
+static int cs42l52_i2c_probe(struct i2c_client *i2c_client,
+			     const struct i2c_device_id *id)
+{
+	struct cs42l52_private *cs42l52;
+	int ret;
+	unsigned int devid = 0;
+	unsigned int reg;
+
+	cs42l52 = devm_kzalloc(&i2c_client->dev, sizeof(struct cs42l52_private),
+			       GFP_KERNEL);
+	if (cs42l52 == NULL)
+		return -ENOMEM;
+	cs42l52->dev = &i2c_client->dev;
+
+	cs42l52->regmap = regmap_init_i2c(i2c_client, &cs42l52_regmap);
+	if (IS_ERR(cs42l52->regmap)) {
+		ret = PTR_ERR(cs42l52->regmap);
+		dev_err(&i2c_client->dev, "regmap_init() failed: %d\n", ret);
+		goto err;
+	}
+
+	i2c_set_clientdata(i2c_client, cs42l52);
+
+	if (dev_get_platdata(&i2c_client->dev))
+		memcpy(&cs42l52->pdata, dev_get_platdata(&i2c_client->dev),
+		       sizeof(cs42l52->pdata));
+
+	ret = regmap_register_patch(cs42l52->regmap, cs42l52_threshold_patch,
+				    ARRAY_SIZE(cs42l52_threshold_patch));
+	if (ret != 0)
+		dev_warn(cs42l52->dev, "Failed to apply regmap patch: %d\n",
+			 ret);
+
+	ret = regmap_read(cs42l52->regmap, CS42L52_CHIP, &reg);
+	devid = reg & CS42L52_CHIP_ID_MASK;
+	if (devid != CS42L52_CHIP_ID) {
+		ret = -ENODEV;
+		dev_err(&i2c_client->dev,
+			"CS42L52 Device ID (%X). Expected %X\n",
+			devid, CS42L52_CHIP_ID);
+		goto err_regmap;
+	}
+
+	regcache_cache_only(cs42l52->regmap, true);
+
+	ret =  snd_soc_register_codec(&i2c_client->dev,
+			&soc_codec_dev_cs42l52, &cs42l52_dai, 1);
+	if (ret < 0)
+		goto err_regmap;
+	return 0;
+
+err_regmap:
+	regmap_exit(cs42l52->regmap);
+
+err:
+	return ret;
+}
+
+static int cs42l52_i2c_remove(struct i2c_client *client)
+{
+	struct cs42l52_private *cs42l52 = i2c_get_clientdata(client);
+
+	snd_soc_unregister_codec(&client->dev);
+	regmap_exit(cs42l52->regmap);
+
+	return 0;
+}
+
+static const struct i2c_device_id cs42l52_id[] = {
+	{ "cs42l52", 0 },
+	{ }
+};
+MODULE_DEVICE_TABLE(i2c, cs42l52_id);
+
+static struct i2c_driver cs42l52_i2c_driver = {
+	.driver = {
+		.name = "cs42l52",
+		.owner = THIS_MODULE,
+	},
+	.id_table = cs42l52_id,
+	.probe =    cs42l52_i2c_probe,
+	.remove =   __devexit_p(cs42l52_i2c_remove),
+};
+
+module_i2c_driver(cs42l52_i2c_driver);
+
+MODULE_DESCRIPTION("ASoC CS42L52 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/cs42l52.h b/sound/soc/codecs/cs42l52.h
new file mode 100644
index 000000000000..60985c059071
--- /dev/null
+++ b/sound/soc/codecs/cs42l52.h
@@ -0,0 +1,274 @@
+/*
+ * cs42l52.h -- CS42L52 ALSA SoC audio driver
+ *
+ * Copyright 2012 CirrusLogic, Inc.
+ *
+ * Author: Georgi Vlaev <joe@nucleusys.com>
+ * Author: 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.
+ *
+ */
+
+#ifndef __CS42L52_H__
+#define __CS42L52_H__
+
+#define CS42L52_NAME				"CS42L52"
+#define CS42L52_DEFAULT_CLK			12000000
+#define CS42L52_MIN_CLK				11000000
+#define CS42L52_MAX_CLK				27000000
+#define CS42L52_DEFAULT_FORMAT			SNDRV_PCM_FMTBIT_S16_LE
+#define CS42L52_DEFAULT_MAX_CHANS		2
+#define CS42L52_SYSCLK				1
+
+#define CS42L52_CHIP_SWICTH			(1 << 17)
+#define CS42L52_ALL_IN_ONE			(1 << 16)
+#define CS42L52_CHIP_ONE			0x00
+#define CS42L52_CHIP_TWO			0x01
+#define CS42L52_CHIP_THR			0x02
+#define CS42L52_CHIP_MASK			0x0f
+
+#define CS42L52_FIX_BITS_CTL			0x00
+#define CS42L52_CHIP				0x01
+#define CS42L52_CHIP_ID				0xE0
+#define CS42L52_CHIP_ID_MASK			0xF8
+#define CS42L52_CHIP_REV_A0			0x00
+#define CS42L52_CHIP_REV_A1			0x01
+#define CS42L52_CHIP_REV_B0			0x02
+#define CS42L52_CHIP_REV_MASK			0x03
+
+#define CS42L52_PWRCTL1				0x02
+#define CS42L52_PWRCTL1_PDN_ALL			0x9F
+#define CS42L52_PWRCTL1_PDN_CHRG		0x80
+#define CS42L52_PWRCTL1_PDN_PGAB		0x10
+#define CS42L52_PWRCTL1_PDN_PGAA		0x08
+#define CS42L52_PWRCTL1_PDN_ADCB		0x04
+#define CS42L52_PWRCTL1_PDN_ADCA		0x02
+#define CS42L52_PWRCTL1_PDN_CODEC		0x01
+
+#define CS42L52_PWRCTL2				0x03
+#define CS42L52_PWRCTL2_OVRDB			(1 << 4)
+#define CS42L52_PWRCTL2_OVRDA			(1 << 3)
+#define	CS42L52_PWRCTL2_PDN_MICB		(1 << 2)
+#define CS42L52_PWRCTL2_PDN_MICB_SHIFT		2
+#define CS42L52_PWRCTL2_PDN_MICA		(1 << 1)
+#define CS42L52_PWRCTL2_PDN_MICA_SHIFT		1
+#define CS42L52_PWRCTL2_PDN_MICBIAS		(1 << 0)
+#define CS42L52_PWRCTL2_PDN_MICBIAS_SHIFT	0
+
+#define CS42L52_PWRCTL3				0x04
+#define CS42L52_PWRCTL3_HPB_PDN_SHIFT		6
+#define CS42L52_PWRCTL3_HPB_ON_LOW		0x00
+#define CS42L52_PWRCTL3_HPB_ON_HIGH		0x01
+#define CS42L52_PWRCTL3_HPB_ALWAYS_ON		0x02
+#define CS42L52_PWRCTL3_HPB_ALWAYS_OFF		0x03
+#define CS42L52_PWRCTL3_HPA_PDN_SHIFT		4
+#define CS42L52_PWRCTL3_HPA_ON_LOW		0x00
+#define CS42L52_PWRCTL3_HPA_ON_HIGH		0x01
+#define CS42L52_PWRCTL3_HPA_ALWAYS_ON		0x02
+#define CS42L52_PWRCTL3_HPA_ALWAYS_OFF		0x03
+#define CS42L52_PWRCTL3_SPKB_PDN_SHIFT		2
+#define CS42L52_PWRCTL3_SPKB_ON_LOW		0x00
+#define CS42L52_PWRCTL3_SPKB_ON_HIGH		0x01
+#define CS42L52_PWRCTL3_SPKB_ALWAYS_ON		0x02
+#define CS42L52_PWRCTL3_PDN_SPKB		(1 << 2)
+#define CS42L52_PWRCTL3_PDN_SPKA		(1 << 0)
+#define CS42L52_PWRCTL3_SPKA_PDN_SHIFT		0
+#define CS42L52_PWRCTL3_SPKA_ON_LOW		0x00
+#define CS42L52_PWRCTL3_SPKA_ON_HIGH		0x01
+#define CS42L52_PWRCTL3_SPKA_ALWAYS_ON		0x02
+
+#define CS42L52_DEFAULT_OUTPUT_STATE		0x05
+#define CS42L52_PWRCTL3_CONF_MASK		0x03
+
+#define CS42L52_CLK_CTL				0x05
+#define CLK_AUTODECT_ENABLE			(1 << 7)
+#define CLK_SPEED_SHIFT				5
+#define CLK_DS_MODE				0x00
+#define CLK_SS_MODE				0x01
+#define CLK_HS_MODE				0x02
+#define CLK_QS_MODE				0x03
+#define CLK_32K_SR_SHIFT			4
+#define CLK_32K					0x01
+#define CLK_NO_32K				0x00
+#define CLK_27M_MCLK_SHIFT			3
+#define CLK_27M_MCLK				0x01
+#define CLK_NO_27M				0x00
+#define CLK_RATIO_SHIFT				1
+#define CLK_R_128				0x00
+#define CLK_R_125				0x01
+#define CLK_R_132				0x02
+#define CLK_R_136				0x03
+
+#define CS42L52_IFACE_CTL1			0x06
+#define CS42L52_IFACE_CTL1_MASTER		(1 << 7)
+#define CS42L52_IFACE_CTL1_SLAVE		(0 << 7)
+#define CS42L52_IFACE_CTL1_INV_SCLK		(1 << 6)
+#define CS42L52_IFACE_CTL1_ADC_FMT_I2S		(1 << 5)
+#define CS42L52_IFACE_CTL1_ADC_FMT_LEFT_J	(0 << 5)
+#define CS42L52_IFACE_CTL1_DSP_MODE_EN		(1 << 4)
+#define CS42L52_IFACE_CTL1_DAC_FMT_LEFT_J	(0 << 2)
+#define CS42L52_IFACE_CTL1_DAC_FMT_I2S		(1 << 2)
+#define CS42L52_IFACE_CTL1_DAC_FMT_RIGHT_J	(2 << 2)
+#define CS42L52_IFACE_CTL1_WL_32BIT		(0x00)
+#define CS42L52_IFACE_CTL1_WL_24BIT		(0x01)
+#define CS42L52_IFACE_CTL1_WL_20BIT		(0x02)
+#define CS42L52_IFACE_CTL1_WL_16BIT		(0x03)
+#define CS42L52_IFACE_CTL1_WL_MASK		0xFFFF
+
+#define CS42L52_IFACE_CTL2			0x07
+#define CS42L52_IFACE_CTL2_SC_MC_EQ		(1 << 6)
+#define CS42L52_IFACE_CTL2_LOOPBACK		(1 << 5)
+#define CS42L52_IFACE_CTL2_S_MODE_OUTPUT_EN	(0 << 4)
+#define CS42L52_IFACE_CTL2_S_MODE_OUTPUT_HIZ	(1 << 4)
+#define CS42L52_IFACE_CTL2_HP_SW_INV		(1 << 3)
+#define CS42L52_IFACE_CTL2_BIAS_LVL		0x07
+
+#define CS42L52_ADC_PGA_A			0x08
+#define CS42L52_ADC_PGA_B			0x09
+#define CS42L52_ADC_SEL_SHIFT			5
+#define CS42L52_ADC_SEL_AIN1			0x00
+#define CS42L52_ADC_SEL_AIN2			0x01
+#define CS42L52_ADC_SEL_AIN3			0x02
+#define CS42L52_ADC_SEL_AIN4			0x03
+#define CS42L52_ADC_SEL_PGA			0x04
+
+#define CS42L52_ANALOG_HPF_CTL			0x0A
+#define CS42L52_HPF_CTL_ANLGSFTB		(1 << 3)
+#define CS42L52_HPF_CTL_ANLGSFTA                (1 << 0)
+
+#define CS42L52_ADC_HPF_FREQ			0x0B
+#define CS42L52_ADC_MISC_CTL			0x0C
+#define CS42L52_ADC_MISC_CTL_SOURCE_DSP		(1 << 6)
+
+#define CS42L52_PB_CTL1				0x0D
+#define CS42L52_PB_CTL1_HP_GAIN_SHIFT		5
+#define CS42L52_PB_CTL1_HP_GAIN_03959		0x00
+#define CS42L52_PB_CTL1_HP_GAIN_04571		0x01
+#define CS42L52_PB_CTL1_HP_GAIN_05111		0x02
+#define CS42L52_PB_CTL1_HP_GAIN_06047		0x03
+#define CS42L52_PB_CTL1_HP_GAIN_07099		0x04
+#define CS42L52_PB_CTL1_HP_GAIN_08399		0x05
+#define CS42L52_PB_CTL1_HP_GAIN_10000		0x06
+#define CS42L52_PB_CTL1_HP_GAIN_11430		0x07
+#define CS42L52_PB_CTL1_INV_PCMB		(1 << 3)
+#define CS42L52_PB_CTL1_INV_PCMA		(1 << 2)
+#define CS42L52_PB_CTL1_MSTB_MUTE		(1 << 1)
+#define CS42L52_PB_CTL1_MSTA_MUTE		(1 << 0)
+#define CS42L52_PB_CTL1_MUTE_MASK		0xFFFD
+#define CS42L52_PB_CTL1_MUTE			3
+#define CS42L52_PB_CTL1_UNMUTE			0
+
+#define CS42L52_MISC_CTL			0x0E
+#define CS42L52_MISC_CTL_DEEMPH			(1 << 2)
+#define CS42L52_MISC_CTL_DIGSFT			(1 << 1)
+#define CS42L52_MISC_CTL_DIGZC			(1 << 0)
+
+#define CS42L52_PB_CTL2				0x0F
+#define CS42L52_PB_CTL2_HPB_MUTE		(1 << 7)
+#define CS42L52_PB_CTL2_HPA_MUTE		(1 << 6)
+#define CS42L52_PB_CTL2_SPKB_MUTE		(1 << 5)
+#define CS42L52_PB_CTL2_SPKA_MUTE		(1 << 4)
+#define CS42L52_PB_CTL2_SPK_SWAP		(1 << 2)
+#define CS42L52_PB_CTL2_SPK_MONO		(1 << 1)
+#define CS42L52_PB_CTL2_SPK_MUTE50		(1 << 0)
+
+#define	CS42L52_MICA_CTL			0x10
+#define CS42L52_MICB_CTL			0x11
+#define	CS42L52_MIC_CTL_MIC_SEL_MASK		0xBF
+#define	CS42L52_MIC_CTL_MIC_SEL_SHIFT		6
+#define CS42L52_MIC_CTL_TYPE_MASK		0xDF
+#define CS42L52_MIC_CTL_TYPE_SHIFT		5
+
+
+#define CS42L52_PGAA_CTL			0x12
+#define CS42L52_PGAB_CTL			0x13
+#define CS42L52_PGAX_CTL_VOL_12DB		24
+#define CS42L52_PGAX_CTL_VOL_6DB		12 /*step size 0.5db*/
+
+#define CS42L52_PASSTHRUA_VOL			0x14
+#define CS42L52_PASSTHRUB_VOL			0x15
+
+#define CS42L52_ADCA_VOL			0x16
+#define CS42L52_ADCB_VOL			0x17
+#define CS42L52_ADCX_VOL_24DB			24 /*step size 1db*/
+#define CS42L52_ADCX_VOL_12DB			12
+#define CS42L52_ADCX_VOL_6DB			6
+
+#define CS42L52_ADCA_MIXER_VOL			0x18
+#define CS42L52_ADCB_MIXER_VOL			0x19
+#define CS42L52_ADC_MIXER_VOL_12DB		0x18
+
+#define CS42L52_PCMA_MIXER_VOL			0x1A
+#define CS42L52_PCMB_MIXER_VOL			0x1B
+
+#define CS42L52_BEEP_FREQ			0x1C
+#define CS42L52_BEEP_VOL			0x1D
+#define CS42L52_BEEP_TONE_CTL			0x1E
+#define CS42L52_BEEP_RATE_SHIFT			4
+#define CS42L52_BEEP_RATE_MASK			0x0F
+
+#define CS42L52_TONE_CTL			0x1F
+#define CS42L52_BEEP_EN_MASK			0x3F
+
+#define CS42L52_MASTERA_VOL			0x20
+#define CS42L52_MASTERB_VOL			0x21
+
+#define CS42L52_HPA_VOL				0x22
+#define CS42L52_HPB_VOL				0x23
+#define CS42L52_DEFAULT_HP_VOL			0xF0
+
+#define CS42L52_SPKA_VOL			0x24
+#define CS42L52_SPKB_VOL			0x25
+#define CS42L52_DEFAULT_SPK_VOL			0xF0
+
+#define CS42L52_ADC_PCM_MIXER			0x26
+
+#define CS42L52_LIMITER_CTL1			0x27
+#define CS42L52_LIMITER_CTL2			0x28
+#define CS42L52_LIMITER_AT_RATE			0x29
+
+#define CS42L52_ALC_CTL				0x2A
+#define CS42L52_ALC_CTL_ALCB_ENABLE_SHIFT	7
+#define CS42L52_ALC_CTL_ALCA_ENABLE_SHIFT	6
+#define CS42L52_ALC_CTL_FASTEST_ATTACK		0
+
+#define CS42L52_ALC_RATE			0x2B
+#define CS42L52_ALC_SLOWEST_RELEASE		0x3F
+
+#define CS42L52_ALC_THRESHOLD			0x2C
+#define CS42L52_ALC_MAX_RATE_SHIFT		5
+#define CS42L52_ALC_MIN_RATE_SHIFT		2
+#define CS42L52_ALC_RATE_0DB			0
+#define CS42L52_ALC_RATE_3DB			1
+#define CS42L52_ALC_RATE_6DB			2
+
+#define CS42L52_NOISE_GATE_CTL			0x2D
+#define CS42L52_NG_ENABLE_SHIFT			6
+#define CS42L52_NG_THRESHOLD_SHIFT		2
+#define CS42L52_NG_MIN_70DB			2
+#define CS42L52_NG_DELAY_SHIFT			0
+#define CS42L52_NG_DELAY_100MS			1
+
+#define CS42L52_CLK_STATUS			0x2E
+#define CS42L52_BATT_COMPEN			0x2F
+
+#define CS42L52_BATT_LEVEL			0x30
+#define CS42L52_SPK_STATUS			0x31
+#define CS42L52_SPK_STATUS_PIN_SHIFT		3
+#define CS42L52_SPK_STATUS_PIN_HIGH		1
+
+#define CS42L52_TEM_CTL				0x32
+#define CS42L52_TEM_CTL_SET			0x80
+#define CS42L52_THE_FOLDBACK			0x33
+#define CS42L52_CHARGE_PUMP			0x34
+#define CS42L52_CHARGE_PUMP_MASK		0xF0
+#define CS42L52_CHARGE_PUMP_SHIFT		4
+#define CS42L52_FIX_BITS1			0x3E
+#define CS42L52_FIX_BITS2			0x47
+
+#define CS42L52_MAX_REGISTER			0x34
+
+#endif
diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c
index 3686417f5ea5..e0d45fdaa750 100644
--- a/sound/soc/codecs/cs42l73.c
+++ b/sound/soc/codecs/cs42l73.c
@@ -43,9 +43,6 @@ struct  cs42l73_private {
 };
 
 static const 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 */
@@ -402,37 +399,37 @@ static const struct snd_kcontrol_new ear_amp_ctl =
 
 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),
+			CS42L73_HPAAVOL, CS42L73_HPBAVOL, 0,
+			0x41, 0x4B, hpaloa_tlv),
 
 	SOC_DOUBLE_R_SX_TLV("LineOut Analog Playback Volume", CS42L73_LOAAVOL,
-			CS42L73_LOBAVOL, 7, 0xffffffC1, 0x0C, hpaloa_tlv),
+			CS42L73_LOBAVOL, 0, 0x41, 0x4B, hpaloa_tlv),
 
 	SOC_DOUBLE_R_SX_TLV("Input PGA Analog Volume", CS42L73_MICAPREPGAAVOL,
-			CS42L73_MICBPREPGABVOL, 5, 0xffffff35,
-			0x34, micpga_tlv),
+			CS42L73_MICBPREPGABVOL, 5, 0x34,
+			0x24, 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),
+			CS42L73_IPBDVOL, 0, 0xA0, 0x6C, ipd_tlv),
 
 	SOC_DOUBLE_R_SX_TLV("HL Digital Playback Volume",
-			CS42L73_HLADVOL, CS42L73_HLBDVOL, 7, 0xffffffE5,
-			0xE4, hl_tlv),
+			CS42L73_HLADVOL, CS42L73_HLBDVOL,
+			0, 0x34, 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),
+		       CS42L73_ADCIPC, 6, 0x01, 1, adc_boost_tlv),
 
-	SOC_SINGLE_TLV("Speakerphone Digital Playback Volume",
-			CS42L73_SPKDVOL, 0, 0xE4, 1, hl_tlv),
+	SOC_SINGLE_SX_TLV("Speakerphone Digital Volume",
+			    CS42L73_SPKDVOL, 0, 0x34, 0xE4, hl_tlv),
 
-	SOC_SINGLE_TLV("Ear Speaker Digital Playback Volume",
-			CS42L73_ESLDVOL, 0, 0xE4, 1, hl_tlv),
+	SOC_SINGLE_SX_TLV("Ear Speaker Digital Volume",
+			    CS42L73_ESLDVOL, 0, 0x34, 0xE4, hl_tlv),
 
 	SOC_DOUBLE_R("Headphone Analog Playback Switch", CS42L73_HPAAVOL,
 			CS42L73_HPBAVOL, 7, 1, 1),
@@ -599,17 +596,17 @@ static const struct snd_soc_dapm_widget cs42l73_dapm_widgets[] = {
 	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,
+	SND_SOC_DAPM_AIF_OUT("XSPOUTL", NULL,  0,
 			CS42L73_PWRCTL2, 1, 1),
-	SND_SOC_DAPM_AIF_OUT("XSPOUTR", "XSP Capture",  0,
+	SND_SOC_DAPM_AIF_OUT("XSPOUTR", NULL,  0,
 			CS42L73_PWRCTL2, 1, 1),
-	SND_SOC_DAPM_AIF_OUT("ASPOUTL", "ASP Capture",  0,
+	SND_SOC_DAPM_AIF_OUT("ASPOUTL", NULL,  0,
 			CS42L73_PWRCTL2, 3, 1),
-	SND_SOC_DAPM_AIF_OUT("ASPOUTR", "ASP Capture",  0,
+	SND_SOC_DAPM_AIF_OUT("ASPOUTR", NULL,  0,
 			CS42L73_PWRCTL2, 3, 1),
-	SND_SOC_DAPM_AIF_OUT("VSPOUTL", "VSP Capture",  0,
+	SND_SOC_DAPM_AIF_OUT("VSPOUTL", NULL,  0,
 			CS42L73_PWRCTL2, 4, 1),
-	SND_SOC_DAPM_AIF_OUT("VSPOUTR", "VSP Capture",  0,
+	SND_SOC_DAPM_AIF_OUT("VSPOUTR", NULL,  0,
 			CS42L73_PWRCTL2, 4, 1),
 
 	SND_SOC_DAPM_PGA("PGA Left", SND_SOC_NOPM, 0, 0, NULL, 0),
@@ -638,21 +635,21 @@ static const struct snd_soc_dapm_widget cs42l73_dapm_widgets[] = {
 	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,
+	SND_SOC_DAPM_AIF_IN("XSPINL", NULL, 0,
 				CS42L73_PWRCTL2, 0, 1),
-	SND_SOC_DAPM_AIF_IN("XSPINR", "XSP Playback", 0,
+	SND_SOC_DAPM_AIF_IN("XSPINR", NULL, 0,
 				CS42L73_PWRCTL2, 0, 1),
-	SND_SOC_DAPM_AIF_IN("XSPINM", "XSP Playback", 0,
+	SND_SOC_DAPM_AIF_IN("XSPINM", NULL, 0,
 				CS42L73_PWRCTL2, 0, 1),
 
-	SND_SOC_DAPM_AIF_IN("ASPINL", "ASP Playback", 0,
+	SND_SOC_DAPM_AIF_IN("ASPINL", NULL, 0,
 				CS42L73_PWRCTL2, 2, 1),
-	SND_SOC_DAPM_AIF_IN("ASPINR", "ASP Playback", 0,
+	SND_SOC_DAPM_AIF_IN("ASPINR", NULL, 0,
 				CS42L73_PWRCTL2, 2, 1),
-	SND_SOC_DAPM_AIF_IN("ASPINM", "ASP Playback", 0,
+	SND_SOC_DAPM_AIF_IN("ASPINM", NULL, 0,
 				CS42L73_PWRCTL2, 2, 1),
 
-	SND_SOC_DAPM_AIF_IN("VSPIN", "VSP Playback", 0,
+	SND_SOC_DAPM_AIF_IN("VSPIN", NULL, 0,
 				CS42L73_PWRCTL2, 4, 1),
 
 	SND_SOC_DAPM_MIXER("HL Left Mixer", SND_SOC_NOPM, 0, 0, NULL, 0),
@@ -776,6 +773,14 @@ static const struct snd_soc_dapm_route cs42l73_audio_map[] = {
 	{"HL Left Mixer", NULL, "VSPIN"},
 	{"HL Right Mixer", NULL, "VSPIN"},
 
+	{"ASPINL", NULL, "ASP Playback"},
+	{"ASPINM", NULL, "ASP Playback"},
+	{"ASPINR", NULL, "ASP Playback"},
+	{"XSPINL", NULL, "XSP Playback"},
+	{"XSPINM", NULL, "XSP Playback"},
+	{"XSPINR", NULL, "XSP Playback"},
+	{"VSPIN", NULL, "VSP Playback"},
+
 	/* Capture Paths */
 	{"MIC1", NULL, "MIC1 Bias"},
 	{"PGA Left Mux", "Mic 1", "MIC1"},
@@ -822,6 +827,13 @@ static const struct snd_soc_dapm_route cs42l73_audio_map[] = {
 
 	{"VSPOUTL", NULL, "VSPL Output Mixer"},
 	{"VSPOUTR", NULL, "VSPR Output Mixer"},
+
+	{"ASP Capture", NULL, "ASPOUTL"},
+	{"ASP Capture", NULL, "ASPOUTR"},
+	{"XSP Capture", NULL, "XSPOUTL"},
+	{"XSP Capture", NULL, "XSPOUTR"},
+	{"VSP Capture", NULL, "VSPOUTL"},
+	{"VSP Capture", NULL, "VSPOUTR"},
 };
 
 struct cs42l73_mclk_div {
@@ -1091,8 +1103,7 @@ 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 snd_soc_codec *codec = dai->codec;
 	struct cs42l73_private *priv = snd_soc_codec_get_drvdata(codec);
 	int id = dai->id;
 	int mclk_coeff;
@@ -1429,25 +1440,7 @@ static struct i2c_driver cs42l73_i2c_driver = {
 
 };
 
-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_i2c_driver(cs42l73_i2c_driver);
 
 MODULE_DESCRIPTION("ASoC CS42L73 driver");
 MODULE_AUTHOR("Georgi Vlaev, Nucleus Systems Ltd, <joe@nucleusys.com>");
diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c
index 7843711729bc..af5db7080519 100644
--- a/sound/soc/codecs/da7210.c
+++ b/sound/soc/codecs/da7210.c
@@ -17,6 +17,7 @@
 
 #include <linux/delay.h>
 #include <linux/i2c.h>
+#include <linux/spi/spi.h>
 #include <linux/regmap.h>
 #include <linux/slab.h>
 #include <linux/module.h>
@@ -27,6 +28,7 @@
 #include <sound/tlv.h>
 
 /* DA7210 register space */
+#define DA7210_PAGE_CONTROL		0x00
 #define DA7210_CONTROL			0x01
 #define DA7210_STATUS			0x02
 #define DA7210_STARTUP1			0x03
@@ -146,6 +148,7 @@
 #define DA7210_DAI_EN			(1 << 7)
 
 /*PLL_DIV3 bit fields */
+#define DA7210_PLL_DIV_L_MASK		(0xF << 0)
 #define DA7210_MCLK_RANGE_10_20_MHZ	(1 << 4)
 #define DA7210_PLL_BYP			(1 << 6)
 
@@ -162,12 +165,16 @@
 #define DA7210_PLL_FS_48000		(0xB << 0)
 #define DA7210_PLL_FS_88200		(0xE << 0)
 #define DA7210_PLL_FS_96000		(0xF << 0)
+#define DA7210_MCLK_DET_EN		(0x1 << 5)
+#define DA7210_MCLK_SRM_EN		(0x1 << 6)
 #define DA7210_PLL_EN			(0x1 << 7)
 
 /* SOFTMUTE bit fields */
 #define DA7210_RAMP_EN			(1 << 6)
 
 /* CONTROL bit fields */
+#define DA7210_REG_EN			(1 << 0)
+#define DA7210_BIAS_EN			(1 << 2)
 #define DA7210_NOISE_SUP_EN		(1 << 3)
 
 /* IN_GAIN bit fields */
@@ -206,6 +213,47 @@
 #define DA7210_OUT2_OUTMIX_L		(1 << 6)
 #define DA7210_OUT2_EN			(1 << 7)
 
+struct pll_div {
+	int fref;
+	int fout;
+	u8 div1;
+	u8 div2;
+	u8 div3;
+	u8 mode;	/* 0 = slave, 1 = master */
+};
+
+/* PLL dividers table */
+static const struct pll_div da7210_pll_div[] = {
+	/* for MASTER mode, fs = 44.1Khz */
+	{ 12000000, 2822400, 0xE8, 0x6C, 0x2, 1},	/* MCLK=12Mhz */
+	{ 13000000, 2822400, 0xDF, 0x28, 0xC, 1},	/* MCLK=13Mhz */
+	{ 13500000, 2822400, 0xDB, 0x0A, 0xD, 1},	/* MCLK=13.5Mhz */
+	{ 14400000, 2822400, 0xD4, 0x5A, 0x2, 1},	/* MCLK=14.4Mhz */
+	{ 19200000, 2822400, 0xBB, 0x43, 0x9, 1},	/* MCLK=19.2Mhz */
+	{ 19680000, 2822400, 0xB9, 0x6D, 0xA, 1},	/* MCLK=19.68Mhz */
+	{ 19800000, 2822400, 0xB8, 0xFB, 0xB, 1},	/* MCLK=19.8Mhz */
+	/* for MASTER mode, fs = 48Khz */
+	{ 12000000, 3072000, 0xF3, 0x12, 0x7, 1},	/* MCLK=12Mhz */
+	{ 13000000, 3072000, 0xE8, 0xFD, 0x5, 1},	/* MCLK=13Mhz */
+	{ 13500000, 3072000, 0xE4, 0x82, 0x3, 1},	/* MCLK=13.5Mhz */
+	{ 14400000, 3072000, 0xDD, 0x3A, 0x0, 1},	/* MCLK=14.4Mhz */
+	{ 19200000, 3072000, 0xC1, 0xEB, 0x8, 1},	/* MCLK=19.2Mhz */
+	{ 19680000, 3072000, 0xBF, 0xEC, 0x0, 1},	/* MCLK=19.68Mhz */
+	{ 19800000, 3072000, 0xBF, 0x70, 0x0, 1},	/* MCLK=19.8Mhz */
+	/* for SLAVE mode with SRM */
+	{ 12000000, 2822400, 0xED, 0xBF, 0x5, 0},	/* MCLK=12Mhz */
+	{ 13000000, 2822400, 0xE4, 0x13, 0x0, 0},	/* MCLK=13Mhz */
+	{ 13500000, 2822400, 0xDF, 0xC6, 0x8, 0},	/* MCLK=13.5Mhz */
+	{ 14400000, 2822400, 0xD8, 0xCA, 0x1, 0},	/* MCLK=14.4Mhz */
+	{ 19200000, 2822400, 0xBE, 0x97, 0x9, 0},	/* MCLK=19.2Mhz */
+	{ 19680000, 2822400, 0xBC, 0xAC, 0xD, 0},	/* MCLK=19.68Mhz */
+	{ 19800000, 2822400, 0xBC, 0x35, 0xE, 0},	/* MCLK=19.8Mhz  */
+};
+
+enum clk_src {
+	DA7210_CLKSRC_MCLK
+};
+
 #define DA7210_VERSION "0.0.1"
 
 /*
@@ -628,9 +676,12 @@ static const struct snd_soc_dapm_route da7210_audio_map[] = {
 /* Codec private data */
 struct da7210_priv {
 	struct regmap *regmap;
+	unsigned int mclk_rate;
+	int master;
 };
 
 static struct reg_default da7210_reg_defaults[] = {
+	{ 0x00, 0x00 },
 	{ 0x01, 0x11 },
 	{ 0x03, 0x00 },
 	{ 0x04, 0x00 },
@@ -713,10 +764,10 @@ static int da7210_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 snd_soc_codec *codec = dai->codec;
+	struct da7210_priv *da7210 = snd_soc_codec_get_drvdata(codec);
 	u32 dai_cfg1;
-	u32 fs, bypass;
+	u32 fs, sysclk;
 
 	/* set DAI source to Left and Right ADC */
 	snd_soc_write(codec, DA7210_DAI_SRC_SEL,
@@ -749,43 +800,43 @@ static int da7210_hw_params(struct snd_pcm_substream *substream,
 	switch (params_rate(params)) {
 	case 8000:
 		fs		= DA7210_PLL_FS_8000;
-		bypass		= DA7210_PLL_BYP;
+		sysclk		= 3072000;
 		break;
 	case 11025:
 		fs		= DA7210_PLL_FS_11025;
-		bypass		= 0;
+		sysclk		= 2822400;
 		break;
 	case 12000:
 		fs		= DA7210_PLL_FS_12000;
-		bypass		= DA7210_PLL_BYP;
+		sysclk		= 3072000;
 		break;
 	case 16000:
 		fs		= DA7210_PLL_FS_16000;
-		bypass		= DA7210_PLL_BYP;
+		sysclk		= 3072000;
 		break;
 	case 22050:
 		fs		= DA7210_PLL_FS_22050;
-		bypass		= 0;
+		sysclk		= 2822400;
 		break;
 	case 32000:
 		fs		= DA7210_PLL_FS_32000;
-		bypass		= DA7210_PLL_BYP;
+		sysclk		= 3072000;
 		break;
 	case 44100:
 		fs		= DA7210_PLL_FS_44100;
-		bypass		= 0;
+		sysclk		= 2822400;
 		break;
 	case 48000:
 		fs		= DA7210_PLL_FS_48000;
-		bypass		= DA7210_PLL_BYP;
+		sysclk		= 3072000;
 		break;
 	case 88200:
 		fs		= DA7210_PLL_FS_88200;
-		bypass		= 0;
+		sysclk		= 2822400;
 		break;
 	case 96000:
 		fs		= DA7210_PLL_FS_96000;
-		bypass		= DA7210_PLL_BYP;
+		sysclk		= 3072000;
 		break;
 	default:
 		return -EINVAL;
@@ -795,8 +846,26 @@ static int da7210_hw_params(struct snd_pcm_substream *substream,
 	snd_soc_update_bits(codec, DA7210_STARTUP1, DA7210_SC_MST_EN, 0);
 
 	snd_soc_update_bits(codec, DA7210_PLL, DA7210_PLL_FS_MASK, fs);
-	snd_soc_update_bits(codec, DA7210_PLL_DIV3, DA7210_PLL_BYP, bypass);
 
+	if (da7210->mclk_rate && (da7210->mclk_rate != sysclk)) {
+		/* PLL mode, disable PLL bypass */
+		snd_soc_update_bits(codec, DA7210_PLL_DIV3, DA7210_PLL_BYP, 0);
+
+		if (!da7210->master) {
+			/* PLL slave mode, also enable SRM */
+			snd_soc_update_bits(codec, DA7210_PLL,
+						   (DA7210_MCLK_SRM_EN |
+						    DA7210_MCLK_DET_EN),
+						   (DA7210_MCLK_SRM_EN |
+						    DA7210_MCLK_DET_EN));
+		}
+	} else {
+		/* PLL bypass mode, enable PLL bypass and Auto Detection */
+		snd_soc_update_bits(codec, DA7210_PLL, DA7210_MCLK_DET_EN,
+						       DA7210_MCLK_DET_EN);
+		snd_soc_update_bits(codec, DA7210_PLL_DIV3, DA7210_PLL_BYP,
+							    DA7210_PLL_BYP);
+	}
 	/* Enable active mode */
 	snd_soc_update_bits(codec, DA7210_STARTUP1,
 			    DA7210_SC_MST_EN, DA7210_SC_MST_EN);
@@ -810,17 +879,24 @@ static int da7210_hw_params(struct snd_pcm_substream *substream,
 static int da7210_set_dai_fmt(struct snd_soc_dai *codec_dai, u32 fmt)
 {
 	struct snd_soc_codec *codec = codec_dai->codec;
+	struct da7210_priv *da7210 = snd_soc_codec_get_drvdata(codec);
 	u32 dai_cfg1;
 	u32 dai_cfg3;
 
 	dai_cfg1 = 0x7f & snd_soc_read(codec, DA7210_DAI_CFG1);
 	dai_cfg3 = 0xfc & snd_soc_read(codec, DA7210_DAI_CFG3);
 
+	if ((snd_soc_read(codec, DA7210_PLL) & DA7210_PLL_EN) &&
+		(!(snd_soc_read(codec, DA7210_PLL_DIV3) & DA7210_PLL_BYP)))
+		return -EINVAL;
+
 	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
 	case SND_SOC_DAIFMT_CBM_CFM:
+		da7210->master = 1;
 		dai_cfg1 |= DA7210_DAI_MODE_MASTER;
 		break;
 	case SND_SOC_DAIFMT_CBS_CFS:
+		da7210->master = 0;
 		dai_cfg1 |= DA7210_DAI_MODE_SLAVE;
 		break;
 	default:
@@ -872,10 +948,101 @@ static int da7210_mute(struct snd_soc_dai *dai, int mute)
 #define DA7210_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
 			SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE)
 
+static int da7210_set_dai_sysclk(struct snd_soc_dai *codec_dai,
+				 int clk_id, unsigned int freq, int dir)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	struct da7210_priv *da7210 = snd_soc_codec_get_drvdata(codec);
+
+	switch (clk_id) {
+	case DA7210_CLKSRC_MCLK:
+		switch (freq) {
+		case 12000000:
+		case 13000000:
+		case 13500000:
+		case 14400000:
+		case 19200000:
+		case 19680000:
+		case 19800000:
+			da7210->mclk_rate = freq;
+			return 0;
+		default:
+			dev_err(codec_dai->dev, "Unsupported MCLK value %d\n",
+				freq);
+			return -EINVAL;
+		}
+		break;
+	default:
+		dev_err(codec_dai->dev, "Unknown clock source %d\n", clk_id);
+		return -EINVAL;
+	}
+}
+
+/**
+ * da7210_set_dai_pll	:Configure the codec PLL
+ * @param codec_dai	: pointer to codec DAI
+ * @param pll_id	: da7210 has only one pll, so pll_id is always zero
+ * @param fref		: MCLK frequency, should be < 20MHz
+ * @param fout		: FsDM value, Refer page 44 & 45 of datasheet
+ * @return int		: Zero for success, negative error code for error
+ *
+ * Note: Supported PLL input frequencies are 12MHz, 13MHz, 13.5MHz, 14.4MHz,
+ *       19.2MHz, 19.6MHz and 19.8MHz
+ */
+static int da7210_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id,
+			      int source, unsigned int fref, unsigned int fout)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	struct da7210_priv *da7210 = snd_soc_codec_get_drvdata(codec);
+
+	u8 pll_div1, pll_div2, pll_div3, cnt;
+
+	/* In slave mode, there is only one set of divisors */
+	if (!da7210->master)
+		fout = 2822400;
+
+	/* Search pll div array for correct divisors */
+	for (cnt = 0; cnt < ARRAY_SIZE(da7210_pll_div); cnt++) {
+		/* check fref, mode  and fout */
+		if ((fref == da7210_pll_div[cnt].fref) &&
+		    (da7210->master ==  da7210_pll_div[cnt].mode) &&
+		    (fout == da7210_pll_div[cnt].fout)) {
+			/* all match, pick up divisors */
+			pll_div1 = da7210_pll_div[cnt].div1;
+			pll_div2 = da7210_pll_div[cnt].div2;
+			pll_div3 = da7210_pll_div[cnt].div3;
+			break;
+		}
+	}
+	if (cnt >= ARRAY_SIZE(da7210_pll_div))
+		goto err;
+
+	/* Disable active mode */
+	snd_soc_update_bits(codec, DA7210_STARTUP1, DA7210_SC_MST_EN, 0);
+	/* Write PLL dividers */
+	snd_soc_write(codec, DA7210_PLL_DIV1, pll_div1);
+	snd_soc_write(codec, DA7210_PLL_DIV2, pll_div2);
+	snd_soc_update_bits(codec, DA7210_PLL_DIV3,
+				   DA7210_PLL_DIV_L_MASK, pll_div3);
+
+	/* Enable PLL */
+	snd_soc_update_bits(codec, DA7210_PLL, DA7210_PLL_EN, DA7210_PLL_EN);
+
+	/* Enable active mode */
+	snd_soc_update_bits(codec, DA7210_STARTUP1, DA7210_SC_MST_EN,
+						    DA7210_SC_MST_EN);
+	return 0;
+err:
+	dev_err(codec_dai->dev, "Unsupported PLL input frequency %d\n", fref);
+	return -EINVAL;
+}
+
 /* DAI operations */
 static const struct snd_soc_dai_ops da7210_dai_ops = {
 	.hw_params	= da7210_hw_params,
 	.set_fmt	= da7210_set_dai_fmt,
+	.set_sysclk	= da7210_set_dai_sysclk,
+	.set_pll	= da7210_set_dai_pll,
 	.digital_mute	= da7210_mute,
 };
 
@@ -915,24 +1082,11 @@ static int da7210_probe(struct snd_soc_codec *codec)
 		return ret;
 	}
 
-	/* FIXME
-	 *
-	 * This driver use fixed value here
-	 * And below settings expects MCLK = 12.288MHz
-	 *
-	 * When you select different MCLK, please check...
-	 *      DA7210_PLL_DIV1 val
-	 *      DA7210_PLL_DIV2 val
-	 *      DA7210_PLL_DIV3 val
-	 *      DA7210_PLL_DIV3 :: DA7210_MCLK_RANGExxx
-	 */
+	da7210->mclk_rate       = 0;    /* This will be set from set_sysclk() */
+	da7210->master          = 0;    /* This will be set from set_fmt() */
 
-	/*
-	 * make sure that DA7210 use bypass mode before start up
-	 */
-	snd_soc_write(codec, DA7210_STARTUP1, 0);
-	snd_soc_write(codec, DA7210_PLL_DIV3,
-		     DA7210_MCLK_RANGE_10_20_MHZ | DA7210_PLL_BYP);
+	/* Enable internal regulator & bias current */
+	snd_soc_write(codec, DA7210_CONTROL, DA7210_REG_EN | DA7210_BIAS_EN);
 
 	/*
 	 * ADC settings
@@ -1007,34 +1161,13 @@ static int da7210_probe(struct snd_soc_codec *codec)
 	/* Enable Aux2 */
 	snd_soc_write(codec, DA7210_AUX2, DA7210_AUX2_EN);
 
+	/* Set PLL Master clock range 10-20 MHz, enable PLL bypass */
+	snd_soc_write(codec, DA7210_PLL_DIV3, DA7210_MCLK_RANGE_10_20_MHZ |
+					      DA7210_PLL_BYP);
+
 	/* Diable PLL and bypass it */
 	snd_soc_write(codec, DA7210_PLL, DA7210_PLL_FS_48000);
 
-	/*
-	 * If 48kHz sound came, it use bypass mode,
-	 * and when it is 44.1kHz, it use PLL.
-	 *
-	 * This time, this driver sets PLL always ON
-	 * and controls bypass/PLL mode by switching
-	 * DA7210_PLL_DIV3 :: DA7210_PLL_BYP bit.
-	 *   see da7210_hw_params
-	 */
-	snd_soc_write(codec, DA7210_PLL_DIV1, 0xE5); /* MCLK = 12.288MHz */
-	snd_soc_write(codec, DA7210_PLL_DIV2, 0x99);
-	snd_soc_write(codec, DA7210_PLL_DIV3, 0x0A |
-		     DA7210_MCLK_RANGE_10_20_MHZ | DA7210_PLL_BYP);
-	snd_soc_update_bits(codec, DA7210_PLL, DA7210_PLL_EN, DA7210_PLL_EN);
-
-	/* As suggested by Dialog */
-	/* unlock */
-	regmap_write(da7210->regmap, DA7210_A_HID_UNLOCK,	0x8B);
-	regmap_write(da7210->regmap, DA7210_A_TEST_UNLOCK,	0xB4);
-	regmap_write(da7210->regmap, DA7210_A_PLL1,		0x01);
-	regmap_write(da7210->regmap, DA7210_A_CP_MODE,		0x7C);
-	/* re-lock */
-	regmap_write(da7210->regmap, DA7210_A_HID_UNLOCK,	0x00);
-	regmap_write(da7210->regmap, DA7210_A_TEST_UNLOCK,	0x00);
-
 	/* Activate all enabled subsystem */
 	snd_soc_write(codec, DA7210_STARTUP1, DA7210_SC_MST_EN);
 
@@ -1055,7 +1188,26 @@ static struct snd_soc_codec_driver soc_codec_dev_da7210 = {
 	.num_dapm_routes	= ARRAY_SIZE(da7210_audio_map),
 };
 
-static struct regmap_config da7210_regmap = {
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+
+static struct reg_default da7210_regmap_i2c_patch[] = {
+
+	/* System controller master disable */
+	{ DA7210_STARTUP1, 0x00 },
+	/* Set PLL Master clock range 10-20 MHz */
+	{ DA7210_PLL_DIV3, DA7210_MCLK_RANGE_10_20_MHZ },
+
+	/* to unlock */
+	{ DA7210_A_HID_UNLOCK, 0x8B},
+	{ DA7210_A_TEST_UNLOCK, 0xB4},
+	{ DA7210_A_PLL1, 0x01},
+	{ DA7210_A_CP_MODE, 0x7C},
+	/* to re-lock */
+	{ DA7210_A_HID_UNLOCK, 0x00},
+	{ DA7210_A_TEST_UNLOCK, 0x00},
+};
+
+static const struct regmap_config da7210_regmap_config_i2c = {
 	.reg_bits = 8,
 	.val_bits = 8,
 
@@ -1066,7 +1218,6 @@ static struct regmap_config da7210_regmap = {
 	.cache_type = REGCACHE_RBTREE,
 };
 
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
 static int __devinit da7210_i2c_probe(struct i2c_client *i2c,
 			   	      const struct i2c_device_id *id)
 {
@@ -1080,13 +1231,18 @@ static int __devinit da7210_i2c_probe(struct i2c_client *i2c,
 
 	i2c_set_clientdata(i2c, da7210);
 
-	da7210->regmap = regmap_init_i2c(i2c, &da7210_regmap);
+	da7210->regmap = regmap_init_i2c(i2c, &da7210_regmap_config_i2c);
 	if (IS_ERR(da7210->regmap)) {
 		ret = PTR_ERR(da7210->regmap);
 		dev_err(&i2c->dev, "regmap_init() failed: %d\n", ret);
 		return ret;
 	}
 
+	ret = regmap_register_patch(da7210->regmap, da7210_regmap_i2c_patch,
+				    ARRAY_SIZE(da7210_regmap_i2c_patch));
+	if (ret != 0)
+		dev_warn(&i2c->dev, "Failed to apply regmap patch: %d\n", ret);
+
 	ret =  snd_soc_register_codec(&i2c->dev,
 			&soc_codec_dev_da7210, &da7210_dai, 1);
 	if (ret < 0) {
@@ -1119,7 +1275,7 @@ MODULE_DEVICE_TABLE(i2c, da7210_i2c_id);
 /* I2C codec control layer */
 static struct i2c_driver da7210_i2c_driver = {
 	.driver = {
-		.name = "da7210-codec",
+		.name = "da7210",
 		.owner = THIS_MODULE,
 	},
 	.probe		= da7210_i2c_probe,
@@ -1128,11 +1284,111 @@ static struct i2c_driver da7210_i2c_driver = {
 };
 #endif
 
+#if defined(CONFIG_SPI_MASTER)
+
+static struct reg_default da7210_regmap_spi_patch[] = {
+	/* Dummy read to give two pulses over nCS for SPI */
+	{ DA7210_AUX2, 0x00 },
+	{ DA7210_AUX2, 0x00 },
+
+	/* System controller master disable */
+	{ DA7210_STARTUP1, 0x00 },
+	/* Set PLL Master clock range 10-20 MHz */
+	{ DA7210_PLL_DIV3, DA7210_MCLK_RANGE_10_20_MHZ },
+
+	/* to set PAGE1 of SPI register space */
+	{ DA7210_PAGE_CONTROL, 0x80 },
+	/* to unlock */
+	{ DA7210_A_HID_UNLOCK, 0x8B},
+	{ DA7210_A_TEST_UNLOCK, 0xB4},
+	{ DA7210_A_PLL1, 0x01},
+	{ DA7210_A_CP_MODE, 0x7C},
+	/* to re-lock */
+	{ DA7210_A_HID_UNLOCK, 0x00},
+	{ DA7210_A_TEST_UNLOCK, 0x00},
+	/* to set back PAGE0 of SPI register space */
+	{ DA7210_PAGE_CONTROL, 0x00 },
+};
+
+static const struct regmap_config da7210_regmap_config_spi = {
+	.reg_bits = 8,
+	.val_bits = 8,
+	.read_flag_mask = 0x01,
+	.write_flag_mask = 0x00,
+
+	.reg_defaults = da7210_reg_defaults,
+	.num_reg_defaults = ARRAY_SIZE(da7210_reg_defaults),
+	.volatile_reg = da7210_volatile_register,
+	.readable_reg = da7210_readable_register,
+	.cache_type = REGCACHE_RBTREE,
+};
+
+static int __devinit da7210_spi_probe(struct spi_device *spi)
+{
+	struct da7210_priv *da7210;
+	int ret;
+
+	da7210 = devm_kzalloc(&spi->dev, sizeof(struct da7210_priv),
+			      GFP_KERNEL);
+	if (!da7210)
+		return -ENOMEM;
+
+	spi_set_drvdata(spi, da7210);
+	da7210->regmap = devm_regmap_init_spi(spi, &da7210_regmap_config_spi);
+	if (IS_ERR(da7210->regmap)) {
+		ret = PTR_ERR(da7210->regmap);
+		dev_err(&spi->dev, "Failed to register regmap: %d\n", ret);
+		return ret;
+	}
+
+	ret = regmap_register_patch(da7210->regmap, da7210_regmap_spi_patch,
+				    ARRAY_SIZE(da7210_regmap_spi_patch));
+	if (ret != 0)
+		dev_warn(&spi->dev, "Failed to apply regmap patch: %d\n", ret);
+
+	ret =  snd_soc_register_codec(&spi->dev,
+			&soc_codec_dev_da7210, &da7210_dai, 1);
+	if (ret < 0)
+		goto err_regmap;
+
+	return ret;
+
+err_regmap:
+	regmap_exit(da7210->regmap);
+
+	return ret;
+}
+
+static int __devexit da7210_spi_remove(struct spi_device *spi)
+{
+	struct da7210_priv *da7210 = spi_get_drvdata(spi);
+	snd_soc_unregister_codec(&spi->dev);
+	regmap_exit(da7210->regmap);
+	return 0;
+}
+
+static struct spi_driver da7210_spi_driver = {
+	.driver = {
+		.name = "da7210",
+		.owner = THIS_MODULE,
+	},
+	.probe = da7210_spi_probe,
+	.remove = __devexit_p(da7210_spi_remove)
+};
+#endif
+
 static int __init da7210_modinit(void)
 {
 	int ret = 0;
 #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
 	ret = i2c_add_driver(&da7210_i2c_driver);
+#endif
+#if defined(CONFIG_SPI_MASTER)
+	ret = spi_register_driver(&da7210_spi_driver);
+	if (ret) {
+		printk(KERN_ERR "Failed to register da7210 SPI driver: %d\n",
+		       ret);
+	}
 #endif
 	return ret;
 }
@@ -1143,6 +1399,9 @@ static void __exit da7210_exit(void)
 #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
 	i2c_del_driver(&da7210_i2c_driver);
 #endif
+#if defined(CONFIG_SPI_MASTER)
+	spi_unregister_driver(&da7210_spi_driver);
+#endif
 }
 module_exit(da7210_exit);
 
diff --git a/sound/soc/codecs/jz4740.c b/sound/soc/codecs/jz4740.c
index 4624e752a188..85d9cabe6d55 100644
--- a/sound/soc/codecs/jz4740.c
+++ b/sound/soc/codecs/jz4740.c
@@ -164,8 +164,7 @@ static int jz4740_codec_hw_params(struct snd_pcm_substream *substream,
 	struct snd_pcm_hw_params *params, struct snd_soc_dai *dai)
 {
 	uint32_t val;
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_codec *codec =rtd->codec;
+	struct snd_soc_codec *codec = dai->codec;
 
 	switch (params_rate(params)) {
 	case 8000:
diff --git a/sound/soc/codecs/lm49453.c b/sound/soc/codecs/lm49453.c
new file mode 100644
index 000000000000..802b9f176b16
--- /dev/null
+++ b/sound/soc/codecs/lm49453.c
@@ -0,0 +1,1550 @@
+/*
+ * lm49453.c  -  LM49453 ALSA Soc Audio driver
+ *
+ * Copyright (c) 2012 Texas Instruments, Inc
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; version 2 of the License.
+ *
+ * Initially based on sound/soc/codecs/wm8350.c
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/version.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/tlv.h>
+#include <sound/jack.h>
+#include <sound/initval.h>
+#include <asm/div64.h>
+#include "lm49453.h"
+
+static struct reg_default lm49453_reg_defs[] = {
+	{ 0, 0x00 },
+	{ 1, 0x00 },
+	{ 2, 0x00 },
+	{ 3, 0x00 },
+	{ 4, 0x00 },
+	{ 5, 0x00 },
+	{ 6, 0x00 },
+	{ 7, 0x00 },
+	{ 8, 0x00 },
+	{ 9, 0x00 },
+	{ 10, 0x00 },
+	{ 11, 0x00 },
+	{ 12, 0x00 },
+	{ 13, 0x00 },
+	{ 14, 0x00 },
+	{ 15, 0x00 },
+	{ 16, 0x00 },
+	{ 17, 0x00 },
+	{ 18, 0x00 },
+	{ 19, 0x00 },
+	{ 20, 0x00 },
+	{ 21, 0x00 },
+	{ 22, 0x00 },
+	{ 23, 0x00 },
+	{ 32, 0x00 },
+	{ 33, 0x00 },
+	{ 35, 0x00 },
+	{ 36, 0x00 },
+	{ 37, 0x00 },
+	{ 46, 0x00 },
+	{ 48, 0x00 },
+	{ 49, 0x00 },
+	{ 51, 0x00 },
+	{ 56, 0x00 },
+	{ 58, 0x00 },
+	{ 59, 0x00 },
+	{ 60, 0x00 },
+	{ 61, 0x00 },
+	{ 62, 0x00 },
+	{ 63, 0x00 },
+	{ 64, 0x00 },
+	{ 65, 0x00 },
+	{ 66, 0x00 },
+	{ 67, 0x00 },
+	{ 68, 0x00 },
+	{ 69, 0x00 },
+	{ 70, 0x00 },
+	{ 71, 0x00 },
+	{ 72, 0x00 },
+	{ 73, 0x00 },
+	{ 74, 0x00 },
+	{ 75, 0x00 },
+	{ 76, 0x00 },
+	{ 77, 0x00 },
+	{ 78, 0x00 },
+	{ 79, 0x00 },
+	{ 80, 0x00 },
+	{ 81, 0x00 },
+	{ 82, 0x00 },
+	{ 83, 0x00 },
+	{ 85, 0x00 },
+	{ 85, 0x00 },
+	{ 86, 0x00 },
+	{ 87, 0x00 },
+	{ 88, 0x00 },
+	{ 89, 0x00 },
+	{ 90, 0x00 },
+	{ 91, 0x00 },
+	{ 92, 0x00 },
+	{ 93, 0x00 },
+	{ 94, 0x00 },
+	{ 95, 0x00 },
+	{ 96, 0x01 },
+	{ 97, 0x00 },
+	{ 98, 0x00 },
+	{ 99, 0x00 },
+	{ 100, 0x00 },
+	{ 101, 0x00 },
+	{ 102, 0x00 },
+	{ 103, 0x01 },
+	{ 105, 0x01 },
+	{ 106, 0x00 },
+	{ 107, 0x01 },
+	{ 107, 0x00 },
+	{ 108, 0x00 },
+	{ 109, 0x00 },
+	{ 110, 0x00 },
+	{ 111, 0x02 },
+	{ 112, 0x02 },
+	{ 113, 0x00 },
+	{ 121, 0x80 },
+	{ 122, 0xBB },
+	{ 123, 0x80 },
+	{ 124, 0xBB },
+	{ 128, 0x00 },
+	{ 130, 0x00 },
+	{ 131, 0x00 },
+	{ 132, 0x00 },
+	{ 133, 0x0A },
+	{ 134, 0x0A },
+	{ 135, 0x0A },
+	{ 136, 0x0F },
+	{ 137, 0x00 },
+	{ 138, 0x73 },
+	{ 139, 0x33 },
+	{ 140, 0x73 },
+	{ 141, 0x33 },
+	{ 142, 0x73 },
+	{ 143, 0x33 },
+	{ 144, 0x73 },
+	{ 145, 0x33 },
+	{ 146, 0x73 },
+	{ 147, 0x33 },
+	{ 148, 0x73 },
+	{ 149, 0x33 },
+	{ 150, 0x73 },
+	{ 151, 0x33 },
+	{ 152, 0x00 },
+	{ 153, 0x00 },
+	{ 154, 0x00 },
+	{ 155, 0x00 },
+	{ 176, 0x00 },
+	{ 177, 0x00 },
+	{ 178, 0x00 },
+	{ 179, 0x00 },
+	{ 180, 0x00 },
+	{ 181, 0x00 },
+	{ 182, 0x00 },
+	{ 183, 0x00 },
+	{ 184, 0x00 },
+	{ 185, 0x00 },
+	{ 186, 0x00 },
+	{ 189, 0x00 },
+	{ 188, 0x00 },
+	{ 194, 0x00 },
+	{ 195, 0x00 },
+	{ 196, 0x00 },
+	{ 197, 0x00 },
+	{ 200, 0x00 },
+	{ 201, 0x00 },
+	{ 202, 0x00 },
+	{ 203, 0x00 },
+	{ 204, 0x00 },
+	{ 205, 0x00 },
+	{ 208, 0x00 },
+	{ 209, 0x00 },
+	{ 210, 0x00 },
+	{ 211, 0x00 },
+	{ 213, 0x00 },
+	{ 214, 0x00 },
+	{ 215, 0x00 },
+	{ 216, 0x00 },
+	{ 217, 0x00 },
+	{ 218, 0x00 },
+	{ 219, 0x00 },
+	{ 221, 0x00 },
+	{ 222, 0x00 },
+	{ 224, 0x00 },
+	{ 225, 0x00 },
+	{ 226, 0x00 },
+	{ 227, 0x00 },
+	{ 228, 0x00 },
+	{ 229, 0x00 },
+	{ 230, 0x13 },
+	{ 231, 0x00 },
+	{ 232, 0x80 },
+	{ 233, 0x0C },
+	{ 234, 0xDD },
+	{ 235, 0x00 },
+	{ 236, 0x04 },
+	{ 237, 0x00 },
+	{ 238, 0x00 },
+	{ 239, 0x00 },
+	{ 240, 0x00 },
+	{ 241, 0x00 },
+	{ 242, 0x00 },
+	{ 243, 0x00 },
+	{ 244, 0x00 },
+	{ 245, 0x00 },
+	{ 248, 0x00 },
+	{ 249, 0x00 },
+	{ 254, 0x00 },
+	{ 255, 0x00 },
+};
+
+/* codec private data */
+struct lm49453_priv {
+	struct regmap *regmap;
+	int fs_rate;
+};
+
+/* capture path controls */
+
+static const char *lm49453_mic2mode_text[] = {"Single Ended", "Differential"};
+
+static const SOC_ENUM_SINGLE_DECL(lm49453_mic2mode_enum, LM49453_P0_MICR_REG, 5,
+				  lm49453_mic2mode_text);
+
+static const char *lm49453_dmic_cfg_text[] = {"DMICDAT1", "DMICDAT2"};
+
+static const SOC_ENUM_SINGLE_DECL(lm49453_dmic12_cfg_enum,
+				  LM49453_P0_DIGITAL_MIC1_CONFIG_REG,
+				  7, lm49453_dmic_cfg_text);
+
+static const SOC_ENUM_SINGLE_DECL(lm49453_dmic34_cfg_enum,
+				  LM49453_P0_DIGITAL_MIC2_CONFIG_REG,
+				  7, lm49453_dmic_cfg_text);
+
+/* MUX Controls */
+static const char *lm49453_adcl_mux_text[] = { "MIC1", "Aux_L" };
+
+static const char *lm49453_adcr_mux_text[] = { "MIC2", "Aux_R" };
+
+static const struct soc_enum lm49453_adcl_enum =
+	SOC_ENUM_SINGLE(LM49453_P0_ANALOG_MIXER_ADC_REG, 0,
+			ARRAY_SIZE(lm49453_adcl_mux_text),
+			lm49453_adcl_mux_text);
+
+static const struct soc_enum lm49453_adcr_enum =
+	SOC_ENUM_SINGLE(LM49453_P0_ANALOG_MIXER_ADC_REG, 1,
+			ARRAY_SIZE(lm49453_adcr_mux_text),
+			lm49453_adcr_mux_text);
+
+static const struct snd_kcontrol_new lm49453_adcl_mux_control =
+	SOC_DAPM_ENUM("ADC Left Mux", lm49453_adcl_enum);
+
+static const struct snd_kcontrol_new lm49453_adcr_mux_control =
+	SOC_DAPM_ENUM("ADC Right Mux", lm49453_adcr_enum);
+
+static const struct snd_kcontrol_new lm49453_headset_left_mixer[] = {
+SOC_DAPM_SINGLE("Port1_1 Switch", LM49453_P0_DACHPL1_REG, 0, 1, 0),
+SOC_DAPM_SINGLE("Port1_2 Switch", LM49453_P0_DACHPL1_REG, 1, 1, 0),
+SOC_DAPM_SINGLE("Port1_3 Switch", LM49453_P0_DACHPL1_REG, 2, 1, 0),
+SOC_DAPM_SINGLE("Port1_4 Switch", LM49453_P0_DACHPL1_REG, 3, 1, 0),
+SOC_DAPM_SINGLE("Port1_5 Switch", LM49453_P0_DACHPL1_REG, 4, 1, 0),
+SOC_DAPM_SINGLE("Port1_6 Switch", LM49453_P0_DACHPL1_REG, 5, 1, 0),
+SOC_DAPM_SINGLE("Port1_7 Switch", LM49453_P0_DACHPL1_REG, 6, 1, 0),
+SOC_DAPM_SINGLE("Port1_8 Switch", LM49453_P0_DACHPL1_REG, 7, 1, 0),
+SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_DACHPL2_REG, 0, 1, 0),
+SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_DACHPL2_REG, 1, 1, 0),
+SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_DACHPL2_REG, 2, 1, 0),
+SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_DACHPL2_REG, 3, 1, 0),
+SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_DACHPL2_REG, 4, 1, 0),
+SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_DACHPL2_REG, 5, 1, 0),
+SOC_DAPM_SINGLE("Port2_1 Switch", LM49453_P0_DACHPL2_REG, 6, 1, 0),
+SOC_DAPM_SINGLE("Port2_2 Switch", LM49453_P0_DACHPL2_REG, 7, 1, 0),
+SOC_DAPM_SINGLE("Sidetone Switch", LM49453_P0_STN_SEL_REG, 0, 0, 0),
+};
+
+static const struct snd_kcontrol_new lm49453_headset_right_mixer[] = {
+SOC_DAPM_SINGLE("Port1_1 Switch", LM49453_P0_DACHPR1_REG, 0, 1, 0),
+SOC_DAPM_SINGLE("Port1_2 Switch", LM49453_P0_DACHPR1_REG, 1, 1, 0),
+SOC_DAPM_SINGLE("Port1_3 Switch", LM49453_P0_DACHPR1_REG, 2, 1, 0),
+SOC_DAPM_SINGLE("Port1_4 Switch", LM49453_P0_DACHPR1_REG, 3, 1, 0),
+SOC_DAPM_SINGLE("Port1_5 Switch", LM49453_P0_DACHPR1_REG, 4, 1, 0),
+SOC_DAPM_SINGLE("Port1_6 Switch", LM49453_P0_DACHPR1_REG, 5, 1, 0),
+SOC_DAPM_SINGLE("Port1_7 Switch", LM49453_P0_DACHPR1_REG, 6, 1, 0),
+SOC_DAPM_SINGLE("Port1_8 Switch", LM49453_P0_DACHPR1_REG, 7, 1, 0),
+SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_DACHPR2_REG, 0, 1, 0),
+SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_DACHPR2_REG, 1, 1, 0),
+SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_DACHPR2_REG, 2, 1, 0),
+SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_DACHPR2_REG, 3, 1, 0),
+SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_DACHPR2_REG, 4, 1, 0),
+SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_DACHPR2_REG, 5, 1, 0),
+SOC_DAPM_SINGLE("Port2_1 Switch", LM49453_P0_DACHPR2_REG, 6, 1, 0),
+SOC_DAPM_SINGLE("Port2_2 Switch", LM49453_P0_DACHPR2_REG, 7, 1, 0),
+SOC_DAPM_SINGLE("Sidetone Switch", LM49453_P0_STN_SEL_REG, 1, 0, 0),
+};
+
+static const struct snd_kcontrol_new lm49453_speaker_left_mixer[] = {
+SOC_DAPM_SINGLE("Port1_1 Switch", LM49453_P0_DACLSL1_REG, 0, 1, 0),
+SOC_DAPM_SINGLE("Port1_2 Switch", LM49453_P0_DACLSL1_REG, 1, 1, 0),
+SOC_DAPM_SINGLE("Port1_3 Switch", LM49453_P0_DACLSL1_REG, 2, 1, 0),
+SOC_DAPM_SINGLE("Port1_4 Switch", LM49453_P0_DACLSL1_REG, 3, 1, 0),
+SOC_DAPM_SINGLE("Port1_5 Switch", LM49453_P0_DACLSL1_REG, 4, 1, 0),
+SOC_DAPM_SINGLE("Port1_6 Switch", LM49453_P0_DACLSL1_REG, 5, 1, 0),
+SOC_DAPM_SINGLE("Port1_7 Switch", LM49453_P0_DACLSL1_REG, 6, 1, 0),
+SOC_DAPM_SINGLE("Port1_8 Switch", LM49453_P0_DACLSL1_REG, 7, 1, 0),
+SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_DACLSL2_REG, 0, 1, 0),
+SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_DACLSL2_REG, 1, 1, 0),
+SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_DACLSL2_REG, 2, 1, 0),
+SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_DACLSL2_REG, 3, 1, 0),
+SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_DACLSL2_REG, 4, 1, 0),
+SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_DACLSL2_REG, 5, 1, 0),
+SOC_DAPM_SINGLE("Port2_1 Switch", LM49453_P0_DACLSL2_REG, 6, 1, 0),
+SOC_DAPM_SINGLE("Port2_2 Switch", LM49453_P0_DACLSL2_REG, 7, 1, 0),
+SOC_DAPM_SINGLE("Sidetone Switch", LM49453_P0_STN_SEL_REG, 2, 0, 0),
+};
+
+static const struct snd_kcontrol_new lm49453_speaker_right_mixer[] = {
+SOC_DAPM_SINGLE("Port1_1 Switch", LM49453_P0_DACLSR1_REG, 0, 1, 0),
+SOC_DAPM_SINGLE("Port1_2 Switch", LM49453_P0_DACLSR1_REG, 1, 1, 0),
+SOC_DAPM_SINGLE("Port1_3 Switch", LM49453_P0_DACLSR1_REG, 2, 1, 0),
+SOC_DAPM_SINGLE("Port1_4 Switch", LM49453_P0_DACLSR1_REG, 3, 1, 0),
+SOC_DAPM_SINGLE("Port1_5 Switch", LM49453_P0_DACLSR1_REG, 4, 1, 0),
+SOC_DAPM_SINGLE("Port1_6 Switch", LM49453_P0_DACLSR1_REG, 5, 1, 0),
+SOC_DAPM_SINGLE("Port1_7 Switch", LM49453_P0_DACLSR1_REG, 6, 1, 0),
+SOC_DAPM_SINGLE("Port1_8 Switch", LM49453_P0_DACLSR1_REG, 7, 1, 0),
+SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_DACLSR2_REG, 0, 1, 0),
+SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_DACLSR2_REG, 1, 1, 0),
+SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_DACLSR2_REG, 2, 1, 0),
+SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_DACLSR2_REG, 3, 1, 0),
+SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_DACLSR2_REG, 4, 1, 0),
+SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_DACLSR2_REG, 5, 1, 0),
+SOC_DAPM_SINGLE("Port2_1 Switch", LM49453_P0_DACLSR2_REG, 6, 1, 0),
+SOC_DAPM_SINGLE("Port2_2 Switch", LM49453_P0_DACLSR2_REG, 7, 1, 0),
+SOC_DAPM_SINGLE("Sidetone Switch", LM49453_P0_STN_SEL_REG, 3, 0, 0),
+};
+
+static const struct snd_kcontrol_new lm49453_haptic_left_mixer[] = {
+SOC_DAPM_SINGLE("Port1_1 Switch", LM49453_P0_DACHAL1_REG, 0, 1, 0),
+SOC_DAPM_SINGLE("Port1_2 Switch", LM49453_P0_DACHAL1_REG, 1, 1, 0),
+SOC_DAPM_SINGLE("Port1_3 Switch", LM49453_P0_DACHAL1_REG, 2, 1, 0),
+SOC_DAPM_SINGLE("Port1_4 Switch", LM49453_P0_DACHAL1_REG, 3, 1, 0),
+SOC_DAPM_SINGLE("Port1_5 Switch", LM49453_P0_DACHAL1_REG, 4, 1, 0),
+SOC_DAPM_SINGLE("Port1_6 Switch", LM49453_P0_DACHAL1_REG, 5, 1, 0),
+SOC_DAPM_SINGLE("Port1_7 Switch", LM49453_P0_DACHAL1_REG, 6, 1, 0),
+SOC_DAPM_SINGLE("Port1_8 Switch", LM49453_P0_DACHAL1_REG, 7, 1, 0),
+SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_DACHAL2_REG, 0, 1, 0),
+SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_DACHAL2_REG, 1, 1, 0),
+SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_DACHAL2_REG, 2, 1, 0),
+SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_DACHAL2_REG, 3, 1, 0),
+SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_DACHAL2_REG, 4, 1, 0),
+SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_DACHAL2_REG, 5, 1, 0),
+SOC_DAPM_SINGLE("Port2_1 Switch", LM49453_P0_DACHAL2_REG, 6, 1, 0),
+SOC_DAPM_SINGLE("Port2_2 Switch", LM49453_P0_DACHAL2_REG, 7, 1, 0),
+SOC_DAPM_SINGLE("Sidetone Switch", LM49453_P0_STN_SEL_REG, 4, 0, 0),
+};
+
+static const struct snd_kcontrol_new lm49453_haptic_right_mixer[] = {
+SOC_DAPM_SINGLE("Port1_1 Switch", LM49453_P0_DACHAR1_REG, 0, 1, 0),
+SOC_DAPM_SINGLE("Port1_2 Switch", LM49453_P0_DACHAR1_REG, 1, 1, 0),
+SOC_DAPM_SINGLE("Port1_3 Switch", LM49453_P0_DACHAR1_REG, 2, 1, 0),
+SOC_DAPM_SINGLE("Port1_4 Switch", LM49453_P0_DACHAR1_REG, 3, 1, 0),
+SOC_DAPM_SINGLE("Port1_5 Switch", LM49453_P0_DACHAR1_REG, 4, 1, 0),
+SOC_DAPM_SINGLE("Port1_6 Switch", LM49453_P0_DACHAR1_REG, 5, 1, 0),
+SOC_DAPM_SINGLE("Port1_7 Switch", LM49453_P0_DACHAR1_REG, 6, 1, 0),
+SOC_DAPM_SINGLE("Port1_8 Switch", LM49453_P0_DACHAR1_REG, 7, 1, 0),
+SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_DACHAR2_REG, 0, 1, 0),
+SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_DACHAR2_REG, 1, 1, 0),
+SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_DACHAR2_REG, 2, 1, 0),
+SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_DACHAR2_REG, 3, 1, 0),
+SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_DACHAR2_REG, 4, 1, 0),
+SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_DACHAR2_REG, 5, 1, 0),
+SOC_DAPM_SINGLE("Port2_1 Switch", LM49453_P0_DACHAR2_REG, 6, 1, 0),
+SOC_DAPM_SINGLE("Port2_2 Switch", LM49453_P0_DACHAR2_REG, 7, 1, 0),
+SOC_DAPM_SINGLE("Sidetone Switch", LM49453_P0_STN_SEL_REG, 5, 0, 0),
+};
+
+static const struct snd_kcontrol_new lm49453_lineout_left_mixer[] = {
+SOC_DAPM_SINGLE("Port1_1 Switch", LM49453_P0_DACLOL1_REG, 0, 1, 0),
+SOC_DAPM_SINGLE("Port1_2 Switch", LM49453_P0_DACLOL1_REG, 1, 1, 0),
+SOC_DAPM_SINGLE("Port1_3 Switch", LM49453_P0_DACLOL1_REG, 2, 1, 0),
+SOC_DAPM_SINGLE("Port1_4 Switch", LM49453_P0_DACLOL1_REG, 3, 1, 0),
+SOC_DAPM_SINGLE("Port1_5 Switch", LM49453_P0_DACLOL1_REG, 4, 1, 0),
+SOC_DAPM_SINGLE("Port1_6 Switch", LM49453_P0_DACLOL1_REG, 5, 1, 0),
+SOC_DAPM_SINGLE("Port1_7 Switch", LM49453_P0_DACLOL1_REG, 6, 1, 0),
+SOC_DAPM_SINGLE("Port1_8 Switch", LM49453_P0_DACLOL1_REG, 7, 1, 0),
+SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_DACLOL2_REG, 0, 1, 0),
+SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_DACLOL2_REG, 1, 1, 0),
+SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_DACLOL2_REG, 2, 1, 0),
+SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_DACLOL2_REG, 3, 1, 0),
+SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_DACLOL2_REG, 4, 1, 0),
+SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_DACLOL2_REG, 5, 1, 0),
+SOC_DAPM_SINGLE("Port2_1 Switch", LM49453_P0_DACLOL2_REG, 6, 1, 0),
+SOC_DAPM_SINGLE("Port2_2 Switch", LM49453_P0_DACLOL2_REG, 7, 1, 0),
+SOC_DAPM_SINGLE("Sidetone Switch", LM49453_P0_STN_SEL_REG, 6, 0, 0),
+};
+
+static const struct snd_kcontrol_new lm49453_lineout_right_mixer[] = {
+SOC_DAPM_SINGLE("Port1_1 Switch", LM49453_P0_DACLOR1_REG, 0, 1, 0),
+SOC_DAPM_SINGLE("Port1_2 Switch", LM49453_P0_DACLOR1_REG, 1, 1, 0),
+SOC_DAPM_SINGLE("Port1_3 Switch", LM49453_P0_DACLOR1_REG, 2, 1, 0),
+SOC_DAPM_SINGLE("Port1_4 Switch", LM49453_P0_DACLOR1_REG, 3, 1, 0),
+SOC_DAPM_SINGLE("Port1_5 Switch", LM49453_P0_DACLOR1_REG, 4, 1, 0),
+SOC_DAPM_SINGLE("Port1_6 Switch", LM49453_P0_DACLOR1_REG, 5, 1, 0),
+SOC_DAPM_SINGLE("Port1_7 Switch", LM49453_P0_DACLOR1_REG, 6, 1, 0),
+SOC_DAPM_SINGLE("Port1_8 Switch", LM49453_P0_DACLOR1_REG, 7, 1, 0),
+SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_DACLOR2_REG, 0, 1, 0),
+SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_DACLOR2_REG, 1, 1, 0),
+SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_DACLOR2_REG, 2, 1, 0),
+SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_DACLOR2_REG, 3, 1, 0),
+SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_DACLOR2_REG, 4, 1, 0),
+SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_DACLOR2_REG, 5, 1, 0),
+SOC_DAPM_SINGLE("Port2_1 Switch", LM49453_P0_DACLOR2_REG, 6, 1, 0),
+SOC_DAPM_SINGLE("Port2_2 Switch", LM49453_P0_DACLOR2_REG, 7, 1, 0),
+SOC_DAPM_SINGLE("Sidetone Switch", LM49453_P0_STN_SEL_REG, 7, 0, 0),
+};
+
+static const struct snd_kcontrol_new lm49453_port1_tx1_mixer[] = {
+SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_PORT1_TX1_REG, 0, 1, 0),
+SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_PORT1_TX1_REG, 1, 1, 0),
+SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_PORT1_TX1_REG, 2, 1, 0),
+SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_PORT1_TX1_REG, 3, 1, 0),
+SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_PORT1_TX1_REG, 4, 1, 0),
+SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_PORT1_TX1_REG, 5, 1, 0),
+SOC_DAPM_SINGLE("Port1_1 Switch", LM49453_P0_PORT1_TX1_REG, 6, 1, 0),
+SOC_DAPM_SINGLE("Port2_1 Switch", LM49453_P0_PORT1_TX1_REG, 7, 1, 0),
+};
+
+static const struct snd_kcontrol_new lm49453_port1_tx2_mixer[] = {
+SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_PORT1_TX2_REG, 0, 1, 0),
+SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_PORT1_TX2_REG, 1, 1, 0),
+SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_PORT1_TX2_REG, 2, 1, 0),
+SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_PORT1_TX2_REG, 3, 1, 0),
+SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_PORT1_TX2_REG, 4, 1, 0),
+SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_PORT1_TX2_REG, 5, 1, 0),
+SOC_DAPM_SINGLE("Port1_2 Switch", LM49453_P0_PORT1_TX2_REG, 6, 1, 0),
+SOC_DAPM_SINGLE("Port2_2 Switch", LM49453_P0_PORT1_TX2_REG, 7, 1, 0),
+};
+
+static const struct snd_kcontrol_new lm49453_port1_tx3_mixer[] = {
+SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_PORT1_TX3_REG, 0, 1, 0),
+SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_PORT1_TX3_REG, 1, 1, 0),
+SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_PORT1_TX3_REG, 2, 1, 0),
+SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_PORT1_TX3_REG, 3, 1, 0),
+SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_PORT1_TX3_REG, 4, 1, 0),
+SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_PORT1_TX3_REG, 5, 1, 0),
+SOC_DAPM_SINGLE("Port1_3 Switch", LM49453_P0_PORT1_TX3_REG, 6, 1, 0),
+};
+
+static const struct snd_kcontrol_new lm49453_port1_tx4_mixer[] = {
+SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_PORT1_TX4_REG, 0, 1, 0),
+SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_PORT1_TX4_REG, 1, 1, 0),
+SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_PORT1_TX4_REG, 2, 1, 0),
+SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_PORT1_TX4_REG, 3, 1, 0),
+SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_PORT1_TX4_REG, 4, 1, 0),
+SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_PORT1_TX4_REG, 5, 1, 0),
+SOC_DAPM_SINGLE("Port1_4 Switch", LM49453_P0_PORT1_TX4_REG, 6, 1, 0),
+};
+
+static const struct snd_kcontrol_new lm49453_port1_tx5_mixer[] = {
+SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_PORT1_TX5_REG, 0, 1, 0),
+SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_PORT1_TX5_REG, 1, 1, 0),
+SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_PORT1_TX5_REG, 2, 1, 0),
+SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_PORT1_TX5_REG, 3, 1, 0),
+SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_PORT1_TX5_REG, 4, 1, 0),
+SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_PORT1_TX5_REG, 5, 1, 0),
+SOC_DAPM_SINGLE("Port1_5 Switch", LM49453_P0_PORT1_TX5_REG, 6, 1, 0),
+};
+
+static const struct snd_kcontrol_new lm49453_port1_tx6_mixer[] = {
+SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_PORT1_TX6_REG, 0, 1, 0),
+SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_PORT1_TX6_REG, 1, 1, 0),
+SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_PORT1_TX6_REG, 2, 1, 0),
+SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_PORT1_TX6_REG, 3, 1, 0),
+SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_PORT1_TX6_REG, 4, 1, 0),
+SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_PORT1_TX6_REG, 5, 1, 0),
+SOC_DAPM_SINGLE("Port1_6 Switch", LM49453_P0_PORT1_TX6_REG, 6, 1, 0),
+};
+
+static const struct snd_kcontrol_new lm49453_port1_tx7_mixer[] = {
+SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_PORT1_TX7_REG, 0, 1, 0),
+SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_PORT1_TX7_REG, 1, 1, 0),
+SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_PORT1_TX7_REG, 2, 1, 0),
+SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_PORT1_TX7_REG, 3, 1, 0),
+SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_PORT1_TX7_REG, 4, 1, 0),
+SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_PORT1_TX7_REG, 5, 1, 0),
+SOC_DAPM_SINGLE("Port1_7 Switch", LM49453_P0_PORT1_TX7_REG, 6, 1, 0),
+};
+
+static const struct snd_kcontrol_new lm49453_port1_tx8_mixer[] = {
+SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_PORT1_TX8_REG, 0, 1, 0),
+SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_PORT1_TX8_REG, 1, 1, 0),
+SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_PORT1_TX8_REG, 2, 1, 0),
+SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_PORT1_TX8_REG, 3, 1, 0),
+SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_PORT1_TX8_REG, 4, 1, 0),
+SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_PORT1_TX8_REG, 5, 1, 0),
+SOC_DAPM_SINGLE("Port1_8 Switch", LM49453_P0_PORT1_TX8_REG, 6, 1, 0),
+};
+
+static const struct snd_kcontrol_new lm49453_port2_tx1_mixer[] = {
+SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_PORT2_TX1_REG, 0, 1, 0),
+SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_PORT2_TX1_REG, 1, 1, 0),
+SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_PORT2_TX1_REG, 2, 1, 0),
+SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_PORT2_TX1_REG, 3, 1, 0),
+SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_PORT2_TX1_REG, 4, 1, 0),
+SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_PORT2_TX1_REG, 5, 1, 0),
+SOC_DAPM_SINGLE("Port1_1 Switch", LM49453_P0_PORT2_TX1_REG, 6, 1, 0),
+SOC_DAPM_SINGLE("Port2_1 Switch", LM49453_P0_PORT2_TX1_REG, 7, 1, 0),
+};
+
+static const struct snd_kcontrol_new lm49453_port2_tx2_mixer[] = {
+SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_PORT2_TX2_REG, 0, 1, 0),
+SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_PORT2_TX2_REG, 1, 1, 0),
+SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_PORT2_TX2_REG, 2, 1, 0),
+SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_PORT2_TX2_REG, 3, 1, 0),
+SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_PORT2_TX2_REG, 4, 1, 0),
+SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_PORT2_TX2_REG, 5, 1, 0),
+SOC_DAPM_SINGLE("Port1_2 Switch", LM49453_P0_PORT2_TX2_REG, 6, 1, 0),
+SOC_DAPM_SINGLE("Port2_2 Switch", LM49453_P0_PORT2_TX2_REG, 7, 1, 0),
+};
+
+/* TLV Declarations */
+static const DECLARE_TLV_DB_SCALE(digital_tlv, -7650, 150, 1);
+static const DECLARE_TLV_DB_SCALE(port_tlv, 0, 600, 0);
+
+static const struct snd_kcontrol_new lm49453_sidetone_mixer_controls[] = {
+/* Sidetone supports mono only */
+SOC_DAPM_SINGLE_TLV("Sidetone ADCL Volume", LM49453_P0_STN_VOL_ADCL_REG,
+		     0, 0x3F, 0, digital_tlv),
+SOC_DAPM_SINGLE_TLV("Sidetone ADCR Volume", LM49453_P0_STN_VOL_ADCR_REG,
+		     0, 0x3F, 0, digital_tlv),
+SOC_DAPM_SINGLE_TLV("Sidetone DMIC1L Volume", LM49453_P0_STN_VOL_DMIC1L_REG,
+		     0, 0x3F, 0, digital_tlv),
+SOC_DAPM_SINGLE_TLV("Sidetone DMIC1R Volume", LM49453_P0_STN_VOL_DMIC1R_REG,
+		     0, 0x3F, 0, digital_tlv),
+SOC_DAPM_SINGLE_TLV("Sidetone DMIC2L Volume", LM49453_P0_STN_VOL_DMIC2L_REG,
+		     0, 0x3F, 0, digital_tlv),
+SOC_DAPM_SINGLE_TLV("Sidetone DMIC2R Volume", LM49453_P0_STN_VOL_DMIC2R_REG,
+		     0, 0x3F, 0, digital_tlv),
+};
+
+static const struct snd_kcontrol_new lm49453_snd_controls[] = {
+	/* mic1 and mic2 supports mono only */
+	SOC_SINGLE_TLV("Mic1 Volume", LM49453_P0_ADC_LEVELL_REG, 0, 6,
+			0, digital_tlv),
+	SOC_SINGLE_TLV("Mic2 Volume", LM49453_P0_ADC_LEVELR_REG, 0, 6,
+			0, digital_tlv),
+
+	SOC_DOUBLE_R_TLV("DMIC1 Volume", LM49453_P0_DMIC1_LEVELL_REG,
+			  LM49453_P0_DMIC1_LEVELR_REG, 0, 6, 0, digital_tlv),
+	SOC_DOUBLE_R_TLV("DMIC2 Volume", LM49453_P0_DMIC2_LEVELL_REG,
+			  LM49453_P0_DMIC2_LEVELR_REG, 0, 6, 0, digital_tlv),
+
+	SOC_DAPM_ENUM("Mic2Mode", lm49453_mic2mode_enum),
+	SOC_DAPM_ENUM("DMIC12 SRC", lm49453_dmic12_cfg_enum),
+	SOC_DAPM_ENUM("DMIC34 SRC", lm49453_dmic34_cfg_enum),
+
+	/* Capture path filter enable */
+	SOC_SINGLE("DMIC1 HPFilter Switch", LM49453_P0_ADC_FX_ENABLES_REG,
+					    0, 1, 0),
+	SOC_SINGLE("DMIC2 HPFilter Switch", LM49453_P0_ADC_FX_ENABLES_REG,
+					    1, 1, 0),
+	SOC_SINGLE("ADC HPFilter Switch", LM49453_P0_ADC_FX_ENABLES_REG,
+					  2, 1, 0),
+
+	SOC_DOUBLE_R_TLV("DAC HP Volume", LM49453_P0_DAC_HP_LEVELL_REG,
+			  LM49453_P0_DAC_HP_LEVELR_REG, 0, 6, 0, digital_tlv),
+	SOC_DOUBLE_R_TLV("DAC LO Volume", LM49453_P0_DAC_LO_LEVELL_REG,
+			  LM49453_P0_DAC_LO_LEVELR_REG, 0, 6, 0, digital_tlv),
+	SOC_DOUBLE_R_TLV("DAC LS Volume", LM49453_P0_DAC_LS_LEVELL_REG,
+			  LM49453_P0_DAC_LS_LEVELR_REG, 0, 6, 0, digital_tlv),
+	SOC_DOUBLE_R_TLV("DAC HA Volume", LM49453_P0_DAC_HA_LEVELL_REG,
+			  LM49453_P0_DAC_HA_LEVELR_REG, 0, 6, 0, digital_tlv),
+
+	SOC_SINGLE_TLV("EP Volume", LM49453_P0_DAC_LS_LEVELL_REG,
+			0, 6, 0, digital_tlv),
+
+	SOC_SINGLE_TLV("PORT1_1_RX_LVL Volume", LM49453_P0_PORT1_RX_LVL1_REG,
+			0, 3, 0, port_tlv),
+	SOC_SINGLE_TLV("PORT1_2_RX_LVL Volume", LM49453_P0_PORT1_RX_LVL1_REG,
+			2, 3, 0, port_tlv),
+	SOC_SINGLE_TLV("PORT1_3_RX_LVL Volume", LM49453_P0_PORT1_RX_LVL1_REG,
+			4, 3, 0, port_tlv),
+	SOC_SINGLE_TLV("PORT1_4_RX_LVL Volume", LM49453_P0_PORT1_RX_LVL1_REG,
+			6, 3, 0, port_tlv),
+	SOC_SINGLE_TLV("PORT1_5_RX_LVL Volume", LM49453_P0_PORT1_RX_LVL2_REG,
+			0, 3, 0, port_tlv),
+	SOC_SINGLE_TLV("PORT1_6_RX_LVL Volume", LM49453_P0_PORT1_RX_LVL2_REG,
+			2, 3, 0, port_tlv),
+	SOC_SINGLE_TLV("PORT1_7_RX_LVL Volume", LM49453_P0_PORT1_RX_LVL2_REG,
+			4, 3, 0, port_tlv),
+	SOC_SINGLE_TLV("PORT1_8_RX_LVL Volume", LM49453_P0_PORT1_RX_LVL2_REG,
+			6, 3, 0, port_tlv),
+
+	SOC_SINGLE_TLV("PORT2_1_RX_LVL Volume", LM49453_P0_PORT2_RX_LVL_REG,
+			0, 3, 0, port_tlv),
+	SOC_SINGLE_TLV("PORT2_2_RX_LVL Volume", LM49453_P0_PORT2_RX_LVL_REG,
+			2, 3, 0, port_tlv),
+
+	SOC_SINGLE("Port1 Playback Switch", LM49453_P0_AUDIO_PORT1_BASIC_REG,
+		    1, 1, 0),
+	SOC_SINGLE("Port2 Playback Switch", LM49453_P0_AUDIO_PORT2_BASIC_REG,
+		    1, 1, 0),
+	SOC_SINGLE("Port1 Capture Switch", LM49453_P0_AUDIO_PORT1_BASIC_REG,
+		    2, 1, 0),
+	SOC_SINGLE("Port2 Capture Switch", LM49453_P0_AUDIO_PORT2_BASIC_REG,
+		    2, 1, 0)
+
+};
+
+/* DAPM widgets */
+static const struct snd_soc_dapm_widget lm49453_dapm_widgets[] = {
+
+	/* All end points HP,EP, LS, Lineout and Haptic */
+	SND_SOC_DAPM_OUTPUT("HPOUTL"),
+	SND_SOC_DAPM_OUTPUT("HPOUTR"),
+	SND_SOC_DAPM_OUTPUT("EPOUT"),
+	SND_SOC_DAPM_OUTPUT("LSOUTL"),
+	SND_SOC_DAPM_OUTPUT("LSOUTR"),
+	SND_SOC_DAPM_OUTPUT("LOOUTR"),
+	SND_SOC_DAPM_OUTPUT("LOOUTL"),
+	SND_SOC_DAPM_OUTPUT("HAOUTL"),
+	SND_SOC_DAPM_OUTPUT("HAOUTR"),
+
+	SND_SOC_DAPM_INPUT("AMIC1"),
+	SND_SOC_DAPM_INPUT("AMIC2"),
+	SND_SOC_DAPM_INPUT("DMIC1DAT"),
+	SND_SOC_DAPM_INPUT("DMIC2DAT"),
+	SND_SOC_DAPM_INPUT("AUXL"),
+	SND_SOC_DAPM_INPUT("AUXR"),
+
+	SND_SOC_DAPM_PGA("PORT1_1_RX", SND_SOC_NOPM, 0, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("PORT1_2_RX", SND_SOC_NOPM, 0, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("PORT1_3_RX", SND_SOC_NOPM, 0, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("PORT1_4_RX", SND_SOC_NOPM, 0, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("PORT1_5_RX", SND_SOC_NOPM, 0, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("PORT1_6_RX", SND_SOC_NOPM, 0, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("PORT1_7_RX", SND_SOC_NOPM, 0, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("PORT1_8_RX", SND_SOC_NOPM, 0, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("PORT2_1_RX", SND_SOC_NOPM, 0, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("PORT2_2_RX", SND_SOC_NOPM, 0, 0, NULL, 0),
+
+	SND_SOC_DAPM_SUPPLY("AMIC1Bias", LM49453_P0_MICL_REG, 6, 0, NULL, 0),
+	SND_SOC_DAPM_SUPPLY("AMIC2Bias", LM49453_P0_MICR_REG, 6, 0, NULL, 0),
+
+	/* playback path driver enables */
+	SND_SOC_DAPM_OUT_DRV("Headset Switch",
+			LM49453_P0_PMC_SETUP_REG, 0, 0, NULL, 0),
+	SND_SOC_DAPM_OUT_DRV("Earpiece Switch",
+			LM49453_P0_EP_REG, 0, 0, NULL, 0),
+	SND_SOC_DAPM_OUT_DRV("Speaker Left Switch",
+			LM49453_P0_DIS_PKVL_FB_REG, 0, 1, NULL, 0),
+	SND_SOC_DAPM_OUT_DRV("Speaker Right Switch",
+			LM49453_P0_DIS_PKVL_FB_REG, 1, 1, NULL, 0),
+	SND_SOC_DAPM_OUT_DRV("Haptic Left Switch",
+			LM49453_P0_DIS_PKVL_FB_REG, 2, 1, NULL, 0),
+	SND_SOC_DAPM_OUT_DRV("Haptic Right Switch",
+			LM49453_P0_DIS_PKVL_FB_REG, 3, 1, NULL, 0),
+
+	/* DAC */
+	SND_SOC_DAPM_DAC("HPL DAC", "Headset", SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_DAC("HPR DAC", "Headset", SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_DAC("LSL DAC", "Speaker", SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_DAC("LSR DAC", "Speaker", SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_DAC("HAL DAC", "Haptic", SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_DAC("HAR DAC", "Haptic", SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_DAC("LOL DAC", "Lineout", SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_DAC("LOR DAC", "Lineout", SND_SOC_NOPM, 0, 0),
+
+
+	SND_SOC_DAPM_PGA("AUXL Input",
+			LM49453_P0_ANALOG_MIXER_ADC_REG, 2, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("AUXR Input",
+			LM49453_P0_ANALOG_MIXER_ADC_REG, 3, 0, NULL, 0),
+
+	SND_SOC_DAPM_PGA("Sidetone", SND_SOC_NOPM, 0, 0, NULL, 0),
+
+	/* ADC */
+	SND_SOC_DAPM_ADC("DMIC1 Left", "Capture", SND_SOC_NOPM, 1, 0),
+	SND_SOC_DAPM_ADC("DMIC1 Right", "Capture", SND_SOC_NOPM, 1, 0),
+	SND_SOC_DAPM_ADC("DMIC2 Left", "Capture", SND_SOC_NOPM, 1, 0),
+	SND_SOC_DAPM_ADC("DMIC2 Right", "Capture", SND_SOC_NOPM, 1, 0),
+
+	SND_SOC_DAPM_ADC("ADC Left", "Capture", SND_SOC_NOPM, 1, 0),
+	SND_SOC_DAPM_ADC("ADC Right", "Capture", SND_SOC_NOPM, 0, 0),
+
+	SND_SOC_DAPM_MUX("ADCL Mux", SND_SOC_NOPM, 0, 0,
+			  &lm49453_adcl_mux_control),
+	SND_SOC_DAPM_MUX("ADCR Mux", SND_SOC_NOPM, 0, 0,
+			  &lm49453_adcr_mux_control),
+
+	SND_SOC_DAPM_MUX("Mic1 Input",
+			SND_SOC_NOPM, 0, 0, &lm49453_adcl_mux_control),
+
+	SND_SOC_DAPM_MUX("Mic2 Input",
+			SND_SOC_NOPM, 0, 0, &lm49453_adcr_mux_control),
+
+	/* AIF */
+	SND_SOC_DAPM_AIF_IN("PORT1_SDI", NULL, 0,
+			    LM49453_P0_PULL_CONFIG1_REG, 2, 0),
+	SND_SOC_DAPM_AIF_IN("PORT2_SDI", NULL, 0,
+			    LM49453_P0_PULL_CONFIG1_REG, 6, 0),
+
+	SND_SOC_DAPM_AIF_OUT("PORT1_SDO", NULL, 0,
+			     LM49453_P0_PULL_CONFIG1_REG, 3, 0),
+	SND_SOC_DAPM_AIF_OUT("PORT2_SDO", NULL, 0,
+			      LM49453_P0_PULL_CONFIG1_REG, 7, 0),
+
+	/* Port1 TX controls */
+	SND_SOC_DAPM_OUT_DRV("P1_1_TX", SND_SOC_NOPM, 0, 0, NULL, 0),
+	SND_SOC_DAPM_OUT_DRV("P1_2_TX", SND_SOC_NOPM, 0, 0, NULL, 0),
+	SND_SOC_DAPM_OUT_DRV("P1_3_TX", SND_SOC_NOPM, 0, 0, NULL, 0),
+	SND_SOC_DAPM_OUT_DRV("P1_4_TX", SND_SOC_NOPM, 0, 0, NULL, 0),
+	SND_SOC_DAPM_OUT_DRV("P1_5_TX", SND_SOC_NOPM, 0, 0, NULL, 0),
+	SND_SOC_DAPM_OUT_DRV("P1_6_TX", SND_SOC_NOPM, 0, 0, NULL, 0),
+	SND_SOC_DAPM_OUT_DRV("P1_7_TX", SND_SOC_NOPM, 0, 0, NULL, 0),
+	SND_SOC_DAPM_OUT_DRV("P1_8_TX", SND_SOC_NOPM, 0, 0, NULL, 0),
+
+	/* Port2 TX controls */
+	SND_SOC_DAPM_OUT_DRV("P2_1_TX", SND_SOC_NOPM, 0, 0, NULL, 0),
+	SND_SOC_DAPM_OUT_DRV("P2_2_TX", SND_SOC_NOPM, 0, 0, NULL, 0),
+
+	/* Sidetone Mixer */
+	SND_SOC_DAPM_MIXER("Sidetone Mixer", SND_SOC_NOPM, 0, 0,
+			    lm49453_sidetone_mixer_controls,
+			    ARRAY_SIZE(lm49453_sidetone_mixer_controls)),
+
+	/* DAC MIXERS */
+	SND_SOC_DAPM_MIXER("HPL Mixer", SND_SOC_NOPM, 0, 0,
+			    lm49453_headset_left_mixer,
+			    ARRAY_SIZE(lm49453_headset_left_mixer)),
+	SND_SOC_DAPM_MIXER("HPR Mixer", SND_SOC_NOPM, 0, 0,
+			    lm49453_headset_right_mixer,
+			    ARRAY_SIZE(lm49453_headset_right_mixer)),
+	SND_SOC_DAPM_MIXER("LOL Mixer", SND_SOC_NOPM, 0, 0,
+			    lm49453_lineout_left_mixer,
+			    ARRAY_SIZE(lm49453_lineout_left_mixer)),
+	SND_SOC_DAPM_MIXER("LOR Mixer", SND_SOC_NOPM, 0, 0,
+			    lm49453_lineout_right_mixer,
+			    ARRAY_SIZE(lm49453_lineout_right_mixer)),
+	SND_SOC_DAPM_MIXER("LSL Mixer", SND_SOC_NOPM, 0, 0,
+			    lm49453_speaker_left_mixer,
+			    ARRAY_SIZE(lm49453_speaker_left_mixer)),
+	SND_SOC_DAPM_MIXER("LSR Mixer", SND_SOC_NOPM, 0, 0,
+			    lm49453_speaker_right_mixer,
+			    ARRAY_SIZE(lm49453_speaker_right_mixer)),
+	SND_SOC_DAPM_MIXER("HAL Mixer", SND_SOC_NOPM, 0, 0,
+			    lm49453_haptic_left_mixer,
+			    ARRAY_SIZE(lm49453_haptic_left_mixer)),
+	SND_SOC_DAPM_MIXER("HAR Mixer", SND_SOC_NOPM, 0, 0,
+			    lm49453_haptic_right_mixer,
+			    ARRAY_SIZE(lm49453_haptic_right_mixer)),
+
+	/* Capture Mixer */
+	SND_SOC_DAPM_MIXER("Port1_1 Mixer", SND_SOC_NOPM, 0, 0,
+			    lm49453_port1_tx1_mixer,
+			    ARRAY_SIZE(lm49453_port1_tx1_mixer)),
+	SND_SOC_DAPM_MIXER("Port1_2 Mixer", SND_SOC_NOPM, 0, 0,
+			    lm49453_port1_tx2_mixer,
+			    ARRAY_SIZE(lm49453_port1_tx2_mixer)),
+	SND_SOC_DAPM_MIXER("Port1_3 Mixer", SND_SOC_NOPM, 0, 0,
+			    lm49453_port1_tx3_mixer,
+			    ARRAY_SIZE(lm49453_port1_tx3_mixer)),
+	SND_SOC_DAPM_MIXER("Port1_4 Mixer", SND_SOC_NOPM, 0, 0,
+			    lm49453_port1_tx4_mixer,
+			    ARRAY_SIZE(lm49453_port1_tx4_mixer)),
+	SND_SOC_DAPM_MIXER("Port1_5 Mixer", SND_SOC_NOPM, 0, 0,
+			    lm49453_port1_tx5_mixer,
+			    ARRAY_SIZE(lm49453_port1_tx5_mixer)),
+	SND_SOC_DAPM_MIXER("Port1_6 Mixer", SND_SOC_NOPM, 0, 0,
+			    lm49453_port1_tx6_mixer,
+			    ARRAY_SIZE(lm49453_port1_tx6_mixer)),
+	SND_SOC_DAPM_MIXER("Port1_7 Mixer", SND_SOC_NOPM, 0, 0,
+			    lm49453_port1_tx7_mixer,
+			    ARRAY_SIZE(lm49453_port1_tx7_mixer)),
+	SND_SOC_DAPM_MIXER("Port1_8 Mixer", SND_SOC_NOPM, 0, 0,
+			    lm49453_port1_tx8_mixer,
+			    ARRAY_SIZE(lm49453_port1_tx8_mixer)),
+
+	SND_SOC_DAPM_MIXER("Port2_1 Mixer", SND_SOC_NOPM, 0, 0,
+			    lm49453_port2_tx1_mixer,
+			    ARRAY_SIZE(lm49453_port2_tx1_mixer)),
+	SND_SOC_DAPM_MIXER("Port2_2 Mixer", SND_SOC_NOPM, 0, 0,
+			    lm49453_port2_tx2_mixer,
+			    ARRAY_SIZE(lm49453_port2_tx2_mixer)),
+};
+
+static const struct snd_soc_dapm_route lm49453_audio_map[] = {
+	/* Port SDI mapping */
+	{ "PORT1_1_RX", "Port1 Playback Switch", "PORT1_SDI" },
+	{ "PORT1_2_RX", "Port1 Playback Switch", "PORT1_SDI" },
+	{ "PORT1_3_RX", "Port1 Playback Switch", "PORT1_SDI" },
+	{ "PORT1_4_RX", "Port1 Playback Switch", "PORT1_SDI" },
+	{ "PORT1_5_RX", "Port1 Playback Switch", "PORT1_SDI" },
+	{ "PORT1_6_RX", "Port1 Playback Switch", "PORT1_SDI" },
+	{ "PORT1_7_RX", "Port1 Playback Switch", "PORT1_SDI" },
+	{ "PORT1_8_RX", "Port1 Playback Switch", "PORT1_SDI" },
+
+	{ "PORT2_1_RX", "Port2 Playback Switch", "PORT2_SDI" },
+	{ "PORT2_2_RX", "Port2 Playback Switch", "PORT2_SDI" },
+
+	/* HP mapping */
+	{ "HPL Mixer", "Port1_1 Switch", "PORT1_1_RX" },
+	{ "HPL Mixer", "Port1_2 Switch", "PORT1_2_RX" },
+	{ "HPL Mixer", "Port1_3 Switch", "PORT1_3_RX" },
+	{ "HPL Mixer", "Port1_4 Switch", "PORT1_4_RX" },
+	{ "HPL Mixer", "Port1_5 Switch", "PORT1_5_RX" },
+	{ "HPL Mixer", "Port1_6 Switch", "PORT1_6_RX" },
+	{ "HPL Mixer", "Port1_7 Switch", "PORT1_7_RX" },
+	{ "HPL Mixer", "Port1_8 Switch", "PORT1_8_RX" },
+
+	{ "HPL Mixer", "Port2_1 Switch", "PORT2_1_RX" },
+	{ "HPL Mixer", "Port2_2 Switch", "PORT2_2_RX" },
+
+	{ "HPL Mixer", "ADCL Switch", "ADC Left" },
+	{ "HPL Mixer", "ADCR Switch", "ADC Right" },
+	{ "HPL Mixer", "DMIC1L Switch", "DMIC1 Left" },
+	{ "HPL Mixer", "DMIC1R Switch", "DMIC1 Right" },
+	{ "HPL Mixer", "DMIC2L Switch", "DMIC2 Left" },
+	{ "HPL Mixer", "DMIC2R Switch", "DMIC2 Right" },
+	{ "HPL Mixer", "Sidetone Switch", "Sidetone" },
+
+	{ "HPL DAC", NULL, "HPL Mixer" },
+
+	{ "HPR Mixer", "Port1_1 Switch", "PORT1_1_RX" },
+	{ "HPR Mixer", "Port1_2 Switch", "PORT1_2_RX" },
+	{ "HPR Mixer", "Port1_3 Switch", "PORT1_3_RX" },
+	{ "HPR Mixer", "Port1_4 Switch", "PORT1_4_RX" },
+	{ "HPR Mixer", "Port1_5 Switch", "PORT1_5_RX" },
+	{ "HPR Mixer", "Port1_6 Switch", "PORT1_6_RX" },
+	{ "HPR Mixer", "Port1_7 Switch", "PORT1_7_RX" },
+	{ "HPR Mixer", "Port1_8 Switch", "PORT1_8_RX" },
+
+	/* Port 2 */
+	{ "HPR Mixer", "Port2_1 Switch", "PORT2_1_RX" },
+	{ "HPR Mixer", "Port2_2 Switch", "PORT2_2_RX" },
+
+	{ "HPR Mixer", "ADCL Switch", "ADC Left" },
+	{ "HPR Mixer", "ADCR Switch", "ADC Right" },
+	{ "HPR Mixer", "DMIC1L Switch", "DMIC1 Left" },
+	{ "HPR Mixer", "DMIC1R Switch", "DMIC1 Right" },
+	{ "HPR Mixer", "DMIC2L Switch", "DMIC2 Left" },
+	{ "HPR Mixer", "DMIC2L Switch", "DMIC2 Right" },
+	{ "HPR Mixer", "Sidetone Switch", "Sidetone" },
+
+	{ "HPR DAC", NULL, "HPR Mixer" },
+
+	{ "HPOUTL", "Headset Switch", "HPL DAC"},
+	{ "HPOUTR", "Headset Switch", "HPR DAC"},
+
+	/* EP map */
+	{ "EPOUT", "Earpiece Switch", "HPL DAC" },
+
+	/* Speaker map */
+	{ "LSL Mixer", "Port1_1 Switch", "PORT1_1_RX" },
+	{ "LSL Mixer", "Port1_2 Switch", "PORT1_2_RX" },
+	{ "LSL Mixer", "Port1_3 Switch", "PORT1_3_RX" },
+	{ "LSL Mixer", "Port1_4 Switch", "PORT1_4_RX" },
+	{ "LSL Mixer", "Port1_5 Switch", "PORT1_5_RX" },
+	{ "LSL Mixer", "Port1_6 Switch", "PORT1_6_RX" },
+	{ "LSL Mixer", "Port1_7 Switch", "PORT1_7_RX" },
+	{ "LSL Mixer", "Port1_8 Switch", "PORT1_8_RX" },
+
+	/* Port 2 */
+	{ "LSL Mixer", "Port2_1 Switch", "PORT2_1_RX" },
+	{ "LSL Mixer", "Port2_2 Switch", "PORT2_2_RX" },
+
+	{ "LSL Mixer", "ADCL Switch", "ADC Left" },
+	{ "LSL Mixer", "ADCR Switch", "ADC Right" },
+	{ "LSL Mixer", "DMIC1L Switch", "DMIC1 Left" },
+	{ "LSL Mixer", "DMIC1R Switch", "DMIC1 Right" },
+	{ "LSL Mixer", "DMIC2L Switch", "DMIC2 Left" },
+	{ "LSL Mixer", "DMIC2R Switch", "DMIC2 Right" },
+	{ "LSL Mixer", "Sidetone Switch", "Sidetone" },
+
+	{ "LSL DAC", NULL, "LSL Mixer" },
+
+	{ "LSR Mixer", "Port1_1 Switch", "PORT1_1_RX" },
+	{ "LSR Mixer", "Port1_2 Switch", "PORT1_2_RX" },
+	{ "LSR Mixer", "Port1_3 Switch", "PORT1_3_RX" },
+	{ "LSR Mixer", "Port1_4 Switch", "PORT1_4_RX" },
+	{ "LSR Mixer", "Port1_5 Switch", "PORT1_5_RX" },
+	{ "LSR Mixer", "Port1_6 Switch", "PORT1_6_RX" },
+	{ "LSR Mixer", "Port1_7 Switch", "PORT1_7_RX" },
+	{ "LSR Mixer", "Port1_8 Switch", "PORT1_8_RX" },
+
+	/* Port 2 */
+	{ "LSR Mixer", "Port2_1 Switch", "PORT2_1_RX" },
+	{ "LSR Mixer", "Port2_2 Switch", "PORT2_2_RX" },
+
+	{ "LSR Mixer", "ADCL Switch", "ADC Left" },
+	{ "LSR Mixer", "ADCR Switch", "ADC Right" },
+	{ "LSR Mixer", "DMIC1L Switch", "DMIC1 Left" },
+	{ "LSR Mixer", "DMIC1R Switch", "DMIC1 Right" },
+	{ "LSR Mixer", "DMIC2L Switch", "DMIC2 Left" },
+	{ "LSR Mixer", "DMIC2R Switch", "DMIC2 Right" },
+	{ "LSR Mixer", "Sidetone Switch", "Sidetone" },
+
+	{ "LSR DAC", NULL, "LSR Mixer" },
+
+	{ "LSOUTL", "Speaker Left Switch", "LSL DAC"},
+	{ "LSOUTR", "Speaker Left Switch", "LSR DAC"},
+
+	/* Haptic map */
+	{ "HAL Mixer", "Port1_1 Switch", "PORT1_1_RX" },
+	{ "HAL Mixer", "Port1_2 Switch", "PORT1_2_RX" },
+	{ "HAL Mixer", "Port1_3 Switch", "PORT1_3_RX" },
+	{ "HAL Mixer", "Port1_4 Switch", "PORT1_4_RX" },
+	{ "HAL Mixer", "Port1_5 Switch", "PORT1_5_RX" },
+	{ "HAL Mixer", "Port1_6 Switch", "PORT1_6_RX" },
+	{ "HAL Mixer", "Port1_7 Switch", "PORT1_7_RX" },
+	{ "HAL Mixer", "Port1_8 Switch", "PORT1_8_RX" },
+
+	/* Port 2 */
+	{ "HAL Mixer", "Port2_1 Switch", "PORT2_1_RX" },
+	{ "HAL Mixer", "Port2_2 Switch", "PORT2_2_RX" },
+
+	{ "HAL Mixer", "ADCL Switch", "ADC Left" },
+	{ "HAL Mixer", "ADCR Switch", "ADC Right" },
+	{ "HAL Mixer", "DMIC1L Switch", "DMIC1 Left" },
+	{ "HAL Mixer", "DMIC1R Switch", "DMIC1 Right" },
+	{ "HAL Mixer", "DMIC2L Switch", "DMIC2 Left" },
+	{ "HAL Mixer", "DMIC2R Switch", "DMIC2 Right" },
+	{ "HAL Mixer", "Sidetone Switch", "Sidetone" },
+
+	{ "HAL DAC", NULL, "HAL Mixer" },
+
+	{ "HAR Mixer", "Port1_1 Switch", "PORT1_1_RX" },
+	{ "HAR Mixer", "Port1_2 Switch", "PORT1_2_RX" },
+	{ "HAR Mixer", "Port1_3 Switch", "PORT1_3_RX" },
+	{ "HAR Mixer", "Port1_4 Switch", "PORT1_4_RX" },
+	{ "HAR Mixer", "Port1_5 Switch", "PORT1_5_RX" },
+	{ "HAR Mixer", "Port1_6 Switch", "PORT1_6_RX" },
+	{ "HAR Mixer", "Port1_7 Switch", "PORT1_7_RX" },
+	{ "HAR Mixer", "Port1_8 Switch", "PORT1_8_RX" },
+
+	/* Port 2 */
+	{ "HAR Mixer", "Port2_1 Switch", "PORT2_1_RX" },
+	{ "HAR Mixer", "Port2_2 Switch", "PORT2_2_RX" },
+
+	{ "HAR Mixer", "ADCL Switch", "ADC Left" },
+	{ "HAR Mixer", "ADCR Switch", "ADC Right" },
+	{ "HAR Mixer", "DMIC1L Switch", "DMIC1 Left" },
+	{ "HAR Mixer", "DMIC1R Switch", "DMIC1 Right" },
+	{ "HAR Mixer", "DMIC2L Switch", "DMIC2 Left" },
+	{ "HAR Mixer", "DMIC2R Switch", "DMIC2 Right" },
+	{ "HAR Mixer", "Sideton Switch", "Sidetone" },
+
+	{ "HAR DAC", NULL, "HAR Mixer" },
+
+	{ "HAOUTL", "Haptic Left Switch", "HAL DAC" },
+	{ "HAOUTR", "Haptic Right Switch", "HAR DAC" },
+
+	/* Lineout map */
+	{ "LOL Mixer", "Port1_1 Switch", "PORT1_1_RX" },
+	{ "LOL Mixer", "Port1_2 Switch", "PORT1_2_RX" },
+	{ "LOL Mixer", "Port1_3 Switch", "PORT1_3_RX" },
+	{ "LOL Mixer", "Port1_4 Switch", "PORT1_4_RX" },
+	{ "LOL Mixer", "Port1_5 Switch", "PORT1_5_RX" },
+	{ "LOL Mixer", "Port1_6 Switch", "PORT1_6_RX" },
+	{ "LOL Mixer", "Port1_7 Switch", "PORT1_7_RX" },
+	{ "LOL Mixer", "Port1_8 Switch", "PORT1_8_RX" },
+
+	/* Port 2 */
+	{ "LOL Mixer", "Port2_1 Switch", "PORT2_1_RX" },
+	{ "LOL Mixer", "Port2_2 Switch", "PORT2_2_RX" },
+
+	{ "LOL Mixer", "ADCL Switch", "ADC Left" },
+	{ "LOL Mixer", "ADCR Switch", "ADC Right" },
+	{ "LOL Mixer", "DMIC1L Switch", "DMIC1 Left" },
+	{ "LOL Mixer", "DMIC1R Switch", "DMIC1 Right" },
+	{ "LOL Mixer", "DMIC2L Switch", "DMIC2 Left" },
+	{ "LOL Mixer", "DMIC2R Switch", "DMIC2 Right" },
+	{ "LOL Mixer", "Sidetone Switch", "Sidetone" },
+
+	{ "LOL DAC", NULL, "LOL Mixer" },
+
+	{ "LOR Mixer", "Port1_1 Switch", "PORT1_1_RX" },
+	{ "LOR Mixer", "Port1_2 Switch", "PORT1_2_RX" },
+	{ "LOR Mixer", "Port1_3 Switch", "PORT1_3_RX" },
+	{ "LOR Mixer", "Port1_4 Switch", "PORT1_4_RX" },
+	{ "LOR Mixer", "Port1_5 Switch", "PORT1_5_RX" },
+	{ "LOR Mixer", "Port1_6 Switch", "PORT1_6_RX" },
+	{ "LOR Mixer", "Port1_7 Switch", "PORT1_7_RX" },
+	{ "LOR Mixer", "Port1_8 Switch", "PORT1_8_RX" },
+
+	/* Port 2 */
+	{ "LOR Mixer", "Port2_1 Switch", "PORT2_1_RX" },
+	{ "LOR Mixer", "Port2_2 Switch", "PORT2_2_RX" },
+
+	{ "LOR Mixer", "ADCL Switch", "ADC Left" },
+	{ "LOR Mixer", "ADCR Switch", "ADC Right" },
+	{ "LOR Mixer", "DMIC1L Switch", "DMIC1 Left" },
+	{ "LOR Mixer", "DMIC1R Switch", "DMIC1 Right" },
+	{ "LOR Mixer", "DMIC2L Switch", "DMIC2 Left" },
+	{ "LOR Mixer", "DMIC2R Switch", "DMIC2 Right" },
+	{ "LOR Mixer", "Sidetone Switch", "Sidetone" },
+
+	{ "LOR DAC", NULL, "LOR Mixer" },
+
+	{ "LOOUTL", NULL, "LOL DAC" },
+	{ "LOOUTR", NULL, "LOR DAC" },
+
+	/* TX map */
+	/* Port1 mappings */
+	{ "Port1_1 Mixer", "ADCL Switch", "ADC Left" },
+	{ "Port1_1 Mixer", "ADCR Switch", "ADC Right" },
+	{ "Port1_1 Mixer", "DMIC1L Switch", "DMIC1 Left" },
+	{ "Port1_1 Mixer", "DMIC1R Switch", "DMIC1 Right" },
+	{ "Port1_1 Mixer", "DMIC2L Switch", "DMIC2 Left" },
+	{ "Port1_1 Mixer", "DMIC2R Switch", "DMIC2 Right" },
+
+	{ "Port1_2 Mixer", "ADCL Switch", "ADC Left" },
+	{ "Port1_2 Mixer", "ADCR Switch", "ADC Right" },
+	{ "Port1_2 Mixer", "DMIC1L Switch", "DMIC1 Left" },
+	{ "Port1_2 Mixer", "DMIC1R Switch", "DMIC1 Right" },
+	{ "Port1_2 Mixer", "DMIC2L Switch", "DMIC2 Left" },
+	{ "Port1_2 Mixer", "DMIC2R Switch", "DMIC2 Right" },
+
+	{ "Port1_3 Mixer", "ADCL Switch", "ADC Left" },
+	{ "Port1_3 Mixer", "ADCR Switch", "ADC Right" },
+	{ "Port1_3 Mixer", "DMIC1L Switch", "DMIC1 Left" },
+	{ "Port1_3 Mixer", "DMIC1R Switch", "DMIC1 Right" },
+	{ "Port1_3 Mixer", "DMIC2L Switch", "DMIC2 Left" },
+	{ "Port1_3 Mixer", "DMIC2R Switch", "DMIC2 Right" },
+
+	{ "Port1_4 Mixer", "ADCL Switch", "ADC Left" },
+	{ "Port1_4 Mixer", "ADCR Switch", "ADC Right" },
+	{ "Port1_4 Mixer", "DMIC1L Switch", "DMIC1 Left" },
+	{ "Port1_4 Mixer", "DMIC1R Switch", "DMIC1 Right" },
+	{ "Port1_4 Mixer", "DMIC2L Switch", "DMIC2 Left" },
+	{ "Port1_4 Mixer", "DMIC2R Switch", "DMIC2 Right" },
+
+	{ "Port1_5 Mixer", "ADCL Switch", "ADC Left" },
+	{ "Port1_5 Mixer", "ADCR Switch", "ADC Right" },
+	{ "Port1_5 Mixer", "DMIC1L Switch", "DMIC1 Left" },
+	{ "Port1_5 Mixer", "DMIC1R Switch", "DMIC1 Right" },
+	{ "Port1_5 Mixer", "DMIC2L Switch", "DMIC2 Left" },
+	{ "Port1_5 Mixer", "DMIC2R Switch", "DMIC2 Right" },
+
+	{ "Port1_6 Mixer", "ADCL Switch", "ADC Left" },
+	{ "Port1_6 Mixer", "ADCR Switch", "ADC Right" },
+	{ "Port1_6 Mixer", "DMIC1L Switch", "DMIC1 Left" },
+	{ "Port1_6 Mixer", "DMIC1R Switch", "DMIC1 Right" },
+	{ "Port1_6 Mixer", "DMIC2L Switch", "DMIC2 Left" },
+	{ "Port1_6 Mixer", "DMIC2R Switch", "DMIC2 Right" },
+
+	{ "Port1_7 Mixer", "ADCL Switch", "ADC Left" },
+	{ "Port1_7 Mixer", "ADCR Switch", "ADC Right" },
+	{ "Port1_7 Mixer", "DMIC1L Switch", "DMIC1 Left" },
+	{ "Port1_7 Mixer", "DMIC1R Switch", "DMIC1 Right" },
+	{ "Port1_7 Mixer", "DMIC2L Switch", "DMIC2 Left" },
+	{ "Port1_7 Mixer", "DMIC2R Switch", "DMIC2 Right" },
+
+	{ "Port1_8 Mixer", "ADCL Switch", "ADC Left" },
+	{ "Port1_8 Mixer", "ADCR Switch", "ADC Right" },
+	{ "Port1_8 Mixer", "DMIC1L Switch", "DMIC1 Left" },
+	{ "Port1_8 Mixer", "DMIC1R Switch", "DMIC1 Right" },
+	{ "Port1_8 Mixer", "DMIC2L Switch", "DMIC2 Left" },
+	{ "Port1_8 Mixer", "DMIC2R Switch", "DMIC2 Right" },
+
+	{ "Port2_1 Mixer", "ADCL Switch", "ADC Left" },
+	{ "Port2_1 Mixer", "ADCR Switch", "ADC Right" },
+	{ "Port2_1 Mixer", "DMIC1L Switch", "DMIC1 Left" },
+	{ "Port2_1 Mixer", "DMIC1R Switch", "DMIC1 Right" },
+	{ "Port2_1 Mixer", "DMIC2L Switch", "DMIC2 Left" },
+	{ "Port2_1 Mixer", "DMIC2R Switch", "DMIC2 Right" },
+
+	{ "Port2_2 Mixer", "ADCL Switch", "ADC Left" },
+	{ "Port2_2 Mixer", "ADCR Switch", "ADC Right" },
+	{ "Port2_2 Mixer", "DMIC1L Switch", "DMIC1 Left" },
+	{ "Port2_2 Mixer", "DMIC1R Switch", "DMIC1 Right" },
+	{ "Port2_2 Mixer", "DMIC2L Switch", "DMIC2 Left" },
+	{ "Port2_2 Mixer", "DMIC2R Switch", "DMIC2 Right" },
+
+	{ "P1_1_TX", NULL, "Port1_1 Mixer" },
+	{ "P1_2_TX", NULL, "Port1_2 Mixer" },
+	{ "P1_3_TX", NULL, "Port1_3 Mixer" },
+	{ "P1_4_TX", NULL, "Port1_4 Mixer" },
+	{ "P1_5_TX", NULL, "Port1_5 Mixer" },
+	{ "P1_6_TX", NULL, "Port1_6 Mixer" },
+	{ "P1_7_TX", NULL, "Port1_7 Mixer" },
+	{ "P1_8_TX", NULL, "Port1_8 Mixer" },
+
+	{ "P2_1_TX", NULL, "Port2_1 Mixer" },
+	{ "P2_2_TX", NULL, "Port2_2 Mixer" },
+
+	{ "PORT1_SDO", "Port1 Capture Switch", "P1_1_TX"},
+	{ "PORT1_SDO", "Port1 Capture Switch", "P1_2_TX"},
+	{ "PORT1_SDO", "Port1 Capture Switch", "P1_3_TX"},
+	{ "PORT1_SDO", "Port1 Capture Switch", "P1_4_TX"},
+	{ "PORT1_SDO", "Port1 Capture Switch", "P1_5_TX"},
+	{ "PORT1_SDO", "Port1 Capture Switch", "P1_6_TX"},
+	{ "PORT1_SDO", "Port1 Capture Switch", "P1_7_TX"},
+	{ "PORT1_SDO", "Port1 Capture Switch", "P1_8_TX"},
+
+	{ "PORT2_SDO", "Port2 Capture Switch", "P2_1_TX"},
+	{ "PORT2_SDO", "Port2 Capture Switch", "P2_2_TX"},
+
+	{ "Mic1 Input", NULL, "AMIC1" },
+	{ "Mic2 Input", NULL, "AMIC2" },
+
+	{ "AUXL Input", NULL, "AUXL" },
+	{ "AUXR Input", NULL, "AUXR" },
+
+	/* AUX connections */
+	{ "ADCL Mux", "Aux_L", "AUXL Input" },
+	{ "ADCL Mux", "MIC1", "Mic1 Input" },
+
+	{ "ADCR Mux", "Aux_R", "AUXR Input" },
+	{ "ADCR Mux", "MIC2", "Mic2 Input" },
+
+	/* ADC connection */
+	{ "ADC Left", NULL, "ADCL Mux"},
+	{ "ADC Right", NULL, "ADCR Mux"},
+
+	{ "DMIC1 Left", NULL, "DMIC1DAT"},
+	{ "DMIC1 Right", NULL, "DMIC1DAT"},
+	{ "DMIC2 Left", NULL, "DMIC2DAT"},
+	{ "DMIC2 Right", NULL, "DMIC2DAT"},
+
+	/* Sidetone map */
+	{ "Sidetone Mixer", NULL, "ADC Left" },
+	{ "Sidetone Mixer", NULL, "ADC Right" },
+	{ "Sidetone Mixer", NULL, "DMIC1 Left" },
+	{ "Sidetone Mixer", NULL, "DMIC1 Right" },
+	{ "Sidetone Mixer", NULL, "DMIC2 Left" },
+	{ "Sidetone Mixer", NULL, "DMIC2 Right" },
+
+	{ "Sidetone", "Sidetone Switch", "Sidetone Mixer" },
+};
+
+static int lm49453_hw_params(struct snd_pcm_substream *substream,
+			     struct snd_pcm_hw_params *params,
+			     struct snd_soc_dai *dai)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	struct lm49453_priv *lm49453 = snd_soc_codec_get_drvdata(codec);
+	u16 clk_div = 0;
+
+	lm49453->fs_rate = params_rate(params);
+
+	/* Setting DAC clock dividers based on substream sample rate. */
+	switch (lm49453->fs_rate) {
+	case 8000:
+	case 16000:
+	case 32000:
+	case 24000:
+	case 48000:
+		clk_div = 256;
+		break;
+	case 11025:
+	case 22050:
+	case 44100:
+		clk_div = 216;
+		break;
+	case 96000:
+		clk_div = 127;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	snd_soc_write(codec, LM49453_P0_ADC_CLK_DIV_REG, clk_div);
+	snd_soc_write(codec, LM49453_P0_DAC_HP_CLK_DIV_REG, clk_div);
+
+	return 0;
+}
+
+static int lm49453_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+
+	u16 aif_val;
+	int mode = 0;
+	int clk_phase = 0;
+	int clk_shift = 0;
+
+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBS_CFS:
+		aif_val = 0;
+		break;
+	case SND_SOC_DAIFMT_CBS_CFM:
+		aif_val = LM49453_AUDIO_PORT1_BASIC_SYNC_MS;
+		break;
+	case SND_SOC_DAIFMT_CBM_CFS:
+		aif_val = LM49453_AUDIO_PORT1_BASIC_CLK_MS;
+		break;
+	case SND_SOC_DAIFMT_CBM_CFM:
+		aif_val = LM49453_AUDIO_PORT1_BASIC_CLK_MS |
+			  LM49453_AUDIO_PORT1_BASIC_SYNC_MS;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+		break;
+	case SND_SOC_DAIFMT_DSP_A:
+		mode = 1;
+		clk_phase = (1 << 5);
+		clk_shift = 1;
+		break;
+	case SND_SOC_DAIFMT_DSP_B:
+		mode = 1;
+		clk_phase = (1 << 5);
+		clk_shift = 0;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	snd_soc_update_bits(codec, LM49453_P0_AUDIO_PORT1_BASIC_REG,
+			    LM49453_AUDIO_PORT1_BASIC_FMT_MASK|BIT(1)|BIT(5),
+			    (aif_val | mode | clk_phase));
+
+	snd_soc_write(codec, LM49453_P0_AUDIO_PORT1_RX_MSB_REG, clk_shift);
+
+	return 0;
+}
+
+static int lm49453_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id,
+				  unsigned int freq, int dir)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	u16 pll_clk = 0;
+
+	switch (freq) {
+	case 12288000:
+	case 26000000:
+	case 19200000:
+		/* pll clk slection */
+		pll_clk = 0;
+		break;
+	case 48000:
+	case 32576:
+		/* fll clk slection */
+		pll_clk = BIT(4);
+		return 0;
+	default:
+		return -EINVAL;
+	}
+
+	snd_soc_update_bits(codec, LM49453_P0_PMC_SETUP_REG, BIT(4), pll_clk);
+
+	return 0;
+}
+
+static int lm49453_hp_mute(struct snd_soc_dai *dai, int mute)
+{
+	snd_soc_update_bits(dai->codec, LM49453_P0_DAC_DSP_REG, BIT(1)|BIT(0),
+			    (mute ? (BIT(1)|BIT(0)) : 0));
+	return 0;
+}
+
+static int lm49453_lo_mute(struct snd_soc_dai *dai, int mute)
+{
+	snd_soc_update_bits(dai->codec, LM49453_P0_DAC_DSP_REG, BIT(3)|BIT(2),
+			    (mute ? (BIT(3)|BIT(2)) : 0));
+	return 0;
+}
+
+static int lm49453_ls_mute(struct snd_soc_dai *dai, int mute)
+{
+	snd_soc_update_bits(dai->codec, LM49453_P0_DAC_DSP_REG, BIT(5)|BIT(4),
+			    (mute ? (BIT(5)|BIT(4)) : 0));
+	return 0;
+}
+
+static int lm49453_ep_mute(struct snd_soc_dai *dai, int mute)
+{
+	snd_soc_update_bits(dai->codec, LM49453_P0_DAC_DSP_REG, BIT(4),
+			    (mute ? BIT(4) : 0));
+	return 0;
+}
+
+static int lm49453_ha_mute(struct snd_soc_dai *dai, int mute)
+{
+	snd_soc_update_bits(dai->codec, LM49453_P0_DAC_DSP_REG, BIT(7)|BIT(6),
+			    (mute ? (BIT(7)|BIT(6)) : 0));
+	return 0;
+}
+
+static int lm49453_set_bias_level(struct snd_soc_codec *codec,
+				  enum snd_soc_bias_level level)
+{
+	struct lm49453_priv *lm49453 = snd_soc_codec_get_drvdata(codec);
+
+	switch (level) {
+	case SND_SOC_BIAS_ON:
+	case SND_SOC_BIAS_PREPARE:
+		break;
+
+	case SND_SOC_BIAS_STANDBY:
+		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF)
+			regcache_sync(lm49453->regmap);
+
+		snd_soc_update_bits(codec, LM49453_P0_PMC_SETUP_REG,
+				    LM49453_PMC_SETUP_CHIP_EN, LM49453_CHIP_EN);
+		break;
+
+	case SND_SOC_BIAS_OFF:
+		snd_soc_update_bits(codec, LM49453_P0_PMC_SETUP_REG,
+				    LM49453_PMC_SETUP_CHIP_EN, 0);
+		break;
+	}
+
+	codec->dapm.bias_level = level;
+
+	return 0;
+}
+
+/* Formates supported by LM49453 driver. */
+#define LM49453_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
+			 SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE)
+
+static struct snd_soc_dai_ops lm49453_headset_dai_ops = {
+	.hw_params	= lm49453_hw_params,
+	.set_sysclk	= lm49453_set_dai_sysclk,
+	.set_fmt	= lm49453_set_dai_fmt,
+	.digital_mute	= lm49453_hp_mute,
+};
+
+static struct snd_soc_dai_ops lm49453_speaker_dai_ops = {
+	.hw_params	= lm49453_hw_params,
+	.set_sysclk	= lm49453_set_dai_sysclk,
+	.set_fmt	= lm49453_set_dai_fmt,
+	.digital_mute	= lm49453_ls_mute,
+};
+
+static struct snd_soc_dai_ops lm49453_haptic_dai_ops = {
+	.hw_params	= lm49453_hw_params,
+	.set_sysclk	= lm49453_set_dai_sysclk,
+	.set_fmt	= lm49453_set_dai_fmt,
+	.digital_mute	= lm49453_ha_mute,
+};
+
+static struct snd_soc_dai_ops lm49453_ep_dai_ops = {
+	.hw_params	= lm49453_hw_params,
+	.set_sysclk	= lm49453_set_dai_sysclk,
+	.set_fmt	= lm49453_set_dai_fmt,
+	.digital_mute	= lm49453_ep_mute,
+};
+
+static struct snd_soc_dai_ops lm49453_lineout_dai_ops = {
+	.hw_params	= lm49453_hw_params,
+	.set_sysclk	= lm49453_set_dai_sysclk,
+	.set_fmt	= lm49453_set_dai_fmt,
+	.digital_mute	= lm49453_lo_mute,
+};
+
+/* LM49453 dai structure. */
+static const struct snd_soc_dai_driver lm49453_dai[] = {
+	{
+		.name = "LM49453 Headset",
+		.playback = {
+			.stream_name = "Headset",
+			.channels_min = 2,
+			.channels_max = 2,
+			.rates = SNDRV_PCM_RATE_8000_192000,
+			.formats = LM49453_FORMATS,
+		},
+		.capture = {
+			.stream_name = "Capture",
+			.channels_min = 1,
+			.channels_max = 5,
+			.rates = SNDRV_PCM_RATE_8000_192000,
+			.formats = LM49453_FORMATS,
+		},
+		.ops = &lm49453_headset_dai_ops,
+		.symmetric_rates = 1,
+	},
+	{
+		.name = "LM49453 Speaker",
+		.playback = {
+			.stream_name = "Speaker",
+			.channels_min = 2,
+			.channels_max = 2,
+			.rates = SNDRV_PCM_RATE_8000_192000,
+			.formats = LM49453_FORMATS,
+		},
+		.ops = &lm49453_speaker_dai_ops,
+	},
+	{
+		.name = "LM49453 Haptic",
+		.playback = {
+			.stream_name = "Haptic",
+			.channels_min = 2,
+			.channels_max = 2,
+			.rates = SNDRV_PCM_RATE_8000_192000,
+			.formats = LM49453_FORMATS,
+		},
+		.ops = &lm49453_haptic_dai_ops,
+	},
+	{
+		.name = "LM49453 Earpiece",
+		.playback = {
+			.stream_name = "Earpiece",
+			.channels_min = 1,
+			.channels_max = 1,
+			.rates = SNDRV_PCM_RATE_8000_192000,
+			.formats = LM49453_FORMATS,
+		},
+		.ops = &lm49453_ep_dai_ops,
+	},
+	{
+		.name = "LM49453 line out",
+		.playback = {
+			.stream_name = "Lineout",
+			.channels_min = 2,
+			.channels_max = 2,
+			.rates = SNDRV_PCM_RATE_8000_192000,
+			.formats = LM49453_FORMATS,
+		},
+		.ops = &lm49453_lineout_dai_ops,
+	},
+};
+
+static int lm49453_suspend(struct snd_soc_codec *codec)
+{
+	lm49453_set_bias_level(codec, SND_SOC_BIAS_OFF);
+	return 0;
+}
+
+static int lm49453_resume(struct snd_soc_codec *codec)
+{
+	lm49453_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+	return 0;
+}
+
+static int lm49453_probe(struct snd_soc_codec *codec)
+{
+	struct lm49453_priv *lm49453 = snd_soc_codec_get_drvdata(codec);
+	int ret = 0;
+
+	codec->control_data = lm49453->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;
+	}
+
+	return 0;
+}
+
+/* power down chip */
+static int lm49453_remove(struct snd_soc_codec *codec)
+{
+	lm49453_set_bias_level(codec, SND_SOC_BIAS_OFF);
+	return 0;
+}
+
+static struct snd_soc_codec_driver soc_codec_dev_lm49453 = {
+	.probe = lm49453_probe,
+	.remove = lm49453_remove,
+	.suspend = lm49453_suspend,
+	.resume = lm49453_resume,
+	.set_bias_level = lm49453_set_bias_level,
+	.controls = lm49453_snd_controls,
+	.num_controls = ARRAY_SIZE(lm49453_snd_controls),
+	.dapm_widgets = lm49453_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(lm49453_dapm_widgets),
+	.dapm_routes = lm49453_audio_map,
+	.num_dapm_routes = ARRAY_SIZE(lm49453_audio_map),
+	.idle_bias_off = true,
+};
+
+static const struct regmap_config lm49453_regmap_config = {
+	.reg_bits = 8,
+	.val_bits = 8,
+
+	.max_register = LM49453_MAX_REGISTER,
+	.reg_defaults = lm49453_reg_defs,
+	.num_reg_defaults = ARRAY_SIZE(lm49453_reg_defs),
+	.cache_type = REGCACHE_RBTREE,
+};
+
+static __devinit int lm49453_i2c_probe(struct i2c_client *i2c,
+				       const struct i2c_device_id *id)
+{
+	struct lm49453_priv *lm49453;
+	int ret = 0;
+
+	lm49453 = devm_kzalloc(&i2c->dev, sizeof(struct lm49453_priv),
+				GFP_KERNEL);
+
+	if (lm49453 == NULL)
+		return -ENOMEM;
+
+	i2c_set_clientdata(i2c, lm49453);
+
+	lm49453->regmap = regmap_init_i2c(i2c, &lm49453_regmap_config);
+	if (IS_ERR(lm49453->regmap)) {
+		ret = PTR_ERR(lm49453->regmap);
+		dev_err(&i2c->dev, "Failed to allocate register map: %d\n",
+			ret);
+		return ret;
+	}
+
+	ret =  snd_soc_register_codec(&i2c->dev,
+				      &soc_codec_dev_lm49453,
+				      lm49453_dai, ARRAY_SIZE(lm49453_dai));
+	if (ret < 0) {
+		dev_err(&i2c->dev, "Failed to register codec: %d\n", ret);
+		regmap_exit(lm49453->regmap);
+		return ret;
+	}
+
+	return ret;
+}
+
+static int __devexit lm49453_i2c_remove(struct i2c_client *client)
+{
+	struct lm49453_priv *lm49453 = i2c_get_clientdata(client);
+
+	snd_soc_unregister_codec(&client->dev);
+	regmap_exit(lm49453->regmap);
+	return 0;
+}
+
+static const struct i2c_device_id lm49453_i2c_id[] = {
+	{ "lm49453", 0 },
+	{ }
+};
+MODULE_DEVICE_TABLE(i2c, lm49453_i2c_id);
+
+static struct i2c_driver lm49453_i2c_driver = {
+	.driver = {
+		.name = "lm49453",
+		.owner = THIS_MODULE,
+	},
+	.probe = lm49453_i2c_probe,
+	.remove = __devexit_p(lm49453_i2c_remove),
+	.id_table = lm49453_i2c_id,
+};
+
+module_i2c_driver(lm49453_i2c_driver);
+
+MODULE_DESCRIPTION("ASoC LM49453 driver");
+MODULE_AUTHOR("M R Swami Reddy <MR.Swami.Reddy@ti.com>");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/codecs/lm49453.h b/sound/soc/codecs/lm49453.h
new file mode 100644
index 000000000000..a63cfa5c0883
--- /dev/null
+++ b/sound/soc/codecs/lm49453.h
@@ -0,0 +1,380 @@
+/*
+ * lm49453.h  -  LM49453 ALSA Soc Audio drive
+ *
+ * Copyright (c) 2012  Texas Instruments, Inc
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; version 2 of the License.
+ *
+ */
+
+#ifndef _LM49453_H
+#define _LM49453_H
+
+#include <linux/bitops.h>
+
+/* LM49453_P0 register space for page0 */
+#define LM49453_P0_PMC_SETUP_REG			0x00
+#define LM49453_P0_PLL_CLK_SEL1_REG			0x01
+#define LM49453_P0_PLL_CLK_SEL2_REG			0x02
+#define LM49453_P0_PMC_CLK_DIV_REG			0x03
+#define LM49453_P0_HSDET_CLK_DIV_REG			0x04
+#define LM49453_P0_DMIC_CLK_DIV_REG			0x05
+#define LM49453_P0_ADC_CLK_DIV_REG			0x06
+#define LM49453_P0_DAC_OT_CLK_DIV_REG			0x07
+#define LM49453_P0_PLL_HF_M_REG				0x08
+#define LM49453_P0_PLL_LF_M_REG				0x09
+#define LM49453_P0_PLL_NL_REG				0x0A
+#define LM49453_P0_PLL_N_MODL_REG			0x0B
+#define LM49453_P0_PLL_N_MODH_REG			0x0C
+#define LM49453_P0_PLL_P1_REG				0x0D
+#define LM49453_P0_PLL_P2_REG				0x0E
+#define LM49453_P0_FLL_REF_FREQL_REG			0x0F
+#define LM49453_P0_FLL_REF_FREQH_REG			0x10
+#define LM49453_P0_VCO_TARGETLL_REG			0x11
+#define LM49453_P0_VCO_TARGETLH_REG			0x12
+#define LM49453_P0_VCO_TARGETHL_REG			0x13
+#define LM49453_P0_VCO_TARGETHH_REG			0x14
+#define LM49453_P0_PLL_CONFIG_REG			0x15
+#define LM49453_P0_DAC_CLK_SEL_REG			0x16
+#define LM49453_P0_DAC_HP_CLK_DIV_REG			0x17
+
+/* Analog Mixer Input Stages */
+#define LM49453_P0_MICL_REG				0x20
+#define LM49453_P0_MICR_REG				0x21
+#define LM49453_P0_EP_REG				0x24
+#define LM49453_P0_DIS_PKVL_FB_REG			0x25
+
+/* Analog Mixer Output Stages */
+#define LM49453_P0_ANALOG_MIXER_ADC_REG			0x2E
+
+/*ADC or DAC */
+#define LM49453_P0_ADC_DSP_REG				0x30
+#define LM49453_P0_DAC_DSP_REG				0x31
+
+/* EFFECTS ENABLES */
+#define LM49453_P0_ADC_FX_ENABLES_REG			0x33
+
+/* GPIO */
+#define LM49453_P0_GPIO1_REG				0x38
+#define LM49453_P0_GPIO2_REG				0x39
+#define LM49453_P0_GPIO3_REG				0x3A
+#define LM49453_P0_HAP_CTL_REG				0x3B
+#define LM49453_P0_HAP_FREQ_PROG_LEFTL_REG		0x3C
+#define LM49453_P0_HAP_FREQ_PROG_LEFTH_REG		0x3D
+#define LM49453_P0_HAP_FREQ_PROG_RIGHTL_REG		0x3E
+#define LM49453_P0_HAP_FREQ_PROG_RIGHTH_REG		0x3F
+
+/* DIGITAL MIXER */
+#define LM49453_P0_DMIX_CLK_SEL_REG			0x40
+#define LM49453_P0_PORT1_RX_LVL1_REG			0x41
+#define LM49453_P0_PORT1_RX_LVL2_REG			0x42
+#define LM49453_P0_PORT2_RX_LVL_REG			0x43
+#define LM49453_P0_PORT1_TX1_REG			0x44
+#define LM49453_P0_PORT1_TX2_REG			0x45
+#define LM49453_P0_PORT1_TX3_REG			0x46
+#define LM49453_P0_PORT1_TX4_REG			0x47
+#define LM49453_P0_PORT1_TX5_REG			0x48
+#define LM49453_P0_PORT1_TX6_REG			0x49
+#define LM49453_P0_PORT1_TX7_REG			0x4A
+#define LM49453_P0_PORT1_TX8_REG			0x4B
+#define LM49453_P0_PORT2_TX1_REG			0x4C
+#define LM49453_P0_PORT2_TX2_REG			0x4D
+#define LM49453_P0_STN_SEL_REG				0x4F
+#define LM49453_P0_DACHPL1_REG				0x50
+#define LM49453_P0_DACHPL2_REG				0x51
+#define LM49453_P0_DACHPR1_REG				0x52
+#define LM49453_P0_DACHPR2_REG				0x53
+#define LM49453_P0_DACLOL1_REG				0x54
+#define LM49453_P0_DACLOL2_REG				0x55
+#define LM49453_P0_DACLOR1_REG				0x56
+#define LM49453_P0_DACLOR2_REG				0x57
+#define LM49453_P0_DACLSL1_REG				0x58
+#define LM49453_P0_DACLSL2_REG				0x59
+#define LM49453_P0_DACLSR1_REG				0x5A
+#define LM49453_P0_DACLSR2_REG				0x5B
+#define LM49453_P0_DACHAL1_REG				0x5C
+#define LM49453_P0_DACHAL2_REG				0x5D
+#define LM49453_P0_DACHAR1_REG				0x5E
+#define LM49453_P0_DACHAR2_REG				0x5F
+
+/* AUDIO PORT 1 (TDM) */
+#define LM49453_P0_AUDIO_PORT1_BASIC_REG		0x60
+#define LM49453_P0_AUDIO_PORT1_CLK_GEN1_REG		0x61
+#define LM49453_P0_AUDIO_PORT1_CLK_GEN2_REG		0x62
+#define LM49453_P0_AUDIO_PORT1_CLK_GEN3_REG		0x63
+#define LM49453_P0_AUDIO_PORT1_SYNC_RATE_REG		0x64
+#define LM49453_P0_AUDIO_PORT1_SYNC_SDO_SETUP_REG	0x65
+#define LM49453_P0_AUDIO_PORT1_DATA_WIDTH_REG		0x66
+#define LM49453_P0_AUDIO_PORT1_RX_MSB_REG		0x67
+#define LM49453_P0_AUDIO_PORT1_TX_MSB_REG		0x68
+#define LM49453_P0_AUDIO_PORT1_TDM_CHANNELS_REG		0x69
+
+/* AUDIO PORT 2 */
+#define LM49453_P0_AUDIO_PORT2_BASIC_REG		0x6A
+#define LM49453_P0_AUDIO_PORT2_CLK_GEN1_REG		0x6B
+#define LM49453_P0_AUDIO_PORT2_CLK_GEN2_REG		0x6C
+#define LM49453_P0_AUDIO_PORT2_SYNC_GEN_REG		0x6D
+#define LM49453_P0_AUDIO_PORT2_DATA_WIDTH_REG		0x6E
+#define LM49453_P0_AUDIO_PORT2_RX_MODE_REG		0x6F
+#define LM49453_P0_AUDIO_PORT2_TX_MODE_REG		0x70
+
+/* SAMPLE RATE */
+#define LM49453_P0_PORT1_SR_LSB_REG			0x79
+#define LM49453_P0_PORT1_SR_MSB_REG			0x7A
+#define LM49453_P0_PORT2_SR_LSB_REG			0x7B
+#define LM49453_P0_PORT2_SR_MSB_REG			0x7C
+
+/* EFFECTS - HPFs */
+#define LM49453_P0_HPF_REG				0x80
+
+/* EFFECTS ADC ALC */
+#define LM49453_P0_ADC_ALC1_REG				0x82
+#define LM49453_P0_ADC_ALC2_REG				0x83
+#define LM49453_P0_ADC_ALC3_REG				0x84
+#define LM49453_P0_ADC_ALC4_REG				0x85
+#define LM49453_P0_ADC_ALC5_REG				0x86
+#define LM49453_P0_ADC_ALC6_REG				0x87
+#define LM49453_P0_ADC_ALC7_REG				0x88
+#define LM49453_P0_ADC_ALC8_REG				0x89
+#define LM49453_P0_DMIC1_LEVELL_REG			0x8A
+#define LM49453_P0_DMIC1_LEVELR_REG			0x8B
+#define LM49453_P0_DMIC2_LEVELL_REG			0x8C
+#define LM49453_P0_DMIC2_LEVELR_REG			0x8D
+#define LM49453_P0_ADC_LEVELL_REG			0x8E
+#define LM49453_P0_ADC_LEVELR_REG			0x8F
+#define LM49453_P0_DAC_HP_LEVELL_REG			0x90
+#define LM49453_P0_DAC_HP_LEVELR_REG			0x91
+#define LM49453_P0_DAC_LO_LEVELL_REG			0x92
+#define LM49453_P0_DAC_LO_LEVELR_REG			0x93
+#define LM49453_P0_DAC_LS_LEVELL_REG			0x94
+#define LM49453_P0_DAC_LS_LEVELR_REG			0x95
+#define LM49453_P0_DAC_HA_LEVELL_REG			0x96
+#define LM49453_P0_DAC_HA_LEVELR_REG			0x97
+#define LM49453_P0_SOFT_MUTE_REG			0x98
+#define LM49453_P0_DMIC_MUTE_CFG_REG			0x99
+#define LM49453_P0_ADC_MUTE_CFG_REG			0x9A
+#define LM49453_P0_DAC_MUTE_CFG_REG			0x9B
+
+/*DIGITAL MIC1 */
+#define LM49453_P0_DIGITAL_MIC1_CONFIG_REG		0xB0
+#define LM49453_P0_DIGITAL_MIC1_DATA_DELAYL_REG		0xB1
+#define LM49453_P0_DIGITAL_MIC1_DATA_DELAYR_REG		0xB2
+
+/*DIGITAL MIC2 */
+#define LM49453_P0_DIGITAL_MIC2_CONFIG_REG		0xB3
+#define LM49453_P0_DIGITAL_MIC2_DATA_DELAYL_REG		0xB4
+#define LM49453_P0_DIGITAL_MIC2_DATA_DELAYR_REG		0xB5
+
+/* ADC DECIMATOR */
+#define LM49453_P0_ADC_DECIMATOR_REG			0xB6
+
+/* DAC CONFIGURE */
+#define LM49453_P0_DAC_CONFIG_REG			0xB7
+
+/* SIDETONE */
+#define LM49453_P0_STN_VOL_ADCL_REG			0xB8
+#define LM49453_P0_STN_VOL_ADCR_REG			0xB9
+#define LM49453_P0_STN_VOL_DMIC1L_REG			0xBA
+#define LM49453_P0_STN_VOL_DMIC1R_REG			0xBB
+#define LM49453_P0_STN_VOL_DMIC2L_REG			0xBC
+#define LM49453_P0_STN_VOL_DMIC2R_REG			0xBD
+
+/* ADC/DAC CLIPPING MONITORS (Read Only/Write to Clear) */
+#define LM49453_P0_ADC_DEC_CLIP_REG			0xC2
+#define LM49453_P0_ADC_HPF_CLIP_REG			0xC3
+#define LM49453_P0_ADC_LVL_CLIP_REG			0xC4
+#define LM49453_P0_DAC_LVL_CLIP_REG			0xC5
+
+/* ADC ALC EFFECT MONITORS (Read Only) */
+#define LM49453_P0_ADC_LVLMONL_REG			0xC8
+#define LM49453_P0_ADC_LVLMONR_REG			0xC9
+#define LM49453_P0_ADC_ALCMONL_REG			0xCA
+#define LM49453_P0_ADC_ALCMONR_REG			0xCB
+#define LM49453_P0_ADC_MUTED_REG			0xCC
+#define LM49453_P0_DAC_MUTED_REG			0xCD
+
+/* HEADSET DETECT */
+#define LM49453_P0_HSD_PPB_LONG_CNT_LIMITL_REG		0xD0
+#define LM49453_P0_HSD_PPB_LONG_CNT_LIMITR_REG		0xD1
+#define LM49453_P0_HSD_PIN3_4_EX_LOOP_CNT_LIMITL_REG	0xD2
+#define LM49453_P0_HSD_PIN3_4_EX_LOOP_CNT_LIMITH_REG	0xD3
+#define LM49453_P0_HSD_TIMEOUT1_REG			0xD4
+#define LM49453_P0_HSD_TIMEOUT2_REG			0xD5
+#define LM49453_P0_HSD_TIMEOUT3_REG			0xD6
+#define LM49453_P0_HSD_PIN3_4_CFG_REG			0xD7
+#define LM49453_P0_HSD_IRQ1_REG				0xD8
+#define LM49453_P0_HSD_IRQ2_REG				0xD9
+#define LM49453_P0_HSD_IRQ3_REG				0xDA
+#define LM49453_P0_HSD_IRQ4_REG				0xDB
+#define LM49453_P0_HSD_IRQ_MASK1_REG			0xDC
+#define LM49453_P0_HSD_IRQ_MASK2_REG			0xDD
+#define LM49453_P0_HSD_IRQ_MASK3_REG			0xDE
+#define LM49453_P0_HSD_R_HPLL_REG			0xE0
+#define LM49453_P0_HSD_R_HPLH_REG			0xE1
+#define LM49453_P0_HSD_R_HPLU_REG			0xE2
+#define LM49453_P0_HSD_R_HPRL_REG			0xE3
+#define LM49453_P0_HSD_R_HPRH_REG			0xE4
+#define LM49453_P0_HSD_R_HPRU_REG			0xE5
+#define LM49453_P0_HSD_VEL_L_FINALL_REG			0xE6
+#define LM49453_P0_HSD_VEL_L_FINALH_REG			0xE7
+#define LM49453_P0_HSD_VEL_L_FINALU_REG			0xE8
+#define LM49453_P0_HSD_RO_FINALL_REG			0xE9
+#define LM49453_P0_HSD_RO_FINALH_REG			0xEA
+#define LM49453_P0_HSD_RO_FINALU_REG			0xEB
+#define LM49453_P0_HSD_VMIC_BIAS_FINALL_REG		0xEC
+#define LM49453_P0_HSD_VMIC_BIAS_FINALH_REG		0xED
+#define LM49453_P0_HSD_VMIC_BIAS_FINALU_REG		0xEE
+#define LM49453_P0_HSD_PIN_CONFIG_REG			0xEF
+#define LM49453_P0_HSD_PLUG_DETECT_BB_IRQ_STATUS1_REG	0xF1
+#define LM49453_P0_HSD_PLUG_DETECT_BB_IRQ_STATUS2_REG	0xF2
+#define LM49453_P0_HSD_PLUG_DETECT_BB_IRQ_STATUS3_REG	0xF3
+#define LM49453_P0_HSD_PLUG_DETECT_BB_IRQ_STATEL_REG	0xF4
+#define LM49453_P0_HSD_PLUG_DETECT_BB_IRQ_STATEH_REG	0xF5
+
+/* I/O PULLDOWN CONFIG */
+#define LM49453_P0_PULL_CONFIG1_REG			0xF8
+#define LM49453_P0_PULL_CONFIG2_REG			0xF9
+#define LM49453_P0_PULL_CONFIG3_REG			0xFA
+
+/* RESET */
+#define LM49453_P0_RESET_REG				0xFE
+
+/* PAGE */
+#define LM49453_PAGE_REG				0xFF
+
+#define LM49453_MAX_REGISTER				(0xFF+1)
+
+/* LM49453_P0_PMC_SETUP_REG (0x00h) */
+#define LM49453_PMC_SETUP_CHIP_EN			(BIT(1)|BIT(0))
+#define LM49453_PMC_SETUP_PLL_EN			BIT(2)
+#define LM49453_PMC_SETUP_PLL_P2_EN			BIT(3)
+#define LM49453_PMC_SETUP_PLL_FLL			BIT(4)
+#define LM49453_PMC_SETUP_MCLK_OVER			BIT(5)
+#define LM49453_PMC_SETUP_RTC_CLK_OVER			BIT(6)
+#define LM49453_PMC_SETUP_CHIP_ACTIVE			BIT(7)
+
+/* Chip Enable bits */
+#define LM49453_CHIP_EN_SHUTDOWN			0x00
+#define LM49453_CHIP_EN					0x01
+#define LM49453_CHIP_EN_HSD_DETECT			0x02
+#define LM49453_CHIP_EN_INVALID_HSD			0x03
+
+/* LM49453_P0_PLL_CLK_SEL1_REG (0x01h) */
+#define LM49453_CLK_SEL1_MCLK_SEL			0x11
+#define LM49453_CLK_SEL1_RTC_SEL			0x11
+#define LM49453_CLK_SEL1_PORT1_SEL			0x10
+#define LM49453_CLK_SEL1_PORT2_SEL			0x11
+
+/* LM49453_P0_PLL_CLK_SEL2_REG (0x02h) */
+#define LM49453_CLK_SEL2_ADC_CLK_SEL			0x38
+
+/* LM49453_P0_FLL_REF_FREQL_REG (0x0F) */
+#define LM49453_FLL_REF_FREQ_VAL			0x8ca0001
+
+/* LM49453_P0_VCO_TARGETLL_REG (0x11) */
+#define LM49453_VCO_TARGET_VAL				0x8ca0001
+
+/* LM49453_P0_ADC_DSP_REG (0x30h) */
+#define LM49453_ADC_DSP_ADC_MUTEL			BIT(0)
+#define LM49453_ADC_DSP_ADC_MUTER			BIT(1)
+#define LM49453_ADC_DSP_DMIC1_MUTEL			BIT(2)
+#define LM49453_ADC_DSP_DMIC1_MUTER			BIT(3)
+#define LM49453_ADC_DSP_DMIC2_MUTEL			BIT(4)
+#define LM49453_ADC_DSP_DMIC2_MUTER			BIT(5)
+#define LM49453_ADC_DSP_MUTE_ALL			0x3F
+
+/* LM49453_P0_DAC_DSP_REG (0x31h) */
+#define LM49453_DAC_DSP_MUTE_ALL			0xFF
+
+/* LM49453_P0_AUDIO_PORT1_BASIC_REG (0x60h) */
+#define LM49453_AUDIO_PORT1_BASIC_FMT_MASK		(BIT(4)|BIT(3))
+#define LM49453_AUDIO_PORT1_BASIC_CLK_MS		BIT(3)
+#define LM49453_AUDIO_PORT1_BASIC_SYNC_MS		BIT(4)
+
+/* LM49453_P0_RESET_REG (0xFEh) */
+#define LM49453_RESET_REG_RST				BIT(0)
+
+/* Page select register bits (0xFF) */
+#define LM49453_PAGE0_SELECT				0x0
+#define LM49453_PAGE1_SELECT				0x1
+
+/* LM49453_P0_HSD_PIN3_4_CFG_REG (Jack Pin config - 0xD7) */
+#define LM49453_JACK_DISABLE				0x00
+#define LM49453_JACK_CONFIG1				0x01
+#define LM49453_JACK_CONFIG2				0x02
+#define LM49453_JACK_CONFIG3				0x03
+#define LM49453_JACK_CONFIG4				0x04
+#define LM49453_JACK_CONFIG5				0x05
+
+/* Page 1 REGISTERS */
+
+/* SIDETONE */
+#define LM49453_P1_SIDETONE_SA0L_REG			0x80
+#define LM49453_P1_SIDETONE_SA0H_REG			0x81
+#define LM49453_P1_SIDETONE_SAB0U_REG			0x82
+#define LM49453_P1_SIDETONE_SB0L_REG			0x83
+#define LM49453_P1_SIDETONE_SB0H_REG			0x84
+#define LM49453_P1_SIDETONE_SH0L_REG			0x85
+#define LM49453_P1_SIDETONE_SH0H_REG			0x86
+#define LM49453_P1_SIDETONE_SH0U_REG			0x87
+#define LM49453_P1_SIDETONE_SA1L_REG			0x88
+#define LM49453_P1_SIDETONE_SA1H_REG			0x89
+#define LM49453_P1_SIDETONE_SAB1U_REG			0x8A
+#define LM49453_P1_SIDETONE_SB1L_REG			0x8B
+#define LM49453_P1_SIDETONE_SB1H_REG			0x8C
+#define LM49453_P1_SIDETONE_SH1L_REG			0x8D
+#define LM49453_P1_SIDETONE_SH1H_REG			0x8E
+#define LM49453_P1_SIDETONE_SH1U_REG			0x8F
+#define LM49453_P1_SIDETONE_SA2L_REG			0x90
+#define LM49453_P1_SIDETONE_SA2H_REG			0x91
+#define LM49453_P1_SIDETONE_SAB2U_REG			0x92
+#define LM49453_P1_SIDETONE_SB2L_REG			0x93
+#define LM49453_P1_SIDETONE_SB2H_REG			0x94
+#define LM49453_P1_SIDETONE_SH2L_REG			0x95
+#define LM49453_P1_SIDETONE_SH2H_REG			0x96
+#define LM49453_P1_SIDETONE_SH2U_REG			0x97
+#define LM49453_P1_SIDETONE_SA3L_REG			0x98
+#define LM49453_P1_SIDETONE_SA3H_REG			0x99
+#define LM49453_P1_SIDETONE_SAB3U_REG			0x9A
+#define LM49453_P1_SIDETONE_SB3L_REG			0x9B
+#define LM49453_P1_SIDETONE_SB3H_REG			0x9C
+#define LM49453_P1_SIDETONE_SH3L_REG			0x9D
+#define LM49453_P1_SIDETONE_SH3H_REG			0x9E
+#define LM49453_P1_SIDETONE_SH3U_REG			0x9F
+#define LM49453_P1_SIDETONE_SA4L_REG			0xA0
+#define LM49453_P1_SIDETONE_SA4H_REG			0xA1
+#define LM49453_P1_SIDETONE_SAB4U_REG			0xA2
+#define LM49453_P1_SIDETONE_SB4L_REG			0xA3
+#define LM49453_P1_SIDETONE_SB4H_REG			0xA4
+#define LM49453_P1_SIDETONE_SH4L_REG			0xA5
+#define LM49453_P1_SIDETONE_SH4H_REG			0xA6
+#define LM49453_P1_SIDETONE_SH4U_REG			0xA7
+#define LM49453_P1_SIDETONE_SA5L_REG			0xA8
+#define LM49453_P1_SIDETONE_SA5H_REG			0xA9
+#define LM49453_P1_SIDETONE_SAB5U_REG			0xAA
+#define LM49453_P1_SIDETONE_SB5L_REG			0xAB
+#define LM49453_P1_SIDETONE_SB5H_REG			0xAC
+#define LM49453_P1_SIDETONE_SH5L_REG			0xAD
+#define LM49453_P1_SIDETONE_SH5H_REG			0xAE
+#define LM49453_P1_SIDETONE_SH5U_REG			0xAF
+
+/* CHARGE PUMP CONFIG */
+#define LM49453_P1_CP_CONFIG1_REG			0xB0
+#define LM49453_P1_CP_CONFIG2_REG			0xB1
+#define LM49453_P1_CP_CONFIG3_REG			0xB2
+#define LM49453_P1_CP_CONFIG4_REG			0xB3
+#define LM49453_P1_CP_LA_VTH1L_REG			0xB4
+#define LM49453_P1_CP_LA_VTH1M_REG			0xB5
+#define LM49453_P1_CP_LA_VTH2L_REG			0xB6
+#define LM49453_P1_CP_LA_VTH2M_REG			0xB7
+#define LM49453_P1_CP_LA_VTH3L_REG			0xB8
+#define LM49453_P1_CP_LA_VTH3H_REG			0xB9
+#define LM49453_P1_CP_CLK_DIV_REG			0xBA
+
+/* DAC */
+#define LM49453_P1_DAC_CHOP_REG				0xC0
+
+#define	LM49453_CLK_SRC_MCLK				1
+#endif
diff --git a/sound/soc/codecs/max98095.c b/sound/soc/codecs/max98095.c
index 0bb511a0388d..35179e2c23c9 100644
--- a/sound/soc/codecs/max98095.c
+++ b/sound/soc/codecs/max98095.c
@@ -24,6 +24,7 @@
 #include <linux/slab.h>
 #include <asm/div64.h>
 #include <sound/max98095.h>
+#include <sound/jack.h>
 #include "max98095.h"
 
 enum max98095_type {
@@ -51,6 +52,8 @@ struct max98095_priv {
 	u8 lin_state;
 	unsigned int mic1pre;
 	unsigned int mic2pre;
+	struct snd_soc_jack *headphone_jack;
+	struct snd_soc_jack *mic_jack;
 };
 
 static const u8 max98095_reg_def[M98095_REG_CNT] = {
@@ -2173,9 +2176,125 @@ static void max98095_handle_pdata(struct snd_soc_codec *codec)
 		max98095_handle_bq_pdata(codec);
 }
 
+static irqreturn_t max98095_report_jack(int irq, void *data)
+{
+	struct snd_soc_codec *codec = data;
+	struct max98095_priv *max98095 = snd_soc_codec_get_drvdata(codec);
+	unsigned int value;
+	int hp_report = 0;
+	int mic_report = 0;
+
+	/* Read the Jack Status Register */
+	value = snd_soc_read(codec, M98095_007_JACK_AUTO_STS);
+
+	/* If ddone is not set, then detection isn't finished yet */
+	if ((value & M98095_DDONE) == 0)
+		return IRQ_NONE;
+
+	/* if hp, check its bit, and if set, clear it */
+	if ((value & M98095_HP_IN || value & M98095_LO_IN) &&
+		max98095->headphone_jack)
+		hp_report |= SND_JACK_HEADPHONE;
+
+	/* if mic, check its bit, and if set, clear it */
+	if ((value & M98095_MIC_IN) && max98095->mic_jack)
+		mic_report |= SND_JACK_MICROPHONE;
+
+	if (max98095->headphone_jack == max98095->mic_jack) {
+		snd_soc_jack_report(max98095->headphone_jack,
+					hp_report | mic_report,
+					SND_JACK_HEADSET);
+	} else {
+		if (max98095->headphone_jack)
+			snd_soc_jack_report(max98095->headphone_jack,
+					hp_report, SND_JACK_HEADPHONE);
+		if (max98095->mic_jack)
+			snd_soc_jack_report(max98095->mic_jack,
+					mic_report, SND_JACK_MICROPHONE);
+	}
+
+	return IRQ_HANDLED;
+}
+
+int max98095_jack_detect_enable(struct snd_soc_codec *codec)
+{
+	struct max98095_priv *max98095 = snd_soc_codec_get_drvdata(codec);
+	int ret = 0;
+	int detect_enable = M98095_JDEN;
+	unsigned int slew = M98095_DEFAULT_SLEW_DELAY;
+
+	if (max98095->pdata->jack_detect_pin5en)
+		detect_enable |= M98095_PIN5EN;
+
+	if (max98095->pdata->jack_detect_delay)
+		slew = max98095->pdata->jack_detect_delay;
+
+	ret = snd_soc_write(codec, M98095_08E_JACK_DC_SLEW, slew);
+	if (ret < 0) {
+		dev_err(codec->dev, "Failed to cfg auto detect %d\n", ret);
+		return ret;
+	}
+
+	/* configure auto detection to be enabled */
+	ret = snd_soc_write(codec, M98095_089_JACK_DET_AUTO, detect_enable);
+	if (ret < 0) {
+		dev_err(codec->dev, "Failed to cfg auto detect %d\n", ret);
+		return ret;
+	}
+
+	return ret;
+}
+
+int max98095_jack_detect_disable(struct snd_soc_codec *codec)
+{
+	int ret = 0;
+
+	/* configure auto detection to be disabled */
+	ret = snd_soc_write(codec, M98095_089_JACK_DET_AUTO, 0x0);
+	if (ret < 0) {
+		dev_err(codec->dev, "Failed to cfg auto detect %d\n", ret);
+		return ret;
+	}
+
+	return ret;
+}
+
+int max98095_jack_detect(struct snd_soc_codec *codec,
+	struct snd_soc_jack *hp_jack, struct snd_soc_jack *mic_jack)
+{
+	struct max98095_priv *max98095 = snd_soc_codec_get_drvdata(codec);
+	struct i2c_client *client = to_i2c_client(codec->dev);
+	int ret = 0;
+
+	max98095->headphone_jack = hp_jack;
+	max98095->mic_jack = mic_jack;
+
+	/* only progress if we have at least 1 jack pointer */
+	if (!hp_jack && !mic_jack)
+		return -EINVAL;
+
+	max98095_jack_detect_enable(codec);
+
+	/* enable interrupts for headphone jack detection */
+	ret = snd_soc_update_bits(codec, M98095_013_JACK_INT_EN,
+		M98095_IDDONE, M98095_IDDONE);
+	if (ret < 0) {
+		dev_err(codec->dev, "Failed to cfg jack irqs %d\n", ret);
+		return ret;
+	}
+
+	max98095_report_jack(client->irq, codec);
+	return 0;
+}
+
 #ifdef CONFIG_PM
 static int max98095_suspend(struct snd_soc_codec *codec)
 {
+	struct max98095_priv *max98095 = snd_soc_codec_get_drvdata(codec);
+
+	if (max98095->headphone_jack || max98095->mic_jack)
+		max98095_jack_detect_disable(codec);
+
 	max98095_set_bias_level(codec, SND_SOC_BIAS_OFF);
 
 	return 0;
@@ -2183,8 +2302,16 @@ static int max98095_suspend(struct snd_soc_codec *codec)
 
 static int max98095_resume(struct snd_soc_codec *codec)
 {
+	struct max98095_priv *max98095 = snd_soc_codec_get_drvdata(codec);
+	struct i2c_client *client = to_i2c_client(codec->dev);
+
 	max98095_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 
+	if (max98095->headphone_jack || max98095->mic_jack) {
+		max98095_jack_detect_enable(codec);
+		max98095_report_jack(client->irq, codec);
+	}
+
 	return 0;
 }
 #else
@@ -2227,6 +2354,7 @@ static int max98095_probe(struct snd_soc_codec *codec)
 {
 	struct max98095_priv *max98095 = snd_soc_codec_get_drvdata(codec);
 	struct max98095_cdata *cdata;
+	struct i2c_client *client;
 	int ret = 0;
 
 	ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_I2C);
@@ -2238,6 +2366,8 @@ static int max98095_probe(struct snd_soc_codec *codec)
 	/* reset the codec, the DSP core, and disable all interrupts */
 	max98095_reset(codec);
 
+	client = to_i2c_client(codec->dev);
+
 	/* initialize private data */
 
 	max98095->sysclk = (unsigned)-1;
@@ -2266,11 +2396,23 @@ static int max98095_probe(struct snd_soc_codec *codec)
 	max98095->mic1pre = 0;
 	max98095->mic2pre = 0;
 
+	if (client->irq) {
+		/* register an audio interrupt */
+		ret = request_threaded_irq(client->irq, NULL,
+			max98095_report_jack,
+			IRQF_TRIGGER_FALLING | IRQF_TRIGGER_RISING,
+			"max98095", codec);
+		if (ret) {
+			dev_err(codec->dev, "Failed to request IRQ: %d\n", ret);
+			goto err_access;
+		}
+	}
+
 	ret = snd_soc_read(codec, M98095_0FF_REV_ID);
 	if (ret < 0) {
 		dev_err(codec->dev, "Failure reading hardware revision: %d\n",
 			ret);
-		goto err_access;
+		goto err_irq;
 	}
 	dev_info(codec->dev, "Hardware revision: %c\n", ret - 0x40 + 'A');
 
@@ -2306,14 +2448,28 @@ static int max98095_probe(struct snd_soc_codec *codec)
 
 	max98095_add_widgets(codec);
 
+	return 0;
+
+err_irq:
+	if (client->irq)
+		free_irq(client->irq, codec);
 err_access:
 	return ret;
 }
 
 static int max98095_remove(struct snd_soc_codec *codec)
 {
+	struct max98095_priv *max98095 = snd_soc_codec_get_drvdata(codec);
+	struct i2c_client *client = to_i2c_client(codec->dev);
+
 	max98095_set_bias_level(codec, SND_SOC_BIAS_OFF);
 
+	if (max98095->headphone_jack || max98095->mic_jack)
+		max98095_jack_detect_disable(codec);
+
+	if (client->irq)
+		free_irq(client->irq, codec);
+
 	return 0;
 }
 
diff --git a/sound/soc/codecs/max98095.h b/sound/soc/codecs/max98095.h
index 891584a0eb03..2ebbe4e894bf 100644
--- a/sound/soc/codecs/max98095.h
+++ b/sound/soc/codecs/max98095.h
@@ -175,11 +175,23 @@
 
 /* MAX98095 Registers Bit Fields */
 
+/* M98095_007_JACK_AUTO_STS */
+	#define M98095_MIC_IN			(1<<3)
+	#define M98095_LO_IN			(1<<5)
+	#define M98095_HP_IN			(1<<6)
+	#define M98095_DDONE			(1<<7)
+
 /* M98095_00F_HOST_CFG */
 	#define M98095_SEG                      (1<<0)
 	#define M98095_XTEN                     (1<<1)
 	#define M98095_MDLLEN                   (1<<2)
 
+/* M98095_013_JACK_INT_EN */
+	#define M98095_IMIC_IN			(1<<3)
+	#define M98095_ILO_IN			(1<<5)
+	#define M98095_IHP_IN			(1<<6)
+	#define M98095_IDDONE			(1<<7)
+
 /* M98095_027_DAI1_CLKMODE, M98095_031_DAI2_CLKMODE, M98095_03B_DAI3_CLKMODE */
 	#define M98095_CLKMODE_MASK             0xFF
 
@@ -255,6 +267,10 @@
 	#define M98095_EQ2EN                    (1<<1)
 	#define M98095_EQ1EN                    (1<<0)
 
+/* M98095_089_JACK_DET_AUTO */
+	#define M98095_PIN5EN			(1<<2)
+	#define M98095_JDEN			(1<<7)
+
 /* M98095_090_PWR_EN_IN */
 	#define M98095_INEN                     (1<<7)
 	#define M98095_MB2EN                    (1<<3)
@@ -296,4 +312,10 @@
 #define M98095_174_DAI1_BQ_BASE             0x74
 #define M98095_17E_DAI2_BQ_BASE             0x7E
 
+/* Default Delay used in Slew Rate Calculation for Jack detection */
+#define M98095_DEFAULT_SLEW_DELAY		0x18
+
+extern int max98095_jack_detect(struct snd_soc_codec *codec,
+	struct snd_soc_jack *hp_jack, struct snd_soc_jack *mic_jack);
+
 #endif
diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c
new file mode 100644
index 000000000000..6276e352125f
--- /dev/null
+++ b/sound/soc/codecs/mc13783.c
@@ -0,0 +1,786 @@
+/*
+ * Copyright 2008 Juergen Beisert, kernel@pengutronix.de
+ * Copyright 2009 Sascha Hauer, s.hauer@pengutronix.de
+ * Copyright 2012 Philippe Retornaz, philippe.retornaz@epfl.ch
+ *
+ * Initial development of this code was funded by
+ * Phytec Messtechnik GmbH, http://www.phytec.de
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * as published by the Free Software Foundation; either version 2
+ * of the License, or (at your option) any later version.
+ * 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 Street, Fifth Floor, Boston,
+ * MA  02110-1301, USA.
+ */
+#include <linux/module.h>
+#include <linux/device.h>
+#include <linux/mfd/mc13xxx.h>
+#include <linux/slab.h>
+#include <sound/core.h>
+#include <sound/control.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/initval.h>
+#include <sound/soc-dapm.h>
+
+#include "mc13783.h"
+
+#define MC13783_AUDIO_RX0	36
+#define MC13783_AUDIO_RX1	37
+#define MC13783_AUDIO_TX	38
+#define MC13783_SSI_NETWORK	39
+#define MC13783_AUDIO_CODEC	40
+#define MC13783_AUDIO_DAC	41
+
+#define AUDIO_RX0_ALSPEN		(1 << 5)
+#define AUDIO_RX0_ALSPSEL		(1 << 7)
+#define AUDIO_RX0_ADDCDC		(1 << 21)
+#define AUDIO_RX0_ADDSTDC		(1 << 22)
+#define AUDIO_RX0_ADDRXIN		(1 << 23)
+
+#define AUDIO_RX1_PGARXEN		(1 << 0);
+#define AUDIO_RX1_PGASTEN		(1 << 5)
+#define AUDIO_RX1_ARXINEN		(1 << 10)
+
+#define AUDIO_TX_AMC1REN		(1 << 5)
+#define AUDIO_TX_AMC1LEN		(1 << 7)
+#define AUDIO_TX_AMC2EN			(1 << 9)
+#define AUDIO_TX_ATXINEN		(1 << 11)
+#define AUDIO_TX_RXINREC		(1 << 13)
+
+#define SSI_NETWORK_CDCTXRXSLOT(x)	(((x) & 0x3) << 2)
+#define SSI_NETWORK_CDCTXSECSLOT(x)	(((x) & 0x3) << 4)
+#define SSI_NETWORK_CDCRXSECSLOT(x)	(((x) & 0x3) << 6)
+#define SSI_NETWORK_CDCRXSECGAIN(x)	(((x) & 0x3) << 8)
+#define SSI_NETWORK_CDCSUMGAIN(x)	(1 << 10)
+#define SSI_NETWORK_CDCFSDLY(x)		(1 << 11)
+#define SSI_NETWORK_DAC_SLOTS_8		(1 << 12)
+#define SSI_NETWORK_DAC_SLOTS_4		(2 << 12)
+#define SSI_NETWORK_DAC_SLOTS_2		(3 << 12)
+#define SSI_NETWORK_DAC_SLOT_MASK	(3 << 12)
+#define SSI_NETWORK_DAC_RXSLOT_0_1	(0 << 14)
+#define SSI_NETWORK_DAC_RXSLOT_2_3	(1 << 14)
+#define SSI_NETWORK_DAC_RXSLOT_4_5	(2 << 14)
+#define SSI_NETWORK_DAC_RXSLOT_6_7	(3 << 14)
+#define SSI_NETWORK_DAC_RXSLOT_MASK	(3 << 14)
+#define SSI_NETWORK_STDCRXSECSLOT(x)	(((x) & 0x3) << 16)
+#define SSI_NETWORK_STDCRXSECGAIN(x)	(((x) & 0x3) << 18)
+#define SSI_NETWORK_STDCSUMGAIN		(1 << 20)
+
+/*
+ * MC13783_AUDIO_CODEC and MC13783_AUDIO_DAC mostly share the same
+ * register layout
+ */
+#define AUDIO_SSI_SEL			(1 << 0)
+#define AUDIO_CLK_SEL			(1 << 1)
+#define AUDIO_CSM			(1 << 2)
+#define AUDIO_BCL_INV			(1 << 3)
+#define AUDIO_CFS_INV			(1 << 4)
+#define AUDIO_CFS(x)			(((x) & 0x3) << 5)
+#define AUDIO_CLK(x)			(((x) & 0x7) << 7)
+#define AUDIO_C_EN			(1 << 11)
+#define AUDIO_C_CLK_EN			(1 << 12)
+#define AUDIO_C_RESET			(1 << 15)
+
+#define AUDIO_CODEC_CDCFS8K16K		(1 << 10)
+#define AUDIO_DAC_CFS_DLY_B		(1 << 10)
+
+struct mc13783_priv {
+	struct snd_soc_codec codec;
+	struct mc13xxx *mc13xxx;
+
+	enum mc13783_ssi_port adc_ssi_port;
+	enum mc13783_ssi_port dac_ssi_port;
+};
+
+static unsigned int mc13783_read(struct snd_soc_codec *codec,
+	unsigned int reg)
+{
+	struct mc13783_priv *priv = snd_soc_codec_get_drvdata(codec);
+	unsigned int value = 0;
+
+	mc13xxx_lock(priv->mc13xxx);
+
+	mc13xxx_reg_read(priv->mc13xxx, reg, &value);
+
+	mc13xxx_unlock(priv->mc13xxx);
+
+	return value;
+}
+
+static int mc13783_write(struct snd_soc_codec *codec,
+	unsigned int reg, unsigned int value)
+{
+	struct mc13783_priv *priv = snd_soc_codec_get_drvdata(codec);
+	int ret;
+
+	mc13xxx_lock(priv->mc13xxx);
+
+	ret = mc13xxx_reg_write(priv->mc13xxx, reg, value);
+
+	mc13xxx_unlock(priv->mc13xxx);
+
+	return ret;
+}
+
+/* Mapping between sample rates and register value */
+static unsigned int mc13783_rates[] = {
+	8000, 11025, 12000, 16000,
+	22050, 24000, 32000, 44100,
+	48000, 64000, 96000
+};
+
+static int mc13783_pcm_hw_params_dac(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;
+	unsigned int rate = params_rate(params);
+	int i;
+
+	for (i = 0; i < ARRAY_SIZE(mc13783_rates); i++) {
+		if (rate == mc13783_rates[i]) {
+			snd_soc_update_bits(codec, MC13783_AUDIO_DAC,
+					0xf << 17, i << 17);
+			return 0;
+		}
+	}
+
+	return -EINVAL;
+}
+
+static int mc13783_pcm_hw_params_codec(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;
+	unsigned int rate = params_rate(params);
+	unsigned int val;
+
+	switch (rate) {
+	case 8000:
+		val = 0;
+		break;
+	case 16000:
+		val = AUDIO_CODEC_CDCFS8K16K;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	snd_soc_update_bits(codec, MC13783_AUDIO_CODEC, AUDIO_CODEC_CDCFS8K16K,
+			val);
+
+	return 0;
+}
+
+static int mc13783_pcm_hw_params_sync(struct snd_pcm_substream *substream,
+				struct snd_pcm_hw_params *params,
+				struct snd_soc_dai *dai)
+{
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+		return mc13783_pcm_hw_params_dac(substream, params, dai);
+	else
+		return mc13783_pcm_hw_params_codec(substream, params, dai);
+}
+
+static int mc13783_set_fmt(struct snd_soc_dai *dai, unsigned int fmt,
+			unsigned int reg)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	unsigned int val = 0;
+	unsigned int mask = AUDIO_CFS(3) | AUDIO_BCL_INV | AUDIO_CFS_INV |
+				AUDIO_CSM | AUDIO_C_CLK_EN | AUDIO_C_RESET;
+
+
+	/* DAI mode */
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+		val |= AUDIO_CFS(2);
+		break;
+	case SND_SOC_DAIFMT_DSP_A:
+		val |= AUDIO_CFS(1);
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	/* DAI clock inversion */
+	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+	case SND_SOC_DAIFMT_NB_NF:
+		val |= AUDIO_BCL_INV;
+		break;
+	case SND_SOC_DAIFMT_NB_IF:
+		val |= AUDIO_BCL_INV | AUDIO_CFS_INV;
+		break;
+	case SND_SOC_DAIFMT_IB_NF:
+		break;
+	case SND_SOC_DAIFMT_IB_IF:
+		val |= AUDIO_CFS_INV;
+		break;
+	}
+
+	/* DAI clock master masks */
+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBM_CFM:
+		val |= AUDIO_C_CLK_EN;
+		break;
+	case SND_SOC_DAIFMT_CBS_CFS:
+		val |= AUDIO_CSM;
+		break;
+	case SND_SOC_DAIFMT_CBM_CFS:
+	case SND_SOC_DAIFMT_CBS_CFM:
+		return -EINVAL;
+	}
+
+	val |= AUDIO_C_RESET;
+
+	snd_soc_update_bits(codec, reg, mask, val);
+
+	return 0;
+}
+
+static int mc13783_set_fmt_async(struct snd_soc_dai *dai, unsigned int fmt)
+{
+	if (dai->id == MC13783_ID_STEREO_DAC)
+		return mc13783_set_fmt(dai, fmt, MC13783_AUDIO_DAC);
+	else
+		return mc13783_set_fmt(dai, fmt, MC13783_AUDIO_CODEC);
+}
+
+static int mc13783_set_fmt_sync(struct snd_soc_dai *dai, unsigned int fmt)
+{
+	int ret;
+
+	ret = mc13783_set_fmt(dai, fmt, MC13783_AUDIO_DAC);
+	if (ret)
+		return ret;
+
+	/*
+	 * In synchronous mode force the voice codec into slave mode
+	 * so that the clock / framesync from the stereo DAC is used
+	 */
+	fmt &= ~SND_SOC_DAIFMT_MASTER_MASK;
+	fmt |= SND_SOC_DAIFMT_CBS_CFS;
+	ret = mc13783_set_fmt(dai, fmt, MC13783_AUDIO_CODEC);
+
+	return ret;
+}
+
+static int mc13783_sysclk[] = {
+	13000000,
+	15360000,
+	16800000,
+	-1,
+	26000000,
+	-1, /* 12000000, invalid for voice codec */
+	-1, /* 3686400, invalid for voice codec */
+	33600000,
+};
+
+static int mc13783_set_sysclk(struct snd_soc_dai *dai,
+				  int clk_id, unsigned int freq, int dir,
+				  unsigned int reg)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	int clk;
+	unsigned int val = 0;
+	unsigned int mask = AUDIO_CLK(0x7) | AUDIO_CLK_SEL;
+
+	for (clk = 0; clk < ARRAY_SIZE(mc13783_sysclk); clk++) {
+		if (mc13783_sysclk[clk] < 0)
+			continue;
+		if (mc13783_sysclk[clk] == freq)
+			break;
+	}
+
+	if (clk == ARRAY_SIZE(mc13783_sysclk))
+		return -EINVAL;
+
+	if (clk_id == MC13783_CLK_CLIB)
+		val |= AUDIO_CLK_SEL;
+
+	val |= AUDIO_CLK(clk);
+
+	snd_soc_update_bits(codec, reg, mask, val);
+
+	return 0;
+}
+
+static int mc13783_set_sysclk_dac(struct snd_soc_dai *dai,
+				  int clk_id, unsigned int freq, int dir)
+{
+	return mc13783_set_sysclk(dai, clk_id, freq, dir, MC13783_AUDIO_DAC);
+}
+
+static int mc13783_set_sysclk_codec(struct snd_soc_dai *dai,
+				  int clk_id, unsigned int freq, int dir)
+{
+	return mc13783_set_sysclk(dai, clk_id, freq, dir, MC13783_AUDIO_CODEC);
+}
+
+static int mc13783_set_sysclk_sync(struct snd_soc_dai *dai,
+				  int clk_id, unsigned int freq, int dir)
+{
+	int ret;
+
+	ret = mc13783_set_sysclk(dai, clk_id, freq, dir, MC13783_AUDIO_DAC);
+	if (ret)
+		return ret;
+
+	return mc13783_set_sysclk(dai, clk_id, freq, dir, MC13783_AUDIO_CODEC);
+}
+
+static int mc13783_set_tdm_slot_dac(struct snd_soc_dai *dai,
+	unsigned int tx_mask, unsigned int rx_mask, int slots,
+	int slot_width)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	unsigned int val = 0;
+	unsigned int mask = SSI_NETWORK_DAC_SLOT_MASK |
+				SSI_NETWORK_DAC_RXSLOT_MASK;
+
+	switch (slots) {
+	case 2:
+		val |= SSI_NETWORK_DAC_SLOTS_2;
+		break;
+	case 4:
+		val |= SSI_NETWORK_DAC_SLOTS_4;
+		break;
+	case 8:
+		val |= SSI_NETWORK_DAC_SLOTS_8;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	switch (rx_mask) {
+	case 0xfffffffc:
+		val |= SSI_NETWORK_DAC_RXSLOT_0_1;
+		break;
+	case 0xfffffff3:
+		val |= SSI_NETWORK_DAC_RXSLOT_2_3;
+		break;
+	case 0xffffffcf:
+		val |= SSI_NETWORK_DAC_RXSLOT_4_5;
+		break;
+	case 0xffffff3f:
+		val |= SSI_NETWORK_DAC_RXSLOT_6_7;
+		break;
+	default:
+		return -EINVAL;
+	};
+
+	snd_soc_update_bits(codec, MC13783_SSI_NETWORK, mask, val);
+
+	return 0;
+}
+
+static int mc13783_set_tdm_slot_codec(struct snd_soc_dai *dai,
+	unsigned int tx_mask, unsigned int rx_mask, int slots,
+	int slot_width)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	unsigned int val = 0;
+	unsigned int mask = 0x3f;
+
+	if (slots != 4)
+		return -EINVAL;
+
+	if (tx_mask != 0xfffffffc)
+		return -EINVAL;
+
+	val |= (0x00 << 2);	/* primary timeslot RX/TX(?) is 0 */
+	val |= (0x01 << 4);	/* secondary timeslot TX is 1 */
+
+	snd_soc_update_bits(codec, MC13783_SSI_NETWORK, mask, val);
+
+	return 0;
+}
+
+static int mc13783_set_tdm_slot_sync(struct snd_soc_dai *dai,
+	unsigned int tx_mask, unsigned int rx_mask, int slots,
+	int slot_width)
+{
+	int ret;
+
+	ret = mc13783_set_tdm_slot_dac(dai, tx_mask, rx_mask, slots,
+			slot_width);
+	if (ret)
+		return ret;
+
+	ret = mc13783_set_tdm_slot_codec(dai, tx_mask, rx_mask, slots,
+			slot_width);
+
+	return ret;
+}
+
+static const struct snd_kcontrol_new mc1l_amp_ctl =
+	SOC_DAPM_SINGLE("Switch", 38, 7, 1, 0);
+
+static const struct snd_kcontrol_new mc1r_amp_ctl =
+	SOC_DAPM_SINGLE("Switch", 38, 5, 1, 0);
+
+static const struct snd_kcontrol_new mc2_amp_ctl =
+	SOC_DAPM_SINGLE("Switch", 38, 9, 1, 0);
+
+static const struct snd_kcontrol_new atx_amp_ctl =
+	SOC_DAPM_SINGLE("Switch", 38, 11, 1, 0);
+
+
+/* Virtual mux. The chip does the input selection automatically
+ * as soon as we enable one input. */
+static const char * const adcl_enum_text[] = {
+	"MC1L", "RXINL",
+};
+
+static const struct soc_enum adcl_enum =
+	SOC_ENUM_SINGLE(0, 0, ARRAY_SIZE(adcl_enum_text), adcl_enum_text);
+
+static const struct snd_kcontrol_new left_input_mux =
+	SOC_DAPM_ENUM_VIRT("Route", adcl_enum);
+
+static const char * const adcr_enum_text[] = {
+	"MC1R", "MC2", "RXINR", "TXIN",
+};
+
+static const struct soc_enum adcr_enum =
+	SOC_ENUM_SINGLE(0, 0, ARRAY_SIZE(adcr_enum_text), adcr_enum_text);
+
+static const struct snd_kcontrol_new right_input_mux =
+	SOC_DAPM_ENUM_VIRT("Route", adcr_enum);
+
+static const struct snd_kcontrol_new samp_ctl =
+	SOC_DAPM_SINGLE("Switch", 36, 3, 1, 0);
+
+static const struct snd_kcontrol_new lamp_ctl =
+	SOC_DAPM_SINGLE("Switch", 36, 5, 1, 0);
+
+static const struct snd_kcontrol_new hlamp_ctl =
+	SOC_DAPM_SINGLE("Switch", 36, 10, 1, 0);
+
+static const struct snd_kcontrol_new hramp_ctl =
+	SOC_DAPM_SINGLE("Switch", 36, 9, 1, 0);
+
+static const struct snd_kcontrol_new llamp_ctl =
+	SOC_DAPM_SINGLE("Switch", 36, 16, 1, 0);
+
+static const struct snd_kcontrol_new lramp_ctl =
+	SOC_DAPM_SINGLE("Switch", 36, 15, 1, 0);
+
+static const struct snd_soc_dapm_widget mc13783_dapm_widgets[] = {
+/* Input */
+	SND_SOC_DAPM_INPUT("MC1LIN"),
+	SND_SOC_DAPM_INPUT("MC1RIN"),
+	SND_SOC_DAPM_INPUT("MC2IN"),
+	SND_SOC_DAPM_INPUT("RXINR"),
+	SND_SOC_DAPM_INPUT("RXINL"),
+	SND_SOC_DAPM_INPUT("TXIN"),
+
+	SND_SOC_DAPM_SUPPLY("MC1 Bias", 38, 0, 0, NULL, 0),
+	SND_SOC_DAPM_SUPPLY("MC2 Bias", 38, 1, 0, NULL, 0),
+
+	SND_SOC_DAPM_SWITCH("MC1L Amp", 38, 7, 0, &mc1l_amp_ctl),
+	SND_SOC_DAPM_SWITCH("MC1R Amp", 38, 5, 0, &mc1r_amp_ctl),
+	SND_SOC_DAPM_SWITCH("MC2 Amp", 38, 9, 0, &mc2_amp_ctl),
+	SND_SOC_DAPM_SWITCH("TXIN Amp", 38, 11, 0, &atx_amp_ctl),
+
+	SND_SOC_DAPM_VIRT_MUX("PGA Left Input Mux", SND_SOC_NOPM, 0, 0,
+			      &left_input_mux),
+	SND_SOC_DAPM_VIRT_MUX("PGA Right Input Mux", SND_SOC_NOPM, 0, 0,
+			      &right_input_mux),
+
+	SND_SOC_DAPM_PGA("PGA Left Input", SND_SOC_NOPM, 0, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("PGA Right Input", SND_SOC_NOPM, 0, 0, NULL, 0),
+
+	SND_SOC_DAPM_ADC("ADC", "Capture", 40, 11, 0),
+	SND_SOC_DAPM_SUPPLY("ADC_Reset", 40, 15, 0, NULL, 0),
+
+/* Output */
+	SND_SOC_DAPM_SUPPLY("DAC_E", 41, 11, 0, NULL, 0),
+	SND_SOC_DAPM_SUPPLY("DAC_Reset", 41, 15, 0, NULL, 0),
+	SND_SOC_DAPM_OUTPUT("RXOUTL"),
+	SND_SOC_DAPM_OUTPUT("RXOUTR"),
+	SND_SOC_DAPM_OUTPUT("HSL"),
+	SND_SOC_DAPM_OUTPUT("HSR"),
+	SND_SOC_DAPM_OUTPUT("LSP"),
+	SND_SOC_DAPM_OUTPUT("SP"),
+
+	SND_SOC_DAPM_SWITCH("Speaker Amp", 36, 3, 0, &samp_ctl),
+	SND_SOC_DAPM_SWITCH("Loudspeaker Amp", SND_SOC_NOPM, 0, 0, &lamp_ctl),
+	SND_SOC_DAPM_SWITCH("Headset Amp Left", 36, 10, 0, &hlamp_ctl),
+	SND_SOC_DAPM_SWITCH("Headset Amp Right", 36, 9, 0, &hramp_ctl),
+	SND_SOC_DAPM_SWITCH("Line out Amp Left", 36, 16, 0, &llamp_ctl),
+	SND_SOC_DAPM_SWITCH("Line out Amp Right", 36, 15, 0, &lramp_ctl),
+	SND_SOC_DAPM_DAC("DAC", "Playback", 36, 22, 0),
+	SND_SOC_DAPM_PGA("DAC PGA", 37, 5, 0, NULL, 0),
+};
+
+static struct snd_soc_dapm_route mc13783_routes[] = {
+/* Input */
+	{ "MC1L Amp", NULL, "MC1LIN"},
+	{ "MC1R Amp", NULL, "MC1RIN" },
+	{ "MC2 Amp", NULL, "MC2IN" },
+	{ "TXIN Amp", NULL, "TXIN"},
+
+	{ "PGA Left Input Mux", "MC1L", "MC1L Amp" },
+	{ "PGA Left Input Mux", "RXINL", "RXINL"},
+	{ "PGA Right Input Mux", "MC1R", "MC1R Amp" },
+	{ "PGA Right Input Mux", "MC2",  "MC2 Amp"},
+	{ "PGA Right Input Mux", "TXIN", "TXIN Amp"},
+	{ "PGA Right Input Mux", "RXINR", "RXINR"},
+
+	{ "PGA Left Input", NULL, "PGA Left Input Mux"},
+	{ "PGA Right Input", NULL, "PGA Right Input Mux"},
+
+	{ "ADC", NULL, "PGA Left Input"},
+	{ "ADC", NULL, "PGA Right Input"},
+	{ "ADC", NULL, "ADC_Reset"},
+
+/* Output */
+	{ "HSL", NULL, "Headset Amp Left" },
+	{ "HSR", NULL, "Headset Amp Right"},
+	{ "RXOUTL", NULL, "Line out Amp Left"},
+	{ "RXOUTR", NULL, "Line out Amp Right"},
+	{ "SP", NULL, "Speaker Amp"},
+	{ "Speaker Amp", NULL, "DAC PGA"},
+	{ "LSP", NULL, "DAC PGA"},
+	{ "Headset Amp Left", NULL, "DAC PGA"},
+	{ "Headset Amp Right", NULL, "DAC PGA"},
+	{ "Line out Amp Left", NULL, "DAC PGA"},
+	{ "Line out Amp Right", NULL, "DAC PGA"},
+	{ "DAC PGA", NULL, "DAC"},
+	{ "DAC", NULL, "DAC_E"},
+};
+
+static const char * const mc13783_3d_mixer[] = {"Stereo", "Phase Mix",
+						"Mono", "Mono Mix"};
+
+static const struct soc_enum mc13783_enum_3d_mixer =
+	SOC_ENUM_SINGLE(MC13783_AUDIO_RX1, 16, ARRAY_SIZE(mc13783_3d_mixer),
+			mc13783_3d_mixer);
+
+static struct snd_kcontrol_new mc13783_control_list[] = {
+	SOC_SINGLE("Loudspeaker enable", MC13783_AUDIO_RX0, 5, 1, 0),
+	SOC_SINGLE("PCM Playback Volume", MC13783_AUDIO_RX1, 6, 15, 0),
+	SOC_DOUBLE("PCM Capture Volume", MC13783_AUDIO_TX, 19, 14, 31, 0),
+	SOC_ENUM("3D Control", mc13783_enum_3d_mixer),
+};
+
+static int mc13783_probe(struct snd_soc_codec *codec)
+{
+	struct mc13783_priv *priv = snd_soc_codec_get_drvdata(codec);
+
+	mc13xxx_lock(priv->mc13xxx);
+
+	/* these are the reset values */
+	mc13xxx_reg_write(priv->mc13xxx, MC13783_AUDIO_RX0, 0x25893);
+	mc13xxx_reg_write(priv->mc13xxx, MC13783_AUDIO_RX1, 0x00d35A);
+	mc13xxx_reg_write(priv->mc13xxx, MC13783_AUDIO_TX, 0x420000);
+	mc13xxx_reg_write(priv->mc13xxx, MC13783_SSI_NETWORK, 0x013060);
+	mc13xxx_reg_write(priv->mc13xxx, MC13783_AUDIO_CODEC, 0x180027);
+	mc13xxx_reg_write(priv->mc13xxx, MC13783_AUDIO_DAC, 0x0e0004);
+
+	if (priv->adc_ssi_port == MC13783_SSI1_PORT)
+		mc13xxx_reg_rmw(priv->mc13xxx, MC13783_AUDIO_CODEC,
+				AUDIO_SSI_SEL, 0);
+	else
+		mc13xxx_reg_rmw(priv->mc13xxx, MC13783_AUDIO_CODEC,
+				0, AUDIO_SSI_SEL);
+
+	if (priv->dac_ssi_port == MC13783_SSI1_PORT)
+		mc13xxx_reg_rmw(priv->mc13xxx, MC13783_AUDIO_DAC,
+				AUDIO_SSI_SEL, 0);
+	else
+		mc13xxx_reg_rmw(priv->mc13xxx, MC13783_AUDIO_DAC,
+				0, AUDIO_SSI_SEL);
+
+	mc13xxx_unlock(priv->mc13xxx);
+
+	return 0;
+}
+
+static int mc13783_remove(struct snd_soc_codec *codec)
+{
+	struct mc13783_priv *priv = snd_soc_codec_get_drvdata(codec);
+
+	mc13xxx_lock(priv->mc13xxx);
+
+	/* Make sure VAUDIOON is off */
+	mc13xxx_reg_rmw(priv->mc13xxx, MC13783_AUDIO_RX0, 0x3, 0);
+
+	mc13xxx_unlock(priv->mc13xxx);
+
+	return 0;
+}
+
+#define MC13783_RATES_RECORD (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000)
+
+#define MC13783_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
+	SNDRV_PCM_FMTBIT_S24_LE)
+
+static struct snd_soc_dai_ops mc13783_ops_dac = {
+	.hw_params	= mc13783_pcm_hw_params_dac,
+	.set_fmt	= mc13783_set_fmt_async,
+	.set_sysclk	= mc13783_set_sysclk_dac,
+	.set_tdm_slot	= mc13783_set_tdm_slot_dac,
+};
+
+static struct snd_soc_dai_ops mc13783_ops_codec = {
+	.hw_params	= mc13783_pcm_hw_params_codec,
+	.set_fmt	= mc13783_set_fmt_async,
+	.set_sysclk	= mc13783_set_sysclk_codec,
+	.set_tdm_slot	= mc13783_set_tdm_slot_codec,
+};
+
+/*
+ * The mc13783 has two SSI ports, both of them can be routed either
+ * to the voice codec or the stereo DAC. When two different SSI ports
+ * are used for the voice codec and the stereo DAC we can do different
+ * formats and sysclock settings for playback and capture
+ * (mc13783-hifi-playback and mc13783-hifi-capture). Using the same port
+ * forces us to use symmetric rates (mc13783-hifi).
+ */
+static struct snd_soc_dai_driver mc13783_dai_async[] = {
+	{
+		.name = "mc13783-hifi-playback",
+		.id = MC13783_ID_STEREO_DAC,
+		.playback = {
+			.stream_name = "Playback",
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = SNDRV_PCM_RATE_8000_96000,
+			.formats = MC13783_FORMATS,
+		},
+		.ops = &mc13783_ops_dac,
+	}, {
+		.name = "mc13783-hifi-capture",
+		.id = MC13783_ID_STEREO_CODEC,
+		.capture = {
+			.stream_name = "Capture",
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = MC13783_RATES_RECORD,
+			.formats = MC13783_FORMATS,
+		},
+		.ops = &mc13783_ops_codec,
+	},
+};
+
+static struct snd_soc_dai_ops mc13783_ops_sync = {
+	.hw_params	= mc13783_pcm_hw_params_sync,
+	.set_fmt	= mc13783_set_fmt_sync,
+	.set_sysclk	= mc13783_set_sysclk_sync,
+	.set_tdm_slot	= mc13783_set_tdm_slot_sync,
+};
+
+static struct snd_soc_dai_driver mc13783_dai_sync[] = {
+	{
+		.name = "mc13783-hifi",
+		.id = MC13783_ID_SYNC,
+		.playback = {
+			.stream_name = "Playback",
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = SNDRV_PCM_RATE_8000_96000,
+			.formats = MC13783_FORMATS,
+		},
+		.capture = {
+			.stream_name = "Capture",
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = MC13783_RATES_RECORD,
+			.formats = MC13783_FORMATS,
+		},
+		.ops = &mc13783_ops_sync,
+		.symmetric_rates = 1,
+	}
+};
+
+static struct snd_soc_codec_driver soc_codec_dev_mc13783 = {
+	.probe		= mc13783_probe,
+	.remove		= mc13783_remove,
+	.read		= mc13783_read,
+	.write		= mc13783_write,
+	.controls	= mc13783_control_list,
+	.num_controls	= ARRAY_SIZE(mc13783_control_list),
+	.dapm_widgets	= mc13783_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(mc13783_dapm_widgets),
+	.dapm_routes	= mc13783_routes,
+	.num_dapm_routes = ARRAY_SIZE(mc13783_routes),
+};
+
+static int mc13783_codec_probe(struct platform_device *pdev)
+{
+	struct mc13xxx *mc13xxx;
+	struct mc13783_priv *priv;
+	struct mc13xxx_codec_platform_data *pdata = pdev->dev.platform_data;
+	int ret;
+
+	mc13xxx = dev_get_drvdata(pdev->dev.parent);
+
+
+	priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL);
+	if (priv == NULL)
+		return -ENOMEM;
+
+	dev_set_drvdata(&pdev->dev, priv);
+	priv->mc13xxx = mc13xxx;
+	if (pdata) {
+		priv->adc_ssi_port = pdata->adc_ssi_port;
+		priv->dac_ssi_port = pdata->dac_ssi_port;
+	} else {
+		priv->adc_ssi_port = MC13783_SSI1_PORT;
+		priv->dac_ssi_port = MC13783_SSI2_PORT;
+	}
+
+	if (priv->adc_ssi_port == priv->dac_ssi_port)
+		ret = snd_soc_register_codec(&pdev->dev, &soc_codec_dev_mc13783,
+			mc13783_dai_sync, ARRAY_SIZE(mc13783_dai_sync));
+	else
+		ret = snd_soc_register_codec(&pdev->dev, &soc_codec_dev_mc13783,
+			mc13783_dai_async, ARRAY_SIZE(mc13783_dai_async));
+
+	if (ret)
+		goto err_register_codec;
+
+	return 0;
+
+err_register_codec:
+	dev_err(&pdev->dev, "register codec failed with %d\n", ret);
+
+	return ret;
+}
+
+static int mc13783_codec_remove(struct platform_device *pdev)
+{
+	snd_soc_unregister_codec(&pdev->dev);
+
+	return 0;
+}
+
+static struct platform_driver mc13783_codec_driver = {
+	.driver = {
+		   .name = "mc13783-codec",
+		   .owner = THIS_MODULE,
+		   },
+	.probe = mc13783_codec_probe,
+	.remove = __devexit_p(mc13783_codec_remove),
+};
+
+module_platform_driver(mc13783_codec_driver);
+
+MODULE_DESCRIPTION("ASoC MC13783 driver");
+MODULE_AUTHOR("Sascha Hauer, Pengutronix <s.hauer@pengutronix.de>");
+MODULE_AUTHOR("Philippe Retornaz <philippe.retornaz@epfl.ch>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/mc13783.h b/sound/soc/codecs/mc13783.h
new file mode 100644
index 000000000000..3a6d1993a217
--- /dev/null
+++ b/sound/soc/codecs/mc13783.h
@@ -0,0 +1,28 @@
+/*
+ * Copyright 2008 Juergen Beisert, kernel@pengutronix.de
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * as published by the Free Software Foundation; either version 2
+ * of the License, or (at your option) any later version.
+ * 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 Street, Fifth Floor, Boston, MA 02110-1301, USA.
+ */
+
+#ifndef MC13783_MIXER_H
+#define MC13783_MIXER_H
+
+#define MC13783_CLK_CLIA	1
+#define MC13783_CLK_CLIB	2
+
+#define MC13783_ID_STEREO_DAC	1
+#define MC13783_ID_STEREO_CODEC	2
+#define MC13783_ID_SYNC		3
+
+#endif /* MC13783_MIXER_H */
diff --git a/sound/soc/codecs/ml26124.c b/sound/soc/codecs/ml26124.c
new file mode 100644
index 000000000000..22cb5bf59273
--- /dev/null
+++ b/sound/soc/codecs/ml26124.c
@@ -0,0 +1,681 @@
+/*
+ * Copyright (C) 2011 LAPIS Semiconductor Co., Ltd.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; version 2 of the License.
+ *
+ * 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., 59 Temple Place, Suite 330, Boston, MA  02111-1307, USA.
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/i2c.h>
+#include <linux/slab.h>
+#include <linux/platform_device.h>
+#include <linux/regmap.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/tlv.h>
+#include "ml26124.h"
+
+#define DVOL_CTL_DVMUTE_ON		BIT(4)	/* Digital volume MUTE On */
+#define DVOL_CTL_DVMUTE_OFF		0	/* Digital volume MUTE Off */
+#define ML26124_SAI_NO_DELAY	BIT(1)
+#define ML26124_SAI_FRAME_SYNC	(BIT(5) | BIT(0)) /* For mono (Telecodec) */
+#define ML26134_CACHESIZE 212
+#define ML26124_VMID	BIT(1)
+#define ML26124_RATES (SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_32000 |\
+		       SNDRV_PCM_RATE_48000)
+#define ML26124_FORMATS (SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE |\
+			 SNDRV_PCM_FMTBIT_S32_LE)
+#define ML26124_NUM_REGISTER ML26134_CACHESIZE
+
+struct ml26124_priv {
+	u32 mclk;
+	u32 rate;
+	struct regmap *regmap;
+	int clk_in;
+	struct snd_pcm_substream *substream;
+};
+
+struct clk_coeff {
+	u32 mclk;
+	u32 rate;
+	u8 pllnl;
+	u8 pllnh;
+	u8 pllml;
+	u8 pllmh;
+	u8 plldiv;
+};
+
+/* ML26124 configuration */
+static const DECLARE_TLV_DB_SCALE(digital_tlv, -7150, 50, 0);
+
+static const DECLARE_TLV_DB_SCALE(alclvl, -2250, 150, 0);
+static const DECLARE_TLV_DB_SCALE(mingain, -1200, 600, 0);
+static const DECLARE_TLV_DB_SCALE(maxgain, -675, 600, 0);
+static const DECLARE_TLV_DB_SCALE(boost_vol, -1200, 75, 0);
+static const DECLARE_TLV_DB_SCALE(ngth, -7650, 150, 0);
+
+static const char * const ml26124_companding[] = {"16bit PCM", "u-law",
+						  "A-law"};
+
+static const struct soc_enum ml26124_adc_companding_enum
+	= SOC_ENUM_SINGLE(ML26124_SAI_TRANS_CTL, 6, 3, ml26124_companding);
+
+static const struct soc_enum ml26124_dac_companding_enum
+	= SOC_ENUM_SINGLE(ML26124_SAI_RCV_CTL, 6, 3, ml26124_companding);
+
+static const struct snd_kcontrol_new ml26124_snd_controls[] = {
+	SOC_SINGLE_TLV("Capture Digital Volume", ML26124_RECORD_DIG_VOL, 0,
+			0xff, 1, digital_tlv),
+	SOC_SINGLE_TLV("Playback Digital Volume", ML26124_PLBAK_DIG_VOL, 0,
+			0xff, 1, digital_tlv),
+	SOC_SINGLE_TLV("Digital Boost Volume", ML26124_DIGI_BOOST_VOL, 0,
+			0x3f, 0, boost_vol),
+	SOC_SINGLE_TLV("EQ Band0 Volume", ML26124_EQ_GAIN_BRAND0, 0,
+			0xff, 1, digital_tlv),
+	SOC_SINGLE_TLV("EQ Band1 Volume", ML26124_EQ_GAIN_BRAND1, 0,
+			0xff, 1, digital_tlv),
+	SOC_SINGLE_TLV("EQ Band2 Volume", ML26124_EQ_GAIN_BRAND2, 0,
+			0xff, 1, digital_tlv),
+	SOC_SINGLE_TLV("EQ Band3 Volume", ML26124_EQ_GAIN_BRAND3, 0,
+			0xff, 1, digital_tlv),
+	SOC_SINGLE_TLV("EQ Band4 Volume", ML26124_EQ_GAIN_BRAND4, 0,
+			0xff, 1, digital_tlv),
+	SOC_SINGLE_TLV("ALC Target Level", ML26124_ALC_TARGET_LEV, 0,
+			0xf, 1, alclvl),
+	SOC_SINGLE_TLV("ALC Min Input Volume", ML26124_ALC_MAXMIN_GAIN, 0,
+			7, 0, mingain),
+	SOC_SINGLE_TLV("ALC Max Input Volume", ML26124_ALC_MAXMIN_GAIN, 4,
+			7, 1, maxgain),
+	SOC_SINGLE_TLV("Playback Limiter Min Input Volume",
+			ML26124_PL_MAXMIN_GAIN, 0, 7, 0, mingain),
+	SOC_SINGLE_TLV("Playback Limiter Max Input Volume",
+			ML26124_PL_MAXMIN_GAIN, 4, 7, 1, maxgain),
+	SOC_SINGLE_TLV("Playback Boost Volume", ML26124_PLYBAK_BOST_VOL, 0,
+			0x3f, 0, boost_vol),
+	SOC_SINGLE("DC High Pass Filter Switch", ML26124_FILTER_EN, 0, 1, 0),
+	SOC_SINGLE("Noise High Pass Filter Switch", ML26124_FILTER_EN, 1, 1, 0),
+	SOC_SINGLE("ZC Switch", ML26124_PW_ZCCMP_PW_MNG, 1,
+		    1, 0),
+	SOC_SINGLE("EQ Band0 Switch", ML26124_FILTER_EN, 2, 1, 0),
+	SOC_SINGLE("EQ Band1 Switch", ML26124_FILTER_EN, 3, 1, 0),
+	SOC_SINGLE("EQ Band2 Switch", ML26124_FILTER_EN, 4, 1, 0),
+	SOC_SINGLE("EQ Band3 Switch", ML26124_FILTER_EN, 5, 1, 0),
+	SOC_SINGLE("EQ Band4 Switch", ML26124_FILTER_EN, 6, 1, 0),
+	SOC_SINGLE("Play Limiter", ML26124_DVOL_CTL, 0, 1, 0),
+	SOC_SINGLE("Capture Limiter", ML26124_DVOL_CTL, 1, 1, 0),
+	SOC_SINGLE("Digital Volume Fade Switch", ML26124_DVOL_CTL, 3, 1, 0),
+	SOC_SINGLE("Digital Switch", ML26124_DVOL_CTL, 4, 1, 0),
+	SOC_ENUM("DAC Companding", ml26124_dac_companding_enum),
+	SOC_ENUM("ADC Companding", ml26124_adc_companding_enum),
+};
+
+static const struct snd_kcontrol_new ml26124_output_mixer_controls[] = {
+	SOC_DAPM_SINGLE("DAC Switch", ML26124_SPK_AMP_OUT, 1, 1, 0),
+	SOC_DAPM_SINGLE("Line in loopback Switch", ML26124_SPK_AMP_OUT, 3, 1,
+			 0),
+	SOC_DAPM_SINGLE("PGA Switch", ML26124_SPK_AMP_OUT, 5, 1, 0),
+};
+
+/* Input mux */
+static const char * const ml26124_input_select[] = {"Analog MIC SingleEnded in",
+				"Digital MIC in", "Analog MIC Differential in"};
+
+static const struct soc_enum ml26124_insel_enum =
+	SOC_ENUM_SINGLE(ML26124_MIC_IF_CTL, 0, 3, ml26124_input_select);
+
+static const struct snd_kcontrol_new ml26124_input_mux_controls =
+	SOC_DAPM_ENUM("Input Select", ml26124_insel_enum);
+
+static const struct snd_kcontrol_new ml26124_line_control =
+	SOC_DAPM_SINGLE("Switch", ML26124_PW_LOUT_PW_MNG, 1, 1, 0);
+
+static const struct snd_soc_dapm_widget ml26124_dapm_widgets[] = {
+	SND_SOC_DAPM_SUPPLY("MCLKEN", ML26124_CLK_EN, 0, 0, NULL, 0),
+	SND_SOC_DAPM_SUPPLY("PLLEN", ML26124_CLK_EN, 1, 0, NULL, 0),
+	SND_SOC_DAPM_SUPPLY("PLLOE", ML26124_CLK_EN, 2, 0, NULL, 0),
+	SND_SOC_DAPM_SUPPLY("MICBIAS", ML26124_PW_REF_PW_MNG, 2, 0, NULL, 0),
+	SND_SOC_DAPM_MIXER("Output Mixer", SND_SOC_NOPM, 0, 0,
+			   &ml26124_output_mixer_controls[0],
+			   ARRAY_SIZE(ml26124_output_mixer_controls)),
+	SND_SOC_DAPM_DAC("DAC", "Playback", ML26124_PW_DAC_PW_MNG, 1, 0),
+	SND_SOC_DAPM_ADC("ADC", "Capture", ML26124_PW_IN_PW_MNG, 1, 0),
+	SND_SOC_DAPM_PGA("PGA", ML26124_PW_IN_PW_MNG, 3, 0, NULL, 0),
+	SND_SOC_DAPM_MUX("Input Mux", SND_SOC_NOPM, 0, 0,
+			  &ml26124_input_mux_controls),
+	SND_SOC_DAPM_SWITCH("Line Out Enable", SND_SOC_NOPM, 0, 0,
+			     &ml26124_line_control),
+	SND_SOC_DAPM_INPUT("MDIN"),
+	SND_SOC_DAPM_INPUT("MIN"),
+	SND_SOC_DAPM_INPUT("LIN"),
+	SND_SOC_DAPM_OUTPUT("SPOUT"),
+	SND_SOC_DAPM_OUTPUT("LOUT"),
+};
+
+static const struct snd_soc_dapm_route ml26124_intercon[] = {
+	/* Supply */
+	{"DAC", NULL, "MCLKEN"},
+	{"ADC", NULL, "MCLKEN"},
+	{"DAC", NULL, "PLLEN"},
+	{"ADC", NULL, "PLLEN"},
+	{"DAC", NULL, "PLLOE"},
+	{"ADC", NULL, "PLLOE"},
+
+	/* output mixer */
+	{"Output Mixer", "DAC Switch", "DAC"},
+	{"Output Mixer", "Line in loopback Switch", "LIN"},
+
+	/* outputs */
+	{"LOUT", NULL, "Output Mixer"},
+	{"SPOUT", NULL, "Output Mixer"},
+	{"Line Out Enable", NULL, "LOUT"},
+
+	/* input */
+	{"ADC", NULL, "Input Mux"},
+	{"Input Mux", "Analog MIC SingleEnded in", "PGA"},
+	{"Input Mux", "Analog MIC Differential in", "PGA"},
+	{"PGA", NULL, "MIN"},
+};
+
+/* PLLOutputFreq(Hz) = InputMclkFreq(Hz) * PLLM / (PLLN * PLLDIV) */
+static const struct clk_coeff coeff_div[] = {
+	{12288000, 16000, 0xc, 0x0, 0x20, 0x0, 0x4},
+	{12288000, 32000, 0xc, 0x0, 0x20, 0x0, 0x4},
+	{12288000, 48000, 0xc, 0x0, 0x30, 0x0, 0x4},
+};
+
+static struct reg_default ml26124_reg[] = {
+	/* CLOCK control Register */
+	{0x00, 0x00 },	/* Sampling Rate */
+	{0x02, 0x00},	/* PLL NL */
+	{0x04, 0x00},	/* PLLNH */
+	{0x06, 0x00},	/* PLLML */
+	{0x08, 0x00},	/* MLLMH */
+	{0x0a, 0x00},	/* PLLDIV */
+	{0x0c, 0x00},	/* Clock Enable */
+	{0x0e, 0x00},	/* CLK Input/Output Control */
+
+	/* System Control Register */
+	{0x10, 0x00},	/* Software RESET */
+	{0x12, 0x00},	/* Record/Playback Run */
+	{0x14, 0x00},	/* Mic Input/Output control */
+
+	/* Power Management Register */
+	{0x20, 0x00},	/* Reference Power Management */
+	{0x22, 0x00},	/* Input Power Management */
+	{0x24, 0x00},	/* DAC Power Management */
+	{0x26, 0x00},	/* SP-AMP Power Management */
+	{0x28, 0x00},	/* LINEOUT Power Management */
+	{0x2a, 0x00},	/* VIDEO Power Management */
+	{0x2e, 0x00},	/* AC-CMP Power Management */
+
+	/* Analog reference Control Register */
+	{0x30, 0x04},	/* MICBIAS Voltage Control */
+
+	/* Input/Output Amplifier Control Register */
+	{0x32, 0x10},	/* MIC Input Volume */
+	{0x38, 0x00},	/* Mic Boost Volume */
+	{0x3a, 0x33},	/* Speaker AMP Volume */
+	{0x48, 0x00},	/* AMP Volume Control Function Enable */
+	{0x4a, 0x00},	/* Amplifier Volume Fader Control */
+
+	/* Analog Path Control Register */
+	{0x54, 0x00},	/* Speaker AMP Output Control */
+	{0x5a, 0x00},	/* Mic IF Control */
+	{0xe8, 0x01},	/* Mic Select Control */
+
+	/* Audio Interface Control Register */
+	{0x60, 0x00},	/* SAI-Trans Control */
+	{0x62, 0x00},	/* SAI-Receive Control */
+	{0x64, 0x00},	/* SAI Mode select */
+
+	/* DSP Control Register */
+	{0x66, 0x01},	/* Filter Func Enable */
+	{0x68, 0x00},	/* Volume Control Func Enable */
+	{0x6A, 0x00},	/* Mixer & Volume Control*/
+	{0x6C, 0xff},	/* Record Digital Volume */
+	{0x70, 0xff},	/* Playback Digital Volume */
+	{0x72, 0x10},	/* Digital Boost Volume */
+	{0x74, 0xe7},	/* EQ gain Band0 */
+	{0x76, 0xe7},	/* EQ gain Band1 */
+	{0x78, 0xe7},	/* EQ gain Band2 */
+	{0x7A, 0xe7},	/* EQ gain Band3 */
+	{0x7C, 0xe7},	/* EQ gain Band4 */
+	{0x7E, 0x00},	/* HPF2 CutOff*/
+	{0x80, 0x00},	/* EQ Band0 Coef0L */
+	{0x82, 0x00},	/* EQ Band0 Coef0H */
+	{0x84, 0x00},	/* EQ Band0 Coef0L */
+	{0x86, 0x00},	/* EQ Band0 Coef0H */
+	{0x88, 0x00},	/* EQ Band1 Coef0L */
+	{0x8A, 0x00},	/* EQ Band1 Coef0H */
+	{0x8C, 0x00},	/* EQ Band1 Coef0L */
+	{0x8E, 0x00},	/* EQ Band1 Coef0H */
+	{0x90, 0x00},	/* EQ Band2 Coef0L */
+	{0x92, 0x00},	/* EQ Band2 Coef0H */
+	{0x94, 0x00},	/* EQ Band2 Coef0L */
+	{0x96, 0x00},	/* EQ Band2 Coef0H */
+	{0x98, 0x00},	/* EQ Band3 Coef0L */
+	{0x9A, 0x00},	/* EQ Band3 Coef0H */
+	{0x9C, 0x00},	/* EQ Band3 Coef0L */
+	{0x9E, 0x00},	/* EQ Band3 Coef0H */
+	{0xA0, 0x00},	/* EQ Band4 Coef0L */
+	{0xA2, 0x00},	/* EQ Band4 Coef0H */
+	{0xA4, 0x00},	/* EQ Band4 Coef0L */
+	{0xA6, 0x00},	/* EQ Band4 Coef0H */
+
+	/* ALC Control Register */
+	{0xb0, 0x00},	/* ALC Mode */
+	{0xb2, 0x02},	/* ALC Attack Time */
+	{0xb4, 0x03},	/* ALC Decay Time */
+	{0xb6, 0x00},	/* ALC Hold Time */
+	{0xb8, 0x0b},	/* ALC Target Level */
+	{0xba, 0x70},	/* ALC Max/Min Gain */
+	{0xbc, 0x00},	/* Noise Gate Threshold */
+	{0xbe, 0x00},	/* ALC ZeroCross TimeOut */
+
+	/* Playback Limiter Control Register */
+	{0xc0, 0x04},	/* PL Attack Time */
+	{0xc2, 0x05},	/* PL Decay Time */
+	{0xc4, 0x0d},	/* PL Target Level */
+	{0xc6, 0x70},	/* PL Max/Min Gain */
+	{0xc8, 0x10},	/* Playback Boost Volume */
+	{0xca, 0x00},	/* PL ZeroCross TimeOut */
+
+	/* Video Amplifier Control Register */
+	{0xd0, 0x01},	/* VIDEO AMP Gain Control */
+	{0xd2, 0x01},	/* VIDEO AMP Setup 1 */
+	{0xd4, 0x01},	/* VIDEO AMP Control2 */
+};
+
+/* Get sampling rate value of sampling rate setting register (0x0) */
+static inline int get_srate(int rate)
+{
+	int srate;
+
+	switch (rate) {
+	case 16000:
+		srate = 3;
+		break;
+	case 32000:
+		srate = 6;
+		break;
+	case 48000:
+		srate = 8;
+		break;
+	default:
+		return -EINVAL;
+	}
+	return srate;
+}
+
+static inline int get_coeff(int mclk, int rate)
+{
+	int i;
+
+	for (i = 0; i < ARRAY_SIZE(coeff_div); i++) {
+		if (coeff_div[i].rate == rate && coeff_div[i].mclk == mclk)
+			return i;
+	}
+	return -EINVAL;
+}
+
+static int ml26124_hw_params(struct snd_pcm_substream *substream,
+			    struct snd_pcm_hw_params *hw_params,
+			    struct snd_soc_dai *dai)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	struct ml26124_priv *priv = snd_soc_codec_get_drvdata(codec);
+	int i = get_coeff(priv->mclk, params_rate(hw_params));
+
+	priv->substream = substream;
+	priv->rate = params_rate(hw_params);
+
+	if (priv->clk_in) {
+		switch (priv->mclk / params_rate(hw_params)) {
+		case 256:
+			snd_soc_update_bits(codec, ML26124_CLK_CTL,
+					    BIT(0) | BIT(1), 1);
+			break;
+		case 512:
+			snd_soc_update_bits(codec, ML26124_CLK_CTL,
+					    BIT(0) | BIT(1), 2);
+			break;
+		case 1024:
+			snd_soc_update_bits(codec, ML26124_CLK_CTL,
+					    BIT(0) | BIT(1), 3);
+			break;
+		default:
+			dev_err(codec->dev, "Unsupported MCLKI\n");
+			break;
+		}
+	} else {
+		snd_soc_update_bits(codec, ML26124_CLK_CTL,
+				    BIT(0) | BIT(1), 0);
+	}
+
+	switch (params_rate(hw_params)) {
+	case 16000:
+		snd_soc_update_bits(codec, ML26124_SMPLING_RATE, 0xf,
+				    get_srate(params_rate(hw_params)));
+		snd_soc_update_bits(codec, ML26124_PLLNL, 0xff,
+				    coeff_div[i].pllnl);
+		snd_soc_update_bits(codec, ML26124_PLLNH, 0x1,
+				    coeff_div[i].pllnh);
+		snd_soc_update_bits(codec, ML26124_PLLML, 0xff,
+				    coeff_div[i].pllml);
+		snd_soc_update_bits(codec, ML26124_PLLMH, 0x3f,
+				    coeff_div[i].pllmh);
+		snd_soc_update_bits(codec, ML26124_PLLDIV, 0x1f,
+				    coeff_div[i].plldiv);
+		break;
+	case 32000:
+		snd_soc_update_bits(codec, ML26124_SMPLING_RATE, 0xf,
+				    get_srate(params_rate(hw_params)));
+		snd_soc_update_bits(codec, ML26124_PLLNL, 0xff,
+				    coeff_div[i].pllnl);
+		snd_soc_update_bits(codec, ML26124_PLLNH, 0x1,
+				    coeff_div[i].pllnh);
+		snd_soc_update_bits(codec, ML26124_PLLML, 0xff,
+				    coeff_div[i].pllml);
+		snd_soc_update_bits(codec, ML26124_PLLMH, 0x3f,
+				    coeff_div[i].pllmh);
+		snd_soc_update_bits(codec, ML26124_PLLDIV, 0x1f,
+				    coeff_div[i].plldiv);
+		break;
+	case 48000:
+		snd_soc_update_bits(codec, ML26124_SMPLING_RATE, 0xf,
+				    get_srate(params_rate(hw_params)));
+		snd_soc_update_bits(codec, ML26124_PLLNL, 0xff,
+				    coeff_div[i].pllnl);
+		snd_soc_update_bits(codec, ML26124_PLLNH, 0x1,
+				    coeff_div[i].pllnh);
+		snd_soc_update_bits(codec, ML26124_PLLML, 0xff,
+				    coeff_div[i].pllml);
+		snd_soc_update_bits(codec, ML26124_PLLMH, 0x3f,
+				    coeff_div[i].pllmh);
+		snd_soc_update_bits(codec, ML26124_PLLDIV, 0x1f,
+				    coeff_div[i].plldiv);
+		break;
+	default:
+		pr_err("%s:this rate is no support for ml26124\n", __func__);
+		return -EINVAL;
+	}
+
+	return 0;
+}
+
+static int ml26124_mute(struct snd_soc_dai *dai, int mute)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	struct ml26124_priv *priv = snd_soc_codec_get_drvdata(codec);
+
+	switch (priv->substream->stream) {
+	case SNDRV_PCM_STREAM_CAPTURE:
+		snd_soc_update_bits(codec, ML26124_REC_PLYBAK_RUN, BIT(0), 1);
+		break;
+	case SNDRV_PCM_STREAM_PLAYBACK:
+		snd_soc_update_bits(codec, ML26124_REC_PLYBAK_RUN, BIT(1), 2);
+		break;
+	}
+
+	if (mute)
+		snd_soc_update_bits(codec, ML26124_DVOL_CTL, BIT(4),
+				    DVOL_CTL_DVMUTE_ON);
+	else
+		snd_soc_update_bits(codec, ML26124_DVOL_CTL, BIT(4),
+				    DVOL_CTL_DVMUTE_OFF);
+
+	return 0;
+}
+
+static int ml26124_set_dai_fmt(struct snd_soc_dai *codec_dai,
+		unsigned int fmt)
+{
+	unsigned char mode;
+	struct snd_soc_codec *codec = codec_dai->codec;
+
+	/* set master/slave audio interface */
+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBM_CFM:
+		mode = 1;
+		break;
+	case SND_SOC_DAIFMT_CBS_CFS:
+		mode = 0;
+		break;
+	default:
+		return -EINVAL;
+	}
+	snd_soc_update_bits(codec, ML26124_SAI_MODE_SEL, BIT(0), mode);
+
+	/* interface format */
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	/* clock inversion */
+	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+	case SND_SOC_DAIFMT_NB_NF:
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	return 0;
+}
+
+static int ml26124_set_dai_sysclk(struct snd_soc_dai *codec_dai,
+		int clk_id, unsigned int freq, int dir)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	struct ml26124_priv *priv = snd_soc_codec_get_drvdata(codec);
+
+	switch (clk_id) {
+	case ML26124_USE_PLLOUT:
+		priv->clk_in = ML26124_USE_PLLOUT;
+		break;
+	case ML26124_USE_MCLKI:
+		priv->clk_in = ML26124_USE_MCLKI;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	priv->mclk = freq;
+
+	return 0;
+}
+
+static int ml26124_set_bias_level(struct snd_soc_codec *codec,
+		enum snd_soc_bias_level level)
+{
+	struct ml26124_priv *priv = snd_soc_codec_get_drvdata(codec);
+
+	switch (level) {
+	case SND_SOC_BIAS_ON:
+		snd_soc_update_bits(codec, ML26124_PW_SPAMP_PW_MNG,
+				    ML26124_R26_MASK, ML26124_BLT_PREAMP_ON);
+		msleep(100);
+		snd_soc_update_bits(codec, ML26124_PW_SPAMP_PW_MNG,
+				    ML26124_R26_MASK,
+				    ML26124_MICBEN_ON | ML26124_BLT_ALL_ON);
+		break;
+	case SND_SOC_BIAS_PREPARE:
+		break;
+	case SND_SOC_BIAS_STANDBY:
+		/* VMID ON */
+		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
+			snd_soc_update_bits(codec, ML26124_PW_REF_PW_MNG,
+					    ML26124_VMID, ML26124_VMID);
+			msleep(500);
+			regcache_sync(priv->regmap);
+		}
+		break;
+	case SND_SOC_BIAS_OFF:
+		/* VMID OFF */
+		snd_soc_update_bits(codec, ML26124_PW_REF_PW_MNG,
+				    ML26124_VMID, 0);
+		break;
+	}
+	codec->dapm.bias_level = level;
+	return 0;
+}
+
+static const struct snd_soc_dai_ops ml26124_dai_ops = {
+	.hw_params	= ml26124_hw_params,
+	.digital_mute	= ml26124_mute,
+	.set_fmt	= ml26124_set_dai_fmt,
+	.set_sysclk	= ml26124_set_dai_sysclk,
+};
+
+static struct snd_soc_dai_driver ml26124_dai = {
+	.name = "ml26124-hifi",
+	.playback = {
+		.stream_name = "Playback",
+		.channels_min = 1,
+		.channels_max = 2,
+		.rates = ML26124_RATES,
+		.formats = ML26124_FORMATS,},
+	.capture = {
+		.stream_name = "Capture",
+		.channels_min = 1,
+		.channels_max = 2,
+		.rates = ML26124_RATES,
+		.formats = ML26124_FORMATS,},
+	.ops = &ml26124_dai_ops,
+	.symmetric_rates = 1,
+};
+
+#ifdef CONFIG_PM
+static int ml26124_suspend(struct snd_soc_codec *codec)
+{
+	ml26124_set_bias_level(codec, SND_SOC_BIAS_OFF);
+
+	return 0;
+}
+
+static int ml26124_resume(struct snd_soc_codec *codec)
+{
+	ml26124_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+	return 0;
+}
+#else
+#define ml26124_suspend NULL
+#define ml26124_resume NULL
+#endif
+
+static int ml26124_probe(struct snd_soc_codec *codec)
+{
+	int ret;
+	struct ml26124_priv *priv = snd_soc_codec_get_drvdata(codec);
+	codec->control_data = priv->regmap;
+
+	ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP);
+	if (ret < 0) {
+		dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
+		return ret;
+	}
+
+	/* Software Reset */
+	snd_soc_update_bits(codec, ML26124_SW_RST, 0x01, 1);
+	snd_soc_update_bits(codec, ML26124_SW_RST, 0x01, 0);
+
+	ml26124_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+	return 0;
+}
+
+static struct snd_soc_codec_driver soc_codec_dev_ml26124 = {
+	.probe =	ml26124_probe,
+	.suspend =	ml26124_suspend,
+	.resume =	ml26124_resume,
+	.set_bias_level = ml26124_set_bias_level,
+	.dapm_widgets = ml26124_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(ml26124_dapm_widgets),
+	.dapm_routes = ml26124_intercon,
+	.num_dapm_routes = ARRAY_SIZE(ml26124_intercon),
+	.controls = ml26124_snd_controls,
+	.num_controls = ARRAY_SIZE(ml26124_snd_controls),
+};
+
+static const struct regmap_config ml26124_i2c_regmap = {
+	.val_bits = 8,
+	.reg_bits = 8,
+	.max_register = ML26124_NUM_REGISTER,
+	.reg_defaults = ml26124_reg,
+	.num_reg_defaults = ARRAY_SIZE(ml26124_reg),
+	.cache_type = REGCACHE_RBTREE,
+	.write_flag_mask = 0x01,
+};
+
+static __devinit int ml26124_i2c_probe(struct i2c_client *i2c,
+				      const struct i2c_device_id *id)
+{
+	struct ml26124_priv *priv;
+	int ret;
+
+	priv = devm_kzalloc(&i2c->dev, sizeof(*priv), GFP_KERNEL);
+	if (!priv)
+		return -ENOMEM;
+
+	i2c_set_clientdata(i2c, priv);
+
+	priv->regmap = regmap_init_i2c(i2c, &ml26124_i2c_regmap);
+	if (IS_ERR(priv->regmap)) {
+		ret = PTR_ERR(priv->regmap);
+		dev_err(&i2c->dev, "regmap_init_i2c() failed: %d\n", ret);
+		return ret;
+	}
+
+	return snd_soc_register_codec(&i2c->dev,
+			&soc_codec_dev_ml26124, &ml26124_dai, 1);
+}
+
+static __devexit int ml26124_i2c_remove(struct i2c_client *client)
+{
+	struct ml26124_priv *priv = i2c_get_clientdata(client);
+
+	snd_soc_unregister_codec(&client->dev);
+	regmap_exit(priv->regmap);
+	return 0;
+}
+
+static const struct i2c_device_id ml26124_i2c_id[] = {
+	{ "ml26124", 0 },
+	{ }
+};
+MODULE_DEVICE_TABLE(i2c, ml26124_i2c_id);
+
+static struct i2c_driver ml26124_i2c_driver = {
+	.driver = {
+		.name = "ml26124",
+		.owner = THIS_MODULE,
+	},
+	.probe = ml26124_i2c_probe,
+	.remove = __devexit_p(ml26124_i2c_remove),
+	.id_table = ml26124_i2c_id,
+};
+
+module_i2c_driver(ml26124_i2c_driver);
+
+MODULE_AUTHOR("Tomoya MORINAGA <tomoya.rohm@gmail.com>");
+MODULE_DESCRIPTION("LAPIS Semiconductor ML26124 ALSA SoC codec driver");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/ml26124.h b/sound/soc/codecs/ml26124.h
new file mode 100644
index 000000000000..5ea0cbb8c46c
--- /dev/null
+++ b/sound/soc/codecs/ml26124.h
@@ -0,0 +1,184 @@
+/*
+ * Copyright (C) 2011 LAPIS Semiconductor Co., Ltd.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; version 2 of the License.
+ *
+ * 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., 59 Temple Place, Suite 330, Boston, MA  02111-1307, USA.
+ */
+
+#ifndef ML26124_H
+#define ML26124_H
+
+/* Clock Control Register */
+#define ML26124_SMPLING_RATE		0x00
+#define ML26124_PLLNL			0x02
+#define ML26124_PLLNH			0x04
+#define ML26124_PLLML			0x06
+#define ML26124_PLLMH			0x08
+#define ML26124_PLLDIV			0x0a
+#define ML26124_CLK_EN			0x0c
+#define ML26124_CLK_CTL			0x0e
+
+/* System Control Register */
+#define ML26124_SW_RST			0x10
+#define ML26124_REC_PLYBAK_RUN		0x12
+#define ML26124_MIC_TIM			0x14
+
+/* Power Mnagement Register */
+#define ML26124_PW_REF_PW_MNG		0x20
+#define ML26124_PW_IN_PW_MNG		0x22
+#define ML26124_PW_DAC_PW_MNG		0x24
+#define ML26124_PW_SPAMP_PW_MNG		0x26
+#define ML26124_PW_LOUT_PW_MNG		0x28
+#define ML26124_PW_VOUT_PW_MNG		0x2a
+#define ML26124_PW_ZCCMP_PW_MNG		0x2e
+
+/* Analog Reference Control Register */
+#define ML26124_PW_MICBIAS_VOL		0x30
+
+/* Input/Output Amplifier Control Register */
+#define ML26124_PW_MIC_IN_VOL		0x32
+#define ML26124_PW_MIC_BOST_VOL		0x38
+#define ML26124_PW_SPK_AMP_VOL		0x3a
+#define ML26124_PW_AMP_VOL_FUNC		0x48
+#define ML26124_PW_AMP_VOL_FADE		0x4a
+
+/* Analog Path Control Register */
+#define ML26124_SPK_AMP_OUT		0x54
+#define ML26124_MIC_IF_CTL		0x5a
+#define ML26124_MIC_SELECT		0xe8
+
+/* Audio Interface Control Register */
+#define ML26124_SAI_TRANS_CTL		0x60
+#define ML26124_SAI_RCV_CTL		0x62
+#define ML26124_SAI_MODE_SEL		0x64
+
+/* DSP Control Register */
+#define ML26124_FILTER_EN		0x66
+#define ML26124_DVOL_CTL		0x68
+#define ML26124_MIXER_VOL_CTL		0x6a
+#define ML26124_RECORD_DIG_VOL		0x6c
+#define ML26124_PLBAK_DIG_VOL		0x70
+#define ML26124_DIGI_BOOST_VOL		0x72
+#define ML26124_EQ_GAIN_BRAND0		0x74
+#define ML26124_EQ_GAIN_BRAND1		0x76
+#define ML26124_EQ_GAIN_BRAND2		0x78
+#define ML26124_EQ_GAIN_BRAND3		0x7a
+#define ML26124_EQ_GAIN_BRAND4		0x7c
+#define ML26124_HPF2_CUTOFF		0x7e
+#define ML26124_EQBRAND0_F0L		0x80
+#define ML26124_EQBRAND0_F0H		0x82
+#define ML26124_EQBRAND0_F1L		0x84
+#define ML26124_EQBRAND0_F1H		0x86
+#define ML26124_EQBRAND1_F0L		0x88
+#define ML26124_EQBRAND1_F0H		0x8a
+#define ML26124_EQBRAND1_F1L		0x8c
+#define ML26124_EQBRAND1_F1H		0x8e
+#define ML26124_EQBRAND2_F0L		0x90
+#define ML26124_EQBRAND2_F0H		0x92
+#define ML26124_EQBRAND2_F1L		0x94
+#define ML26124_EQBRAND2_F1H		0x96
+#define ML26124_EQBRAND3_F0L		0x98
+#define ML26124_EQBRAND3_F0H		0x9a
+#define ML26124_EQBRAND3_F1L		0x9c
+#define ML26124_EQBRAND3_F1H		0x9e
+#define ML26124_EQBRAND4_F0L		0xa0
+#define ML26124_EQBRAND4_F0H		0xa2
+#define ML26124_EQBRAND4_F1L		0xa4
+#define ML26124_EQBRAND4_F1H		0xa6
+
+/* ALC Control Register */
+#define ML26124_ALC_MODE		0xb0
+#define ML26124_ALC_ATTACK_TIM		0xb2
+#define ML26124_ALC_DECAY_TIM		0xb4
+#define ML26124_ALC_HOLD_TIM		0xb6
+#define ML26124_ALC_TARGET_LEV		0xb8
+#define ML26124_ALC_MAXMIN_GAIN		0xba
+#define ML26124_NOIS_GATE_THRSH		0xbc
+#define ML26124_ALC_ZERO_TIMOUT		0xbe
+
+/* Playback Limiter Control Register */
+#define ML26124_PL_ATTACKTIME		0xc0
+#define ML26124_PL_DECAYTIME		0xc2
+#define ML26124_PL_TARGETTIME		0xc4
+#define ML26124_PL_MAXMIN_GAIN		0xc6
+#define ML26124_PLYBAK_BOST_VOL		0xc8
+#define ML26124_PL_0CROSS_TIMOUT	0xca
+
+/* Video Amplifer Control Register */
+#define ML26124_VIDEO_AMP_GAIN_CTL	0xd0
+#define ML26124_VIDEO_AMP_SETUP1	0xd2
+#define ML26124_VIDEO_AMP_CTL2		0xd4
+
+/* Clock select for machine driver */
+#define ML26124_USE_PLL			0
+#define ML26124_USE_MCLKI_256FS		1
+#define ML26124_USE_MCLKI_512FS		2
+#define ML26124_USE_MCLKI_1024FS	3
+
+/* Register Mask */
+#define ML26124_R0_MASK	0xf
+#define ML26124_R2_MASK	0xff
+#define ML26124_R4_MASK	0x1
+#define ML26124_R6_MASK	0xf
+#define ML26124_R8_MASK	0x3f
+#define ML26124_Ra_MASK	0x1f
+#define ML26124_Rc_MASK	0x1f
+#define ML26124_Re_MASK	0x7
+#define ML26124_R10_MASK	0x1
+#define ML26124_R12_MASK	0x17
+#define ML26124_R14_MASK	0x3f
+#define ML26124_R20_MASK	0x47
+#define ML26124_R22_MASK	0xa
+#define ML26124_R24_MASK	0x2
+#define ML26124_R26_MASK	0x1f
+#define ML26124_R28_MASK	0x2
+#define ML26124_R2a_MASK	0x2
+#define ML26124_R2e_MASK	0x2
+#define ML26124_R30_MASK	0x7
+#define ML26124_R32_MASK	0x3f
+#define ML26124_R38_MASK	0x38
+#define ML26124_R3a_MASK	0x3f
+#define ML26124_R48_MASK	0x3
+#define ML26124_R4a_MASK	0x7
+#define ML26124_R54_MASK	0x2a
+#define ML26124_R5a_MASK	0x3
+#define ML26124_Re8_MASK	0x3
+#define ML26124_R60_MASK	0xff
+#define ML26124_R62_MASK	0xff
+#define ML26124_R64_MASK	0x1
+#define ML26124_R66_MASK	0xff
+#define ML26124_R68_MASK	0x3b
+#define ML26124_R6a_MASK	0xf3
+#define ML26124_R6c_MASK	0xff
+#define ML26124_R70_MASK	0xff
+
+#define ML26124_MCLKEN		BIT(0)
+#define ML26124_PLLEN		BIT(1)
+#define ML26124_PLLOE		BIT(2)
+#define ML26124_MCLKOE		BIT(3)
+
+#define ML26124_BLT_ALL_ON	0x1f
+#define ML26124_BLT_PREAMP_ON	0x13
+
+#define ML26124_MICBEN_ON	BIT(2)
+
+enum ml26124_regs {
+	ML26124_MCLK = 0,
+};
+
+enum ml26124_clk_in {
+	ML26124_USE_PLLOUT = 0,
+	ML26124_USE_MCLKI,
+};
+
+#endif
diff --git a/sound/soc/codecs/omap-hdmi.c b/sound/soc/codecs/omap-hdmi.c
new file mode 100644
index 000000000000..1bf5c74f5f96
--- /dev/null
+++ b/sound/soc/codecs/omap-hdmi.c
@@ -0,0 +1,69 @@
+/*
+ * ALSA SoC codec driver for HDMI audio on OMAP processors.
+ * Copyright (C) 2012 Texas Instruments Incorporated - http://www.ti.com/
+ * Author: Ricardo Neri <ricardo.neri@ti.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
+ *
+ */
+#include <linux/module.h>
+#include <sound/soc.h>
+
+#define DRV_NAME "hdmi-audio-codec"
+
+static struct snd_soc_codec_driver omap_hdmi_codec;
+
+static struct snd_soc_dai_driver omap_hdmi_codec_dai = {
+	.name = "omap-hdmi-hifi",
+	.playback = {
+		.channels_min = 2,
+		.channels_max = 8,
+		.rates = SNDRV_PCM_RATE_32000 |
+			SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |
+			SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 |
+			SNDRV_PCM_RATE_176400 | SNDRV_PCM_RATE_192000,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE |
+			SNDRV_PCM_FMTBIT_S24_LE,
+	},
+};
+
+static __devinit int omap_hdmi_codec_probe(struct platform_device *pdev)
+{
+	return snd_soc_register_codec(&pdev->dev, &omap_hdmi_codec,
+			&omap_hdmi_codec_dai, 1);
+}
+
+static __devexit int omap_hdmi_codec_remove(struct platform_device *pdev)
+{
+	snd_soc_unregister_codec(&pdev->dev);
+	return 0;
+}
+
+static struct platform_driver omap_hdmi_codec_driver = {
+	.driver		= {
+		.name	= DRV_NAME,
+		.owner	= THIS_MODULE,
+	},
+
+	.probe		= omap_hdmi_codec_probe,
+	.remove		= __devexit_p(omap_hdmi_codec_remove),
+};
+
+module_platform_driver(omap_hdmi_codec_driver);
+
+MODULE_AUTHOR("Ricardo Neri <ricardo.neri@ti.com>");
+MODULE_DESCRIPTION("ASoC OMAP HDMI codec driver");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:" DRV_NAME);
diff --git a/sound/soc/codecs/rt5631.c b/sound/soc/codecs/rt5631.c
index 20c324c7c349..960d0e93cce9 100644
--- a/sound/soc/codecs/rt5631.c
+++ b/sound/soc/codecs/rt5631.c
@@ -18,7 +18,7 @@
 #include <linux/delay.h>
 #include <linux/pm.h>
 #include <linux/i2c.h>
-#include <linux/spi/spi.h>
+#include <linux/regmap.h>
 #include <sound/core.h>
 #include <sound/pcm.h>
 #include <sound/pcm_params.h>
@@ -30,6 +30,7 @@
 #include "rt5631.h"
 
 struct rt5631_priv {
+	struct regmap *regmap;
 	int codec_version;
 	int master;
 	int sysclk;
@@ -38,33 +39,33 @@ struct rt5631_priv {
 	int dmic_used_flag;
 };
 
-static const u16 rt5631_reg[RT5631_VENDOR_ID2 + 1] = {
-	[RT5631_SPK_OUT_VOL] = 0x8888,
-	[RT5631_HP_OUT_VOL] = 0x8080,
-	[RT5631_MONO_AXO_1_2_VOL] = 0xa080,
-	[RT5631_AUX_IN_VOL] = 0x0808,
-	[RT5631_ADC_REC_MIXER] = 0xf0f0,
-	[RT5631_VDAC_DIG_VOL] = 0x0010,
-	[RT5631_OUTMIXER_L_CTRL] = 0xffc0,
-	[RT5631_OUTMIXER_R_CTRL] = 0xffc0,
-	[RT5631_AXO1MIXER_CTRL] = 0x88c0,
-	[RT5631_AXO2MIXER_CTRL] = 0x88c0,
-	[RT5631_DIG_MIC_CTRL] = 0x3000,
-	[RT5631_MONO_INPUT_VOL] = 0x8808,
-	[RT5631_SPK_MIXER_CTRL] = 0xf8f8,
-	[RT5631_SPK_MONO_OUT_CTRL] = 0xfc00,
-	[RT5631_SPK_MONO_HP_OUT_CTRL] = 0x4440,
-	[RT5631_SDP_CTRL] = 0x8000,
-	[RT5631_MONO_SDP_CTRL] = 0x8000,
-	[RT5631_STEREO_AD_DA_CLK_CTRL] = 0x2010,
-	[RT5631_GEN_PUR_CTRL_REG] = 0x0e00,
-	[RT5631_INT_ST_IRQ_CTRL_2] = 0x071a,
-	[RT5631_MISC_CTRL] = 0x2040,
-	[RT5631_DEPOP_FUN_CTRL_2] = 0x8000,
-	[RT5631_SOFT_VOL_CTRL] = 0x07e0,
-	[RT5631_ALC_CTRL_1] = 0x0206,
-	[RT5631_ALC_CTRL_3] = 0x2000,
-	[RT5631_PSEUDO_SPATL_CTRL] = 0x0553,
+static const struct reg_default rt5631_reg[] = {
+	{ RT5631_SPK_OUT_VOL, 0x8888 },
+	{ RT5631_HP_OUT_VOL, 0x8080 },
+	{ RT5631_MONO_AXO_1_2_VOL, 0xa080 },
+	{ RT5631_AUX_IN_VOL, 0x0808 },
+	{ RT5631_ADC_REC_MIXER, 0xf0f0 },
+	{ RT5631_VDAC_DIG_VOL, 0x0010 },
+	{ RT5631_OUTMIXER_L_CTRL, 0xffc0 },
+	{ RT5631_OUTMIXER_R_CTRL, 0xffc0 },
+	{ RT5631_AXO1MIXER_CTRL, 0x88c0 },
+	{ RT5631_AXO2MIXER_CTRL, 0x88c0 },
+	{ RT5631_DIG_MIC_CTRL, 0x3000 },
+	{ RT5631_MONO_INPUT_VOL, 0x8808 },
+	{ RT5631_SPK_MIXER_CTRL, 0xf8f8 },
+	{ RT5631_SPK_MONO_OUT_CTRL, 0xfc00 },
+	{ RT5631_SPK_MONO_HP_OUT_CTRL, 0x4440 },
+	{ RT5631_SDP_CTRL, 0x8000 },
+	{ RT5631_MONO_SDP_CTRL, 0x8000 },
+	{ RT5631_STEREO_AD_DA_CLK_CTRL, 0x2010 },
+	{ RT5631_GEN_PUR_CTRL_REG, 0x0e00 },
+	{ RT5631_INT_ST_IRQ_CTRL_2, 0x071a },
+	{ RT5631_MISC_CTRL, 0x2040 },
+	{ RT5631_DEPOP_FUN_CTRL_2, 0x8000 },
+	{ RT5631_SOFT_VOL_CTRL, 0x07e0 },
+	{ RT5631_ALC_CTRL_1, 0x0206 },
+	{ RT5631_ALC_CTRL_3, 0x2000 },
+	{ RT5631_PSEUDO_SPATL_CTRL, 0x0553 },
 };
 
 /**
@@ -96,8 +97,7 @@ static int rt5631_reset(struct snd_soc_codec *codec)
 	return snd_soc_write(codec, RT5631_RESET, 0);
 }
 
-static int rt5631_volatile_register(struct snd_soc_codec *codec,
-				    unsigned int reg)
+static bool rt5631_volatile_register(struct device *dev, unsigned int reg)
 {
 	switch (reg) {
 	case RT5631_RESET:
@@ -111,8 +111,7 @@ static int rt5631_volatile_register(struct snd_soc_codec *codec,
 	}
 }
 
-static int rt5631_readable_register(struct snd_soc_codec *codec,
-				    unsigned int reg)
+static bool rt5631_readable_register(struct device *dev, unsigned int reg)
 {
 	switch (reg) {
 	case RT5631_RESET:
@@ -1361,8 +1360,7 @@ static int get_coeff(int mclk, int rate, int timesofbclk)
 static int rt5631_hifi_pcm_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 snd_soc_codec *codec = dai->codec;
 	struct rt5631_priv *rt5631 = snd_soc_codec_get_drvdata(codec);
 	int timesofbclk = 32, coeff;
 	unsigned int iface = 0;
@@ -1544,6 +1542,8 @@ static int rt5631_codec_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id,
 static int rt5631_set_bias_level(struct snd_soc_codec *codec,
 			enum snd_soc_bias_level level)
 {
+	struct rt5631_priv *rt5631 = snd_soc_codec_get_drvdata(codec);
+
 	switch (level) {
 	case SND_SOC_BIAS_ON:
 	case SND_SOC_BIAS_PREPARE:
@@ -1561,8 +1561,8 @@ static int rt5631_set_bias_level(struct snd_soc_codec *codec,
 			snd_soc_update_bits(codec, RT5631_PWR_MANAG_ADD3,
 				RT5631_PWR_FAST_VREF_CTRL,
 				RT5631_PWR_FAST_VREF_CTRL);
-			codec->cache_only = false;
-			snd_soc_cache_sync(codec);
+			regcache_cache_only(rt5631->regmap, false);
+			regcache_sync(rt5631->regmap);
 		}
 		break;
 
@@ -1587,7 +1587,9 @@ static int rt5631_probe(struct snd_soc_codec *codec)
 	unsigned int val;
 	int ret;
 
-	ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_I2C);
+	codec->control_data = rt5631->regmap;
+
+	ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP);
 	if (ret != 0) {
 		dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
 		return ret;
@@ -1698,12 +1700,6 @@ static struct snd_soc_codec_driver soc_codec_dev_rt5631 = {
 	.suspend = rt5631_suspend,
 	.resume = rt5631_resume,
 	.set_bias_level = rt5631_set_bias_level,
-	.reg_cache_size = RT5631_VENDOR_ID2 + 1,
-	.reg_word_size = sizeof(u16),
-	.reg_cache_default = rt5631_reg,
-	.volatile_register = rt5631_volatile_register,
-	.readable_register = rt5631_readable_register,
-	.reg_cache_step = 1,
 	.controls = rt5631_snd_controls,
 	.num_controls = ARRAY_SIZE(rt5631_snd_controls),
 	.dapm_widgets = rt5631_dapm_widgets,
@@ -1718,6 +1714,18 @@ static const struct i2c_device_id rt5631_i2c_id[] = {
 };
 MODULE_DEVICE_TABLE(i2c, rt5631_i2c_id);
 
+static const struct regmap_config rt5631_regmap_config = {
+	.reg_bits = 8,
+	.val_bits = 16,
+
+	.readable_reg = rt5631_readable_register,
+	.volatile_reg = rt5631_volatile_register,
+	.max_register = RT5631_VENDOR_ID2,
+	.reg_defaults = rt5631_reg,
+	.num_reg_defaults = ARRAY_SIZE(rt5631_reg),
+	.cache_type = REGCACHE_RBTREE,
+};
+
 static int rt5631_i2c_probe(struct i2c_client *i2c,
 		    const struct i2c_device_id *id)
 {
@@ -1731,6 +1739,10 @@ static int rt5631_i2c_probe(struct i2c_client *i2c,
 
 	i2c_set_clientdata(i2c, rt5631);
 
+	rt5631->regmap = devm_regmap_init_i2c(i2c, &rt5631_regmap_config);
+	if (IS_ERR(rt5631->regmap))
+		return PTR_ERR(rt5631->regmap);
+
 	ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5631,
 			rt5631_dai, ARRAY_SIZE(rt5631_dai));
 	return ret;
@@ -1752,17 +1764,7 @@ static struct i2c_driver rt5631_i2c_driver = {
 	.id_table = rt5631_i2c_id,
 };
 
-static int __init rt5631_modinit(void)
-{
-	return i2c_add_driver(&rt5631_i2c_driver);
-}
-module_init(rt5631_modinit);
-
-static void __exit rt5631_modexit(void)
-{
-	i2c_del_driver(&rt5631_i2c_driver);
-}
-module_exit(rt5631_modexit);
+module_i2c_driver(rt5631_i2c_driver);
 
 MODULE_DESCRIPTION("ASoC RT5631 driver");
 MODULE_AUTHOR("flove <flove@realtek.com>");
diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c
index c395ec370445..8af6a5245b18 100644
--- a/sound/soc/codecs/sgtl5000.c
+++ b/sound/soc/codecs/sgtl5000.c
@@ -84,8 +84,8 @@ static struct regulator_consumer_supply ldo_consumer[] = {
 
 static struct regulator_init_data ldo_init_data = {
 	.constraints = {
-		.min_uV                 = 850000,
-		.max_uV                 = 1600000,
+		.min_uV                 = 1200000,
+		.max_uV                 = 1200000,
 		.valid_modes_mask       = REGULATOR_MODE_NORMAL,
 		.valid_ops_mask         = REGULATOR_CHANGE_STATUS,
 	},
@@ -197,9 +197,9 @@ static const struct snd_soc_dapm_widget sgtl5000_dapm_widgets[] = {
 	SND_SOC_DAPM_OUTPUT("HP_OUT"),
 	SND_SOC_DAPM_OUTPUT("LINE_OUT"),
 
-	SND_SOC_DAPM_MICBIAS_E("Mic Bias", SGTL5000_CHIP_MIC_CTRL, 8, 0,
-				mic_bias_event,
-				SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
+	SND_SOC_DAPM_SUPPLY("Mic Bias", SGTL5000_CHIP_MIC_CTRL, 8, 0,
+			    mic_bias_event,
+			    SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
 
 	SND_SOC_DAPM_PGA("HP", SGTL5000_CHIP_ANA_POWER, 4, 0, NULL, 0),
 	SND_SOC_DAPM_PGA("LO", SGTL5000_CHIP_ANA_POWER, 0, 0, NULL, 0),
@@ -665,8 +665,7 @@ static int sgtl5000_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 snd_soc_codec *codec = dai->codec;
 	struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec);
 	int channels = params_channels(params);
 	int i2s_ctl = 0;
@@ -1455,17 +1454,7 @@ static struct i2c_driver sgtl5000_i2c_driver = {
 	.id_table = sgtl5000_id,
 };
 
-static int __init sgtl5000_modinit(void)
-{
-	return i2c_add_driver(&sgtl5000_i2c_driver);
-}
-module_init(sgtl5000_modinit);
-
-static void __exit sgtl5000_exit(void)
-{
-	i2c_del_driver(&sgtl5000_i2c_driver);
-}
-module_exit(sgtl5000_exit);
+module_i2c_driver(sgtl5000_i2c_driver);
 
 MODULE_DESCRIPTION("Freescale SGTL5000 ALSA SoC Codec Driver");
 MODULE_AUTHOR("Zeng Zhaoming <zengzm.kernel@gmail.com>");
diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c
index de2b20544ceb..079066fef425 100644
--- a/sound/soc/codecs/ssm2602.c
+++ b/sound/soc/codecs/ssm2602.c
@@ -33,6 +33,7 @@
 #include <linux/pm.h>
 #include <linux/i2c.h>
 #include <linux/spi/spi.h>
+#include <linux/regmap.h>
 #include <linux/slab.h>
 #include <sound/core.h>
 #include <sound/pcm.h>
@@ -43,8 +44,6 @@
 
 #include "ssm2602.h"
 
-#define SSM2602_VERSION "0.1"
-
 enum ssm2602_type {
 	SSM2602,
 	SSM2604,
@@ -53,10 +52,12 @@ enum ssm2602_type {
 /* codec private data */
 struct ssm2602_priv {
 	unsigned int sysclk;
-	enum snd_soc_control_type control_type;
+	struct snd_pcm_hw_constraint_list *sysclk_constraints;
 	struct snd_pcm_substream *master_substream;
 	struct snd_pcm_substream *slave_substream;
 
+	struct regmap *regmap;
+
 	enum ssm2602_type type;
 	unsigned int clk_out_pwr;
 };
@@ -73,7 +74,6 @@ static const u16 ssm2602_reg[SSM2602_CACHEREGNUM] = {
 	0x0000, 0x0000
 };
 
-#define ssm2602_reset(c)	snd_soc_write(c, SSM2602_RESET, 0)
 
 /*Appending several "None"s just for OSS mixer use*/
 static const char *ssm2602_input_select[] = {
@@ -195,6 +195,24 @@ static const struct snd_soc_dapm_route ssm2604_routes[] = {
 	{"ADC", NULL, "Line Input"},
 };
 
+static const unsigned int ssm2602_rates_12288000[] = {
+	8000, 32000, 48000, 96000,
+};
+
+static struct snd_pcm_hw_constraint_list ssm2602_constraints_12288000 = {
+	.list = ssm2602_rates_12288000,
+	.count = ARRAY_SIZE(ssm2602_rates_12288000),
+};
+
+static const unsigned int ssm2602_rates_11289600[] = {
+	8000, 44100, 88200,
+};
+
+static struct snd_pcm_hw_constraint_list ssm2602_constraints_11289600 = {
+	.list = ssm2602_rates_11289600,
+	.count = ARRAY_SIZE(ssm2602_rates_11289600),
+};
+
 struct ssm2602_coeff {
 	u32 mclk;
 	u32 rate;
@@ -254,11 +272,10 @@ static int ssm2602_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 snd_soc_codec *codec = dai->codec;
 	struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec);
-	u16 iface = snd_soc_read(codec, SSM2602_IFACE) & 0xfff3;
 	int srate = ssm2602_get_coeff(ssm2602->sysclk, params_rate(params));
+	unsigned int iface;
 
 	if (substream == ssm2602->slave_substream) {
 		dev_dbg(codec->dev, "Ignoring hw_params for slave substream\n");
@@ -268,31 +285,34 @@ static int ssm2602_hw_params(struct snd_pcm_substream *substream,
 	if (srate < 0)
 		return srate;
 
-	snd_soc_write(codec, SSM2602_SRATE, srate);
+	regmap_write(ssm2602->regmap, SSM2602_SRATE, srate);
 
 	/* bit size */
 	switch (params_format(params)) {
 	case SNDRV_PCM_FORMAT_S16_LE:
+		iface = 0x0;
 		break;
 	case SNDRV_PCM_FORMAT_S20_3LE:
-		iface |= 0x0004;
+		iface = 0x4;
 		break;
 	case SNDRV_PCM_FORMAT_S24_LE:
-		iface |= 0x0008;
+		iface = 0x8;
 		break;
 	case SNDRV_PCM_FORMAT_S32_LE:
-		iface |= 0x000c;
+		iface = 0xc;
 		break;
+	default:
+		return -EINVAL;
 	}
-	snd_soc_write(codec, SSM2602_IFACE, iface);
+	regmap_update_bits(ssm2602->regmap, SSM2602_IFACE,
+		IFACE_AUDIO_DATA_LEN, iface);
 	return 0;
 }
 
 static int ssm2602_startup(struct snd_pcm_substream *substream,
 			   struct snd_soc_dai *dai)
 {
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_codec *codec = rtd->codec;
+	struct snd_soc_codec *codec = dai->codec;
 	struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec);
 	struct snd_pcm_runtime *master_runtime;
 
@@ -322,14 +342,19 @@ static int ssm2602_startup(struct snd_pcm_substream *substream,
 	} else
 		ssm2602->master_substream = substream;
 
+	if (ssm2602->sysclk_constraints) {
+		snd_pcm_hw_constraint_list(substream->runtime, 0,
+				   SNDRV_PCM_HW_PARAM_RATE,
+				   ssm2602->sysclk_constraints);
+	}
+
 	return 0;
 }
 
 static void ssm2602_shutdown(struct snd_pcm_substream *substream,
 			     struct snd_soc_dai *dai)
 {
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_codec *codec = rtd->codec;
+	struct snd_soc_codec *codec = dai->codec;
 	struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec);
 
 	if (ssm2602->master_substream == substream)
@@ -341,14 +366,14 @@ static void ssm2602_shutdown(struct snd_pcm_substream *substream,
 
 static int ssm2602_mute(struct snd_soc_dai *dai, int mute)
 {
-	struct snd_soc_codec *codec = dai->codec;
+	struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(dai->codec);
 
 	if (mute)
-		snd_soc_update_bits(codec, SSM2602_APDIGI,
+		regmap_update_bits(ssm2602->regmap, SSM2602_APDIGI,
 				    APDIGI_ENABLE_DAC_MUTE,
 				    APDIGI_ENABLE_DAC_MUTE);
 	else
-		snd_soc_update_bits(codec, SSM2602_APDIGI,
+		regmap_update_bits(ssm2602->regmap, SSM2602_APDIGI,
 				    APDIGI_ENABLE_DAC_MUTE, 0);
 	return 0;
 }
@@ -364,16 +389,21 @@ static int ssm2602_set_dai_sysclk(struct snd_soc_dai *codec_dai,
 			return -EINVAL;
 
 		switch (freq) {
-		case 11289600:
-		case 12000000:
 		case 12288000:
-		case 16934400:
 		case 18432000:
-			ssm2602->sysclk = freq;
+			ssm2602->sysclk_constraints = &ssm2602_constraints_12288000;
+			break;
+		case 11289600:
+		case 16934400:
+			ssm2602->sysclk_constraints = &ssm2602_constraints_11289600;
+			break;
+		case 12000000:
+			ssm2602->sysclk_constraints = NULL;
 			break;
 		default:
 			return -EINVAL;
 		}
+		ssm2602->sysclk = freq;
 	} else {
 		unsigned int mask;
 
@@ -393,7 +423,7 @@ static int ssm2602_set_dai_sysclk(struct snd_soc_dai *codec_dai,
 		else
 			ssm2602->clk_out_pwr &= ~mask;
 
-		snd_soc_update_bits(codec, SSM2602_PWR,
+		regmap_update_bits(ssm2602->regmap, SSM2602_PWR,
 			PWR_CLK_OUT_PDN | PWR_OSC_PDN, ssm2602->clk_out_pwr);
 	}
 
@@ -403,8 +433,8 @@ static int ssm2602_set_dai_sysclk(struct snd_soc_dai *codec_dai,
 static int ssm2602_set_dai_fmt(struct snd_soc_dai *codec_dai,
 		unsigned int fmt)
 {
-	struct snd_soc_codec *codec = codec_dai->codec;
-	u16 iface = 0;
+	struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec_dai->codec);
+	unsigned int iface = 0;
 
 	/* set master/slave audio interface */
 	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
@@ -455,7 +485,7 @@ static int ssm2602_set_dai_fmt(struct snd_soc_dai *codec_dai,
 	}
 
 	/* set iface */
-	snd_soc_write(codec, SSM2602_IFACE, iface);
+	regmap_write(ssm2602->regmap, SSM2602_IFACE, iface);
 	return 0;
 }
 
@@ -467,7 +497,7 @@ static int ssm2602_set_bias_level(struct snd_soc_codec *codec,
 	switch (level) {
 	case SND_SOC_BIAS_ON:
 		/* vref/mid on, osc and clkout on if enabled */
-		snd_soc_update_bits(codec, SSM2602_PWR,
+		regmap_update_bits(ssm2602->regmap, SSM2602_PWR,
 			PWR_POWER_OFF | PWR_CLK_OUT_PDN | PWR_OSC_PDN,
 			ssm2602->clk_out_pwr);
 		break;
@@ -475,13 +505,13 @@ static int ssm2602_set_bias_level(struct snd_soc_codec *codec,
 		break;
 	case SND_SOC_BIAS_STANDBY:
 		/* everything off except vref/vmid, */
-		snd_soc_update_bits(codec, SSM2602_PWR,
+		regmap_update_bits(ssm2602->regmap, SSM2602_PWR,
 			PWR_POWER_OFF | PWR_CLK_OUT_PDN | PWR_OSC_PDN,
 			PWR_CLK_OUT_PDN | PWR_OSC_PDN);
 		break;
 	case SND_SOC_BIAS_OFF:
 		/* everything off */
-		snd_soc_update_bits(codec, SSM2602_PWR,
+		regmap_update_bits(ssm2602->regmap, SSM2602_PWR,
 			PWR_POWER_OFF, PWR_POWER_OFF);
 		break;
 
@@ -540,12 +570,13 @@ static int ssm2602_resume(struct snd_soc_codec *codec)
 
 static int ssm2602_probe(struct snd_soc_codec *codec)
 {
+	struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec);
 	struct snd_soc_dapm_context *dapm = &codec->dapm;
 	int ret;
 
-	snd_soc_update_bits(codec, SSM2602_LOUT1V,
+	regmap_update_bits(ssm2602->regmap, SSM2602_LOUT1V,
 			    LOUT1V_LRHP_BOTH, LOUT1V_LRHP_BOTH);
-	snd_soc_update_bits(codec, SSM2602_ROUT1V,
+	regmap_update_bits(ssm2602->regmap, SSM2602_ROUT1V,
 			    ROUT1V_RLHP_BOTH, ROUT1V_RLHP_BOTH);
 
 	ret = snd_soc_add_codec_controls(codec, ssm2602_snd_controls,
@@ -581,27 +612,26 @@ static int ssm260x_probe(struct snd_soc_codec *codec)
 	struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec);
 	int ret;
 
-	pr_info("ssm2602 Audio Codec %s", SSM2602_VERSION);
-
-	ret = snd_soc_codec_set_cache_io(codec, 7, 9, ssm2602->control_type);
+	codec->control_data = ssm2602->regmap;
+	ret = snd_soc_codec_set_cache_io(codec, 0, 0, SND_SOC_REGMAP);
 	if (ret < 0) {
 		dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
 		return ret;
 	}
 
-	ret = ssm2602_reset(codec);
+	ret = regmap_write(ssm2602->regmap, SSM2602_RESET, 0);
 	if (ret < 0) {
 		dev_err(codec->dev, "Failed to issue reset: %d\n", ret);
 		return ret;
 	}
 
 	/* set the update bits */
-	snd_soc_update_bits(codec, SSM2602_LINVOL,
+	regmap_update_bits(ssm2602->regmap, SSM2602_LINVOL,
 			    LINVOL_LRIN_BOTH, LINVOL_LRIN_BOTH);
-	snd_soc_update_bits(codec, SSM2602_RINVOL,
+	regmap_update_bits(ssm2602->regmap, SSM2602_RINVOL,
 			    RINVOL_RLIN_BOTH, RINVOL_RLIN_BOTH);
 	/*select Line in as default input*/
-	snd_soc_write(codec, SSM2602_APANA, APANA_SELECT_DAC |
+	regmap_write(ssm2602->regmap, SSM2602_APANA, APANA_SELECT_DAC |
 			APANA_ENABLE_MIC_BOOST);
 
 	switch (ssm2602->type) {
@@ -634,9 +664,6 @@ static struct snd_soc_codec_driver soc_codec_dev_ssm2602 = {
 	.suspend =	ssm2602_suspend,
 	.resume =	ssm2602_resume,
 	.set_bias_level = ssm2602_set_bias_level,
-	.reg_cache_size = ARRAY_SIZE(ssm2602_reg),
-	.reg_word_size = sizeof(u16),
-	.reg_cache_default = ssm2602_reg,
 
 	.controls = ssm260x_snd_controls,
 	.num_controls = ARRAY_SIZE(ssm260x_snd_controls),
@@ -646,6 +673,23 @@ static struct snd_soc_codec_driver soc_codec_dev_ssm2602 = {
 	.num_dapm_routes = ARRAY_SIZE(ssm260x_routes),
 };
 
+static bool ssm2602_register_volatile(struct device *dev, unsigned int reg)
+{
+	return reg == SSM2602_RESET;
+}
+
+static const struct regmap_config ssm2602_regmap_config = {
+	.val_bits = 9,
+	.reg_bits = 7,
+
+	.max_register = SSM2602_RESET,
+	.volatile_reg = ssm2602_register_volatile,
+
+	.cache_type = REGCACHE_RBTREE,
+	.reg_defaults_raw = ssm2602_reg,
+	.num_reg_defaults_raw = ARRAY_SIZE(ssm2602_reg),
+};
+
 #if defined(CONFIG_SPI_MASTER)
 static int __devinit ssm2602_spi_probe(struct spi_device *spi)
 {
@@ -658,9 +702,12 @@ static int __devinit ssm2602_spi_probe(struct spi_device *spi)
 		return -ENOMEM;
 
 	spi_set_drvdata(spi, ssm2602);
-	ssm2602->control_type = SND_SOC_SPI;
 	ssm2602->type = SSM2602;
 
+	ssm2602->regmap = devm_regmap_init_spi(spi, &ssm2602_regmap_config);
+	if (IS_ERR(ssm2602->regmap))
+		return PTR_ERR(ssm2602->regmap);
+
 	ret = snd_soc_register_codec(&spi->dev,
 			&soc_codec_dev_ssm2602, &ssm2602_dai, 1);
 	return ret;
@@ -701,9 +748,12 @@ static int __devinit ssm2602_i2c_probe(struct i2c_client *i2c,
 		return -ENOMEM;
 
 	i2c_set_clientdata(i2c, ssm2602);
-	ssm2602->control_type = SND_SOC_I2C;
 	ssm2602->type = id->driver_data;
 
+	ssm2602->regmap = devm_regmap_init_i2c(i2c, &ssm2602_regmap_config);
+	if (IS_ERR(ssm2602->regmap))
+		return PTR_ERR(ssm2602->regmap);
+
 	ret = snd_soc_register_codec(&i2c->dev,
 			&soc_codec_dev_ssm2602, &ssm2602_dai, 1);
 	return ret;
diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c
index 7db6fa515028..8d717f4b5a87 100644
--- a/sound/soc/codecs/sta32x.c
+++ b/sound/soc/codecs/sta32x.c
@@ -609,8 +609,7 @@ static int sta32x_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 snd_soc_codec *codec = dai->codec;
 	struct sta32x_priv *sta32x = snd_soc_codec_get_drvdata(codec);
 	unsigned int rate;
 	int i, mcs = -1, ir = -1;
diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c
index df1e07ffac32..31762ebdd774 100644
--- a/sound/soc/codecs/tlv320aic23.c
+++ b/sound/soc/codecs/tlv320aic23.c
@@ -34,8 +34,6 @@
 
 #include "tlv320aic23.h"
 
-#define AIC23_VERSION "0.1"
-
 /*
  * AIC23 register cache
  */
@@ -325,8 +323,7 @@ static int tlv320aic23_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 snd_soc_codec *codec = dai->codec;
 	u16 iface_reg;
 	int ret;
 	struct aic23 *aic23 = snd_soc_codec_get_drvdata(codec);
@@ -371,8 +368,7 @@ static int tlv320aic23_hw_params(struct snd_pcm_substream *substream,
 static int tlv320aic23_pcm_prepare(struct snd_pcm_substream *substream,
 				   struct snd_soc_dai *dai)
 {
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_codec *codec = rtd->codec;
+	struct snd_soc_codec *codec = dai->codec;
 
 	/* set active */
 	snd_soc_write(codec, TLV320AIC23_ACTIVE, 0x0001);
@@ -383,8 +379,7 @@ static int tlv320aic23_pcm_prepare(struct snd_pcm_substream *substream,
 static void tlv320aic23_shutdown(struct snd_pcm_substream *substream,
 				 struct snd_soc_dai *dai)
 {
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_codec *codec = rtd->codec;
+	struct snd_soc_codec *codec = dai->codec;
 	struct aic23 *aic23 = snd_soc_codec_get_drvdata(codec);
 
 	/* deactivate */
@@ -548,8 +543,6 @@ static int tlv320aic23_probe(struct snd_soc_codec *codec)
 	struct aic23 *aic23 = snd_soc_codec_get_drvdata(codec);
 	int ret;
 
-	printk(KERN_INFO "AIC23 Audio Codec %s\n", AIC23_VERSION);
-
 	ret = snd_soc_codec_set_cache_io(codec, 7, 9, aic23->control_type);
 	if (ret < 0) {
 		dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
diff --git a/sound/soc/codecs/tlv320aic26.c b/sound/soc/codecs/tlv320aic26.c
index 802064b5030d..85944e953578 100644
--- a/sound/soc/codecs/tlv320aic26.c
+++ b/sound/soc/codecs/tlv320aic26.c
@@ -126,8 +126,7 @@ static int aic26_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 snd_soc_codec *codec = dai->codec;
 	struct aic26 *aic26 = snd_soc_codec_get_drvdata(codec);
 	int fsref, divisor, wlen, pval, jval, dval, qval;
 	u16 reg;
diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index 8d20f6ec20f3..64d2a4fa34b2 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -802,8 +802,7 @@ static int aic3x_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 snd_soc_codec *codec = dai->codec;
 	struct aic3x_priv *aic3x = snd_soc_codec_get_drvdata(codec);
 	int codec_clk = 0, bypass_pll = 0, fsref, last_clk = 0;
 	u8 data, j, r, p, pll_q, pll_p = 1, pll_r = 1, pll_j = 1;
@@ -1161,24 +1160,6 @@ static int aic3x_set_bias_level(struct snd_soc_codec *codec,
 	return 0;
 }
 
-void aic3x_set_headset_detection(struct snd_soc_codec *codec, int detect,
-				 int headset_debounce, int button_debounce)
-{
-	u8 val;
-
-	val = ((detect & AIC3X_HEADSET_DETECT_MASK)
-		<< AIC3X_HEADSET_DETECT_SHIFT) |
-	      ((headset_debounce & AIC3X_HEADSET_DEBOUNCE_MASK)
-		<< AIC3X_HEADSET_DEBOUNCE_SHIFT) |
-	      ((button_debounce & AIC3X_BUTTON_DEBOUNCE_MASK)
-		<< AIC3X_BUTTON_DEBOUNCE_SHIFT);
-
-	if (detect & AIC3X_HEADSET_DETECT_MASK)
-		val |= AIC3X_HEADSET_DETECT_ENABLED;
-
-	snd_soc_write(codec, AIC3X_HEADSET_DETECT_CTRL_A, val);
-}
-
 #define AIC3X_RATES	SNDRV_PCM_RATE_8000_96000
 #define AIC3X_FORMATS	(SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \
 			 SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE)
diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c
index 4587ddd0fbf8..0dd41077ab79 100644
--- a/sound/soc/codecs/tlv320dac33.c
+++ b/sound/soc/codecs/tlv320dac33.c
@@ -62,8 +62,10 @@
 #define UTHR_FROM_PERIOD_SIZE(samples, playrate, burstrate) \
 	(((samples)*5000) / (((burstrate)*5000) / ((burstrate) - (playrate))))
 
-static void dac33_calculate_times(struct snd_pcm_substream *substream);
-static int dac33_prepare_chip(struct snd_pcm_substream *substream);
+static void dac33_calculate_times(struct snd_pcm_substream *substream,
+				  struct snd_soc_codec *codec);
+static int dac33_prepare_chip(struct snd_pcm_substream *substream,
+			      struct snd_soc_codec *codec);
 
 enum dac33_state {
 	DAC33_IDLE = 0,
@@ -427,8 +429,8 @@ static int dac33_playback_event(struct snd_soc_dapm_widget *w,
 	switch (event) {
 	case SND_SOC_DAPM_PRE_PMU:
 		if (likely(dac33->substream)) {
-			dac33_calculate_times(dac33->substream);
-			dac33_prepare_chip(dac33->substream);
+			dac33_calculate_times(dac33->substream, w->codec);
+			dac33_prepare_chip(dac33->substream, w->codec);
 		}
 		break;
 	case SND_SOC_DAPM_POST_PMD:
@@ -799,8 +801,7 @@ static void dac33_oscwait(struct snd_soc_codec *codec)
 static int dac33_startup(struct snd_pcm_substream *substream,
 			   struct snd_soc_dai *dai)
 {
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_codec *codec = rtd->codec;
+	struct snd_soc_codec *codec = dai->codec;
 	struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec);
 
 	/* Stream started, save the substream pointer */
@@ -812,8 +813,7 @@ static int dac33_startup(struct snd_pcm_substream *substream,
 static void dac33_shutdown(struct snd_pcm_substream *substream,
 			     struct snd_soc_dai *dai)
 {
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_codec *codec = rtd->codec;
+	struct snd_soc_codec *codec = dai->codec;
 	struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec);
 
 	dac33->substream = NULL;
@@ -825,8 +825,7 @@ static int dac33_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 snd_soc_codec *codec = dai->codec;
 	struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec);
 
 	/* Check parameters for validity */
@@ -868,10 +867,9 @@ static int dac33_hw_params(struct snd_pcm_substream *substream,
  * writes happens in different order, than dac33 might end up in unknown state.
  * Use the known, working sequence of register writes to initialize the dac33.
  */
-static int dac33_prepare_chip(struct snd_pcm_substream *substream)
+static int dac33_prepare_chip(struct snd_pcm_substream *substream,
+			      struct snd_soc_codec *codec)
 {
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_codec *codec = rtd->codec;
 	struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec);
 	unsigned int oscset, ratioset, pwr_ctrl, reg_tmp;
 	u8 aictrl_a, aictrl_b, fifoctrl_a;
@@ -1067,10 +1065,9 @@ static int dac33_prepare_chip(struct snd_pcm_substream *substream)
 	return 0;
 }
 
-static void dac33_calculate_times(struct snd_pcm_substream *substream)
+static void dac33_calculate_times(struct snd_pcm_substream *substream,
+				  struct snd_soc_codec *codec)
 {
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_codec *codec = rtd->codec;
 	struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec);
 	unsigned int period_size = substream->runtime->period_size;
 	unsigned int rate = substream->runtime->rate;
@@ -1128,8 +1125,7 @@ static void dac33_calculate_times(struct snd_pcm_substream *substream)
 static int dac33_pcm_trigger(struct snd_pcm_substream *substream, int cmd,
 			     struct snd_soc_dai *dai)
 {
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_codec *codec = rtd->codec;
+	struct snd_soc_codec *codec = dai->codec;
 	struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec);
 	int ret = 0;
 
@@ -1161,8 +1157,7 @@ static snd_pcm_sframes_t dac33_dai_delay(
 			struct snd_pcm_substream *substream,
 			struct snd_soc_dai *dai)
 {
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_codec *codec = rtd->codec;
+	struct snd_soc_codec *codec = dai->codec;
 	struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec);
 	unsigned long long t0, t1, t_now;
 	unsigned int time_delta, uthr;
diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c
index 170cf9a8fc79..391fcfc7b63b 100644
--- a/sound/soc/codecs/twl4030.c
+++ b/sound/soc/codecs/twl4030.c
@@ -1685,8 +1685,7 @@ static void twl4030_tdm_enable(struct snd_soc_codec *codec, int direction,
 static int twl4030_startup(struct snd_pcm_substream *substream,
 			   struct snd_soc_dai *dai)
 {
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_codec *codec = rtd->codec;
+	struct snd_soc_codec *codec = dai->codec;
 	struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec);
 
 	if (twl4030->master_substream) {
@@ -1715,8 +1714,7 @@ static int twl4030_startup(struct snd_pcm_substream *substream,
 static void twl4030_shutdown(struct snd_pcm_substream *substream,
 			     struct snd_soc_dai *dai)
 {
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_codec *codec = rtd->codec;
+	struct snd_soc_codec *codec = dai->codec;
 	struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec);
 
 	if (twl4030->master_substream == substream)
@@ -1740,8 +1738,7 @@ static int twl4030_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 snd_soc_codec *codec = dai->codec;
 	struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec);
 	u8 mode, old_mode, format, old_format;
 
@@ -1974,8 +1971,7 @@ static void twl4030_voice_enable(struct snd_soc_codec *codec, int direction,
 static int twl4030_voice_startup(struct snd_pcm_substream *substream,
 		struct snd_soc_dai *dai)
 {
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_codec *codec = rtd->codec;
+	struct snd_soc_codec *codec = dai->codec;
 	struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec);
 	u8 mode;
 
@@ -2007,8 +2003,7 @@ static int twl4030_voice_startup(struct snd_pcm_substream *substream,
 static void twl4030_voice_shutdown(struct snd_pcm_substream *substream,
 				struct snd_soc_dai *dai)
 {
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_codec *codec = rtd->codec;
+	struct snd_soc_codec *codec = dai->codec;
 
 	/* Enable voice digital filters */
 	twl4030_voice_enable(codec, substream->stream, 0);
@@ -2017,8 +2012,7 @@ static void twl4030_voice_shutdown(struct snd_pcm_substream *substream,
 static int twl4030_voice_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 snd_soc_codec *codec = dai->codec;
 	struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec);
 	u8 old_mode, mode;
 
diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c
index dc7509b9d53a..a36e9fcdf184 100644
--- a/sound/soc/codecs/twl6040.c
+++ b/sound/soc/codecs/twl6040.c
@@ -46,17 +46,6 @@
 #define TWL6040_OUTHF_0dB 0x03
 #define TWL6040_OUTHF_M52dB 0x1D
 
-#define TWL6040_RAMP_NONE	0
-#define TWL6040_RAMP_UP		1
-#define TWL6040_RAMP_DOWN	2
-
-#define TWL6040_HSL_VOL_MASK	0x0F
-#define TWL6040_HSL_VOL_SHIFT	0
-#define TWL6040_HSR_VOL_MASK	0xF0
-#define TWL6040_HSR_VOL_SHIFT	4
-#define TWL6040_HF_VOL_MASK	0x1F
-#define TWL6040_HF_VOL_SHIFT	0
-
 /* Shadow register used by the driver */
 #define TWL6040_REG_SW_SHADOW	0x2F
 #define TWL6040_CACHEREGNUM	(TWL6040_REG_SW_SHADOW + 1)
@@ -64,18 +53,6 @@
 /* TWL6040_REG_SW_SHADOW (0x2F) fields */
 #define TWL6040_EAR_PATH_ENABLE	0x01
 
-struct twl6040_output {
-	u16 active;
-	u16 left_vol;
-	u16 right_vol;
-	u16 left_step;
-	u16 right_step;
-	unsigned int step_delay;
-	u16 ramp;
-	struct delayed_work work;
-	struct completion ramp_done;
-};
-
 struct twl6040_jack_data {
 	struct snd_soc_jack *jack;
 	struct delayed_work work;
@@ -100,8 +77,6 @@ struct twl6040_data {
 	struct snd_soc_codec *codec;
 	struct workqueue_struct *workqueue;
 	struct mutex mutex;
-	struct twl6040_output headset;
-	struct twl6040_output handsfree;
 };
 
 /*
@@ -311,318 +286,6 @@ static void twl6040_restore_regs(struct snd_soc_codec *codec)
 	}
 }
 
-/*
- * Ramp HS PGA volume to minimise pops at stream startup and shutdown.
- */
-static inline int twl6040_hs_ramp_step(struct snd_soc_codec *codec,
-			unsigned int left_step, unsigned int right_step)
-{
-
-	struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec);
-	struct twl6040_output *headset = &priv->headset;
-	int left_complete = 0, right_complete = 0;
-	u8 reg, val;
-
-	/* left channel */
-	left_step = (left_step > 0xF) ? 0xF : left_step;
-	reg = twl6040_read_reg_cache(codec, TWL6040_REG_HSGAIN);
-	val = (~reg & TWL6040_HSL_VOL_MASK);
-
-	if (headset->ramp == TWL6040_RAMP_UP) {
-		/* ramp step up */
-		if (val < headset->left_vol) {
-			if (val + left_step > headset->left_vol)
-				val = headset->left_vol;
-			else
-				val += left_step;
-
-			reg &= ~TWL6040_HSL_VOL_MASK;
-			twl6040_write(codec, TWL6040_REG_HSGAIN,
-					(reg | (~val & TWL6040_HSL_VOL_MASK)));
-		} else {
-			left_complete = 1;
-		}
-	} else if (headset->ramp == TWL6040_RAMP_DOWN) {
-		/* ramp step down */
-		if (val > 0x0) {
-			if ((int)val - (int)left_step < 0)
-				val = 0;
-			else
-				val -= left_step;
-
-			reg &= ~TWL6040_HSL_VOL_MASK;
-			twl6040_write(codec, TWL6040_REG_HSGAIN, reg |
-						(~val & TWL6040_HSL_VOL_MASK));
-		} else {
-			left_complete = 1;
-		}
-	}
-
-	/* right channel */
-	right_step = (right_step > 0xF) ? 0xF : right_step;
-	reg = twl6040_read_reg_cache(codec, TWL6040_REG_HSGAIN);
-	val = (~reg & TWL6040_HSR_VOL_MASK) >> TWL6040_HSR_VOL_SHIFT;
-
-	if (headset->ramp == TWL6040_RAMP_UP) {
-		/* ramp step up */
-		if (val < headset->right_vol) {
-			if (val + right_step > headset->right_vol)
-				val = headset->right_vol;
-			else
-				val += right_step;
-
-			reg &= ~TWL6040_HSR_VOL_MASK;
-			twl6040_write(codec, TWL6040_REG_HSGAIN,
-				(reg | (~val << TWL6040_HSR_VOL_SHIFT)));
-		} else {
-			right_complete = 1;
-		}
-	} else if (headset->ramp == TWL6040_RAMP_DOWN) {
-		/* ramp step down */
-		if (val > 0x0) {
-			if ((int)val - (int)right_step < 0)
-				val = 0;
-			else
-				val -= right_step;
-
-			reg &= ~TWL6040_HSR_VOL_MASK;
-			twl6040_write(codec, TWL6040_REG_HSGAIN,
-					 reg | (~val << TWL6040_HSR_VOL_SHIFT));
-		} else {
-			right_complete = 1;
-		}
-	}
-
-	return left_complete & right_complete;
-}
-
-/*
- * Ramp HF PGA volume to minimise pops at stream startup and shutdown.
- */
-static inline int twl6040_hf_ramp_step(struct snd_soc_codec *codec,
-			unsigned int left_step, unsigned int right_step)
-{
-	struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec);
-	struct twl6040_output *handsfree = &priv->handsfree;
-	int left_complete = 0, right_complete = 0;
-	u16 reg, val;
-
-	/* left channel */
-	left_step = (left_step > 0x1D) ? 0x1D : left_step;
-	reg = twl6040_read_reg_cache(codec, TWL6040_REG_HFLGAIN);
-	reg = 0x1D - reg;
-	val = (reg & TWL6040_HF_VOL_MASK);
-	if (handsfree->ramp == TWL6040_RAMP_UP) {
-		/* ramp step up */
-		if (val < handsfree->left_vol) {
-			if (val + left_step > handsfree->left_vol)
-				val = handsfree->left_vol;
-			else
-				val += left_step;
-
-			reg &= ~TWL6040_HF_VOL_MASK;
-			twl6040_write(codec, TWL6040_REG_HFLGAIN,
-						reg | (0x1D - val));
-		} else {
-			left_complete = 1;
-		}
-	} else if (handsfree->ramp == TWL6040_RAMP_DOWN) {
-		/* ramp step down */
-		if (val > 0) {
-			if ((int)val - (int)left_step < 0)
-				val = 0;
-			else
-				val -= left_step;
-
-			reg &= ~TWL6040_HF_VOL_MASK;
-			twl6040_write(codec, TWL6040_REG_HFLGAIN,
-						reg | (0x1D - val));
-		} else {
-			left_complete = 1;
-		}
-	}
-
-	/* right channel */
-	right_step = (right_step > 0x1D) ? 0x1D : right_step;
-	reg = twl6040_read_reg_cache(codec, TWL6040_REG_HFRGAIN);
-	reg = 0x1D - reg;
-	val = (reg & TWL6040_HF_VOL_MASK);
-	if (handsfree->ramp == TWL6040_RAMP_UP) {
-		/* ramp step up */
-		if (val < handsfree->right_vol) {
-			if (val + right_step > handsfree->right_vol)
-				val = handsfree->right_vol;
-			else
-				val += right_step;
-
-			reg &= ~TWL6040_HF_VOL_MASK;
-			twl6040_write(codec, TWL6040_REG_HFRGAIN,
-						reg | (0x1D - val));
-		} else {
-			right_complete = 1;
-		}
-	} else if (handsfree->ramp == TWL6040_RAMP_DOWN) {
-		/* ramp step down */
-		if (val > 0) {
-			if ((int)val - (int)right_step < 0)
-				val = 0;
-			else
-				val -= right_step;
-
-			reg &= ~TWL6040_HF_VOL_MASK;
-			twl6040_write(codec, TWL6040_REG_HFRGAIN,
-						reg | (0x1D - val));
-		}
-	}
-
-	return left_complete & right_complete;
-}
-
-/*
- * This work ramps both output PGAs at stream start/stop time to
- * minimise pop associated with DAPM power switching.
- */
-static void twl6040_pga_hs_work(struct work_struct *work)
-{
-	struct twl6040_data *priv =
-		container_of(work, struct twl6040_data, headset.work.work);
-	struct snd_soc_codec *codec = priv->codec;
-	struct twl6040_output *headset = &priv->headset;
-	int i, headset_complete;
-
-	/* do we need to ramp at all ? */
-	if (headset->ramp == TWL6040_RAMP_NONE)
-		return;
-
-	/* HS PGA gain range: 0x0 - 0xf (0 - 15) */
-	for (i = 0; i < 16; i++) {
-		headset_complete = twl6040_hs_ramp_step(codec,
-						headset->left_step,
-						headset->right_step);
-
-		/* ramp finished ? */
-		if (headset_complete)
-			break;
-
-		schedule_timeout_interruptible(
-				msecs_to_jiffies(headset->step_delay));
-	}
-
-	if (headset->ramp == TWL6040_RAMP_DOWN) {
-		headset->active = 0;
-		complete(&headset->ramp_done);
-	} else {
-		headset->active = 1;
-	}
-	headset->ramp = TWL6040_RAMP_NONE;
-}
-
-static void twl6040_pga_hf_work(struct work_struct *work)
-{
-	struct twl6040_data *priv =
-		container_of(work, struct twl6040_data, handsfree.work.work);
-	struct snd_soc_codec *codec = priv->codec;
-	struct twl6040_output *handsfree = &priv->handsfree;
-	int i, handsfree_complete;
-
-	/* do we need to ramp at all ? */
-	if (handsfree->ramp == TWL6040_RAMP_NONE)
-		return;
-
-	/*
-	 * HF PGA gain range: 0x00 - 0x1d (0 - 29) */
-	for (i = 0; i < 30; i++) {
-		handsfree_complete = twl6040_hf_ramp_step(codec,
-						handsfree->left_step,
-						handsfree->right_step);
-
-		/* ramp finished ? */
-		if (handsfree_complete)
-			break;
-
-		schedule_timeout_interruptible(
-				msecs_to_jiffies(handsfree->step_delay));
-	}
-
-
-	if (handsfree->ramp == TWL6040_RAMP_DOWN) {
-		handsfree->active = 0;
-		complete(&handsfree->ramp_done);
-	} else
-		handsfree->active = 1;
-	handsfree->ramp = TWL6040_RAMP_NONE;
-}
-
-static int out_drv_event(struct snd_soc_dapm_widget *w,
-			struct snd_kcontrol *kcontrol, int event)
-{
-	struct snd_soc_codec *codec = w->codec;
-	struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec);
-	struct twl6040_output *out;
-	struct delayed_work *work;
-
-	switch (w->shift) {
-	case 2: /* Headset output driver */
-		out = &priv->headset;
-		work = &out->work;
-		/*
-		 * Make sure, that we do not mess up variables for already
-		 * executing work.
-		 */
-		cancel_delayed_work_sync(work);
-
-		out->left_step = priv->hs_left_step;
-		out->right_step = priv->hs_right_step;
-		out->step_delay = 5;	/* 5 ms between volume ramp steps */
-		break;
-	case 4: /* Handsfree output driver */
-		out = &priv->handsfree;
-		work = &out->work;
-		/*
-		 * Make sure, that we do not mess up variables for already
-		 * executing work.
-		 */
-		cancel_delayed_work_sync(work);
-
-		out->left_step = priv->hf_left_step;
-		out->right_step = priv->hf_right_step;
-		out->step_delay = 5;	/* 5 ms between volume ramp steps */
-		break;
-	default:
-		return -1;
-	}
-
-	switch (event) {
-	case SND_SOC_DAPM_POST_PMU:
-		if (out->active)
-			break;
-
-		/* don't use volume ramp for power-up */
-		out->ramp = TWL6040_RAMP_UP;
-		out->left_step = out->left_vol;
-		out->right_step = out->right_vol;
-
-		queue_delayed_work(priv->workqueue, work, msecs_to_jiffies(1));
-		break;
-
-	case SND_SOC_DAPM_PRE_PMD:
-		if (!out->active)
-			break;
-
-		/* use volume ramp for power-down */
-		out->ramp = TWL6040_RAMP_DOWN;
-		INIT_COMPLETION(out->ramp_done);
-
-		queue_delayed_work(priv->workqueue, work, msecs_to_jiffies(1));
-
-		wait_for_completion_timeout(&out->ramp_done,
-					    msecs_to_jiffies(2000));
-		break;
-	}
-
-	return 0;
-}
-
 /* set headset dac and driver power mode */
 static int headset_power_mode(struct snd_soc_codec *codec, int high_perf)
 {
@@ -747,71 +410,6 @@ static irqreturn_t twl6040_audio_handler(int irq, void *data)
 	return IRQ_HANDLED;
 }
 
-static int twl6040_put_volsw(struct snd_kcontrol *kcontrol,
-				  struct snd_ctl_elem_value *ucontrol)
-{
-	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
-	struct twl6040_data *twl6040_priv = snd_soc_codec_get_drvdata(codec);
-	struct twl6040_output *out = NULL;
-	struct soc_mixer_control *mc =
-		(struct soc_mixer_control *)kcontrol->private_value;
-	int ret;
-
-	/* For HS and HF we shadow the values and only actually write
-	 * them out when active in order to ensure the amplifier comes on
-	 * as quietly as possible. */
-	switch (mc->reg) {
-	case TWL6040_REG_HSGAIN:
-		out = &twl6040_priv->headset;
-		break;
-	case TWL6040_REG_HFLGAIN:
-		out = &twl6040_priv->handsfree;
-		break;
-	default:
-		dev_warn(codec->dev, "%s: Unexpected register: 0x%02x\n",
-					__func__, mc->reg);
-		return -EINVAL;
-	}
-
-	out->left_vol = ucontrol->value.integer.value[0];
-	out->right_vol = ucontrol->value.integer.value[1];
-	if (!out->active)
-		return 1;
-
-	ret = snd_soc_put_volsw(kcontrol, ucontrol);
-	if (ret < 0)
-		return ret;
-
-	return 1;
-}
-
-static int twl6040_get_volsw(struct snd_kcontrol *kcontrol,
-				struct snd_ctl_elem_value *ucontrol)
-{
-	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
-	struct twl6040_data *twl6040_priv = snd_soc_codec_get_drvdata(codec);
-	struct twl6040_output *out = &twl6040_priv->headset;
-	struct soc_mixer_control *mc =
-		(struct soc_mixer_control *)kcontrol->private_value;
-
-	switch (mc->reg) {
-	case TWL6040_REG_HSGAIN:
-		out = &twl6040_priv->headset;
-		break;
-	case TWL6040_REG_HFLGAIN:
-		out = &twl6040_priv->handsfree;
-		break;
-	default:
-		dev_warn(codec->dev, "%s: Unexpected register: 0x%02x\n",
-					__func__, mc->reg);
-		return -EINVAL;
-	}
-
-	ucontrol->value.integer.value[0] = out->left_vol;
-	ucontrol->value.integer.value[1] = out->right_vol;
-	return 0;
-}
-
 static int twl6040_soc_dapm_put_vibra_enum(struct snd_kcontrol *kcontrol,
 	struct snd_ctl_elem_value *ucontrol)
 {
@@ -1076,12 +674,10 @@ static const struct snd_kcontrol_new twl6040_snd_controls[] = {
 		TWL6040_REG_LINEGAIN, 0, 3, 7, 0, afm_amp_tlv),
 
 	/* Playback gains */
-	SOC_DOUBLE_EXT_TLV("Headset Playback Volume",
-		TWL6040_REG_HSGAIN, 0, 4, 0xF, 1, twl6040_get_volsw,
-		twl6040_put_volsw, hs_tlv),
-	SOC_DOUBLE_R_EXT_TLV("Handsfree Playback Volume",
-		TWL6040_REG_HFLGAIN, TWL6040_REG_HFRGAIN, 0, 0x1D, 1,
-		twl6040_get_volsw, twl6040_put_volsw, hf_tlv),
+	SOC_DOUBLE_TLV("Headset Playback Volume",
+		TWL6040_REG_HSGAIN, 0, 4, 0xF, 1, hs_tlv),
+	SOC_DOUBLE_R_TLV("Handsfree Playback Volume",
+		TWL6040_REG_HFLGAIN, TWL6040_REG_HFRGAIN, 0, 0x1D, 1, hf_tlv),
 	SOC_SINGLE_TLV("Earphone Playback Volume",
 		TWL6040_REG_EARCTL, 1, 0xF, 1, ep_tlv),
 
@@ -1180,22 +776,14 @@ static const struct snd_soc_dapm_widget twl6040_dapm_widgets[] = {
 			&auxr_switch_control),
 
 	/* Analog playback drivers */
-	SND_SOC_DAPM_OUT_DRV_E("HF Left Driver",
-			TWL6040_REG_HFLCTL, 4, 0, NULL, 0,
-			out_drv_event,
-			SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
-	SND_SOC_DAPM_OUT_DRV_E("HF Right Driver",
-			TWL6040_REG_HFRCTL, 4, 0, NULL, 0,
-			out_drv_event,
-			SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
-	SND_SOC_DAPM_OUT_DRV_E("HS Left Driver",
-			TWL6040_REG_HSLCTL, 2, 0, NULL, 0,
-			out_drv_event,
-			SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
-	SND_SOC_DAPM_OUT_DRV_E("HS Right Driver",
-			TWL6040_REG_HSRCTL, 2, 0, NULL, 0,
-			out_drv_event,
-			SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
+	SND_SOC_DAPM_OUT_DRV("HF Left Driver",
+			TWL6040_REG_HFLCTL, 4, 0, NULL, 0),
+	SND_SOC_DAPM_OUT_DRV("HF Right Driver",
+			TWL6040_REG_HFRCTL, 4, 0, NULL, 0),
+	SND_SOC_DAPM_OUT_DRV("HS Left Driver",
+			TWL6040_REG_HSLCTL, 2, 0, NULL, 0),
+	SND_SOC_DAPM_OUT_DRV("HS Right Driver",
+			TWL6040_REG_HSRCTL, 2, 0, NULL, 0),
 	SND_SOC_DAPM_OUT_DRV_E("Earphone Driver",
 			TWL6040_REG_EARCTL, 0, 0, NULL, 0,
 			twl6040_ep_drv_event,
@@ -1339,8 +927,7 @@ static int twl6040_set_bias_level(struct snd_soc_codec *codec,
 static int twl6040_startup(struct snd_pcm_substream *substream,
 			struct snd_soc_dai *dai)
 {
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_codec *codec = rtd->codec;
+	struct snd_soc_codec *codec = dai->codec;
 	struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec);
 
 	snd_pcm_hw_constraint_list(substream->runtime, 0,
@@ -1354,8 +941,7 @@ static int twl6040_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 snd_soc_codec *codec = dai->codec;
 	struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec);
 	int rate;
 
@@ -1391,8 +977,7 @@ static int twl6040_hw_params(struct snd_pcm_substream *substream,
 static int twl6040_prepare(struct snd_pcm_substream *substream,
 			struct snd_soc_dai *dai)
 {
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_codec *codec = rtd->codec;
+	struct snd_soc_codec *codec = dai->codec;
 	struct twl6040 *twl6040 = codec->control_data;
 	struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec);
 	int ret;
@@ -1570,14 +1155,9 @@ static int twl6040_probe(struct snd_soc_codec *codec)
 	}
 
 	INIT_DELAYED_WORK(&priv->hs_jack.work, twl6040_accessory_work);
-	INIT_DELAYED_WORK(&priv->headset.work, twl6040_pga_hs_work);
-	INIT_DELAYED_WORK(&priv->handsfree.work, twl6040_pga_hf_work);
 
 	mutex_init(&priv->mutex);
 
-	init_completion(&priv->headset.ramp_done);
-	init_completion(&priv->handsfree.ramp_done);
-
 	ret = request_threaded_irq(priv->plug_irq, NULL, twl6040_audio_handler,
 				   0, "twl6040_irq_plug", codec);
 	if (ret) {
diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c
index 797b0dde2c68..6c3d43b8ee85 100644
--- a/sound/soc/codecs/uda134x.c
+++ b/sound/soc/codecs/uda134x.c
@@ -159,8 +159,7 @@ static int uda134x_mute(struct snd_soc_dai *dai, int mute)
 static int uda134x_startup(struct snd_pcm_substream *substream,
 	struct snd_soc_dai *dai)
 {
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_codec *codec =rtd->codec;
+	struct snd_soc_codec *codec = dai->codec;
 	struct uda134x_priv *uda134x = snd_soc_codec_get_drvdata(codec);
 	struct snd_pcm_runtime *master_runtime;
 
@@ -191,8 +190,7 @@ static int uda134x_startup(struct snd_pcm_substream *substream,
 static void uda134x_shutdown(struct snd_pcm_substream *substream,
 	struct snd_soc_dai *dai)
 {
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_codec *codec = rtd->codec;
+	struct snd_soc_codec *codec = dai->codec;
 	struct uda134x_priv *uda134x = snd_soc_codec_get_drvdata(codec);
 
 	if (uda134x->master_substream == substream)
diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c
index 4f1b23d7e404..2502214b84ab 100644
--- a/sound/soc/codecs/uda1380.c
+++ b/sound/soc/codecs/uda1380.c
@@ -502,8 +502,7 @@ static int uda1380_set_dai_fmt_capture(struct snd_soc_dai *codec_dai,
 static int uda1380_trigger(struct snd_pcm_substream *substream, int cmd,
 		struct snd_soc_dai *dai)
 {
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_codec *codec = rtd->codec;
+	struct snd_soc_codec *codec = dai->codec;
 	struct uda1380_priv *uda1380 = snd_soc_codec_get_drvdata(codec);
 	int mixer = uda1380_read_reg_cache(codec, UDA1380_MIXER);
 
@@ -528,8 +527,7 @@ static int uda1380_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 snd_soc_codec *codec = dai->codec;
 	u16 clk = uda1380_read_reg_cache(codec, UDA1380_CLK);
 
 	/* set WSPLL power and divider if running from this clock */
diff --git a/sound/soc/codecs/wl1273.c b/sound/soc/codecs/wl1273.c
index 3d868dc40092..7b24d6d192e1 100644
--- a/sound/soc/codecs/wl1273.c
+++ b/sound/soc/codecs/wl1273.c
@@ -293,8 +293,7 @@ static const struct snd_kcontrol_new wl1273_controls[] = {
 static int wl1273_startup(struct snd_pcm_substream *substream,
 			  struct snd_soc_dai *dai)
 {
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_codec *codec = rtd->codec;
+	struct snd_soc_codec *codec = dai->codec;
 	struct wl1273_priv *wl1273 = snd_soc_codec_get_drvdata(codec);
 
 	switch (wl1273->mode) {
@@ -329,8 +328,7 @@ static int wl1273_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 wl1273_priv *wl1273 = snd_soc_codec_get_drvdata(rtd->codec);
+	struct wl1273_priv *wl1273 = snd_soc_codec_get_drvdata(dai->codec);
 	struct wl1273_core *core = wl1273->core;
 	unsigned int rate, width, r;
 
diff --git a/sound/soc/codecs/wm1250-ev1.c b/sound/soc/codecs/wm1250-ev1.c
index aefb4f89be0e..e0b51e9f8b12 100644
--- a/sound/soc/codecs/wm1250-ev1.c
+++ b/sound/soc/codecs/wm1250-ev1.c
@@ -79,22 +79,65 @@ static const struct snd_soc_dapm_route wm1250_ev1_dapm_routes[] = {
 	{ "WM1250 Output", NULL, "DAC" },
 };
 
+static int wm1250_ev1_hw_params(struct snd_pcm_substream *substream,
+				struct snd_pcm_hw_params *params,
+				struct snd_soc_dai *dai)
+{
+	struct wm1250_priv *wm1250 = snd_soc_codec_get_drvdata(dai->codec);
+
+	switch (params_rate(params)) {
+	case 8000:
+		gpio_set_value(wm1250->gpios[WM1250_EV1_GPIO_CLK_SEL0].gpio,
+			       1);
+		gpio_set_value(wm1250->gpios[WM1250_EV1_GPIO_CLK_SEL1].gpio,
+			       1);
+		break;
+	case 16000:
+		gpio_set_value(wm1250->gpios[WM1250_EV1_GPIO_CLK_SEL0].gpio,
+			       0);
+		gpio_set_value(wm1250->gpios[WM1250_EV1_GPIO_CLK_SEL1].gpio,
+			       1);
+		break;
+	case 32000:
+		gpio_set_value(wm1250->gpios[WM1250_EV1_GPIO_CLK_SEL0].gpio,
+			       1);
+		gpio_set_value(wm1250->gpios[WM1250_EV1_GPIO_CLK_SEL1].gpio,
+			       0);
+		break;
+	case 64000:
+		gpio_set_value(wm1250->gpios[WM1250_EV1_GPIO_CLK_SEL0].gpio,
+			       0);
+		gpio_set_value(wm1250->gpios[WM1250_EV1_GPIO_CLK_SEL1].gpio,
+			       0);
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	return 0;
+}
+
+static const struct snd_soc_dai_ops wm1250_ev1_ops = {
+	.hw_params = wm1250_ev1_hw_params,
+};
+
 static struct snd_soc_dai_driver wm1250_ev1_dai = {
 	.name = "wm1250-ev1",
 	.playback = {
 		.stream_name = "Playback",
 		.channels_min = 1,
-		.channels_max = 1,
+		.channels_max = 2,
 		.rates = SNDRV_PCM_RATE_8000,
 		.formats = SNDRV_PCM_FMTBIT_S16_LE,
 	},
 	.capture = {
 		.stream_name = "Capture",
 		.channels_min = 1,
-		.channels_max = 1,
+		.channels_max = 2,
 		.rates = SNDRV_PCM_RATE_8000,
 		.formats = SNDRV_PCM_FMTBIT_S16_LE,
 	},
+	.ops = &wm1250_ev1_ops,
 };
 
 static struct snd_soc_codec_driver soc_codec_dev_wm1250_ev1 = {
@@ -215,23 +258,7 @@ static struct i2c_driver wm1250_ev1_i2c_driver = {
 	.id_table = wm1250_ev1_i2c_id,
 };
 
-static int __init wm1250_ev1_modinit(void)
-{
-	int ret = 0;
-
-	ret = i2c_add_driver(&wm1250_ev1_i2c_driver);
-	if (ret != 0)
-		pr_err("Failed to register WM1250-EV1 I2C driver: %d\n", ret);
-
-	return ret;
-}
-module_init(wm1250_ev1_modinit);
-
-static void __exit wm1250_ev1_exit(void)
-{
-	i2c_del_driver(&wm1250_ev1_i2c_driver);
-}
-module_exit(wm1250_ev1_exit);
+module_i2c_driver(wm1250_ev1_i2c_driver);
 
 MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>");
 MODULE_DESCRIPTION("WM1250-EV1 audio I/O module driver");
diff --git a/sound/soc/codecs/wm5100-tables.c b/sound/soc/codecs/wm5100-tables.c
index 9a18fae68204..e167207a19cc 100644
--- a/sound/soc/codecs/wm5100-tables.c
+++ b/sound/soc/codecs/wm5100-tables.c
@@ -32,7 +32,18 @@ bool wm5100_volatile_register(struct device *dev, unsigned int reg)
 	case WM5100_MIC_DETECT_3:
 		return 1;
 	default:
-		return 0;
+		if ((reg >= WM5100_DSP1_PM_0 && reg <= WM5100_DSP1_PM_1535) ||
+		    (reg >= WM5100_DSP1_ZM_0 && reg <= WM5100_DSP1_ZM_2047) ||
+		    (reg >= WM5100_DSP1_DM_0 && reg <= WM5100_DSP1_DM_511) ||
+		    (reg >= WM5100_DSP2_PM_0 && reg <= WM5100_DSP2_PM_1535) ||
+		    (reg >= WM5100_DSP2_ZM_0 && reg <= WM5100_DSP2_ZM_2047) ||
+		    (reg >= WM5100_DSP2_DM_0 && reg <= WM5100_DSP2_DM_511) ||
+		    (reg >= WM5100_DSP3_PM_0 && reg <= WM5100_DSP3_PM_1535) ||
+		    (reg >= WM5100_DSP3_ZM_0 && reg <= WM5100_DSP3_ZM_2047) ||
+		    (reg >= WM5100_DSP3_DM_0 && reg <= WM5100_DSP3_DM_511))
+			return 1;
+		else
+			return 0;
 	}
 }
 
@@ -697,9 +708,110 @@ bool wm5100_readable_register(struct device *dev, unsigned int reg)
 	case WM5100_HPLPF3_2:
 	case WM5100_HPLPF4_1:
 	case WM5100_HPLPF4_2:
+	case WM5100_DSP1_CONTROL_1:
+	case WM5100_DSP1_CONTROL_2:
+	case WM5100_DSP1_CONTROL_3:
+	case WM5100_DSP1_CONTROL_4:
+	case WM5100_DSP1_CONTROL_5:
+	case WM5100_DSP1_CONTROL_6:
+	case WM5100_DSP1_CONTROL_7:
+	case WM5100_DSP1_CONTROL_8:
+	case WM5100_DSP1_CONTROL_9:
+	case WM5100_DSP1_CONTROL_10:
+	case WM5100_DSP1_CONTROL_11:
+	case WM5100_DSP1_CONTROL_12:
+	case WM5100_DSP1_CONTROL_13:
+	case WM5100_DSP1_CONTROL_14:
+	case WM5100_DSP1_CONTROL_15:
+	case WM5100_DSP1_CONTROL_16:
+	case WM5100_DSP1_CONTROL_17:
+	case WM5100_DSP1_CONTROL_18:
+	case WM5100_DSP1_CONTROL_19:
+	case WM5100_DSP1_CONTROL_20:
+	case WM5100_DSP1_CONTROL_21:
+	case WM5100_DSP1_CONTROL_22:
+	case WM5100_DSP1_CONTROL_23:
+	case WM5100_DSP1_CONTROL_24:
+	case WM5100_DSP1_CONTROL_25:
+	case WM5100_DSP1_CONTROL_26:
+	case WM5100_DSP1_CONTROL_27:
+	case WM5100_DSP1_CONTROL_28:
+	case WM5100_DSP1_CONTROL_29:
+	case WM5100_DSP1_CONTROL_30:
+	case WM5100_DSP2_CONTROL_1:
+	case WM5100_DSP2_CONTROL_2:
+	case WM5100_DSP2_CONTROL_3:
+	case WM5100_DSP2_CONTROL_4:
+	case WM5100_DSP2_CONTROL_5:
+	case WM5100_DSP2_CONTROL_6:
+	case WM5100_DSP2_CONTROL_7:
+	case WM5100_DSP2_CONTROL_8:
+	case WM5100_DSP2_CONTROL_9:
+	case WM5100_DSP2_CONTROL_10:
+	case WM5100_DSP2_CONTROL_11:
+	case WM5100_DSP2_CONTROL_12:
+	case WM5100_DSP2_CONTROL_13:
+	case WM5100_DSP2_CONTROL_14:
+	case WM5100_DSP2_CONTROL_15:
+	case WM5100_DSP2_CONTROL_16:
+	case WM5100_DSP2_CONTROL_17:
+	case WM5100_DSP2_CONTROL_18:
+	case WM5100_DSP2_CONTROL_19:
+	case WM5100_DSP2_CONTROL_20:
+	case WM5100_DSP2_CONTROL_21:
+	case WM5100_DSP2_CONTROL_22:
+	case WM5100_DSP2_CONTROL_23:
+	case WM5100_DSP2_CONTROL_24:
+	case WM5100_DSP2_CONTROL_25:
+	case WM5100_DSP2_CONTROL_26:
+	case WM5100_DSP2_CONTROL_27:
+	case WM5100_DSP2_CONTROL_28:
+	case WM5100_DSP2_CONTROL_29:
+	case WM5100_DSP2_CONTROL_30:
+	case WM5100_DSP3_CONTROL_1:
+	case WM5100_DSP3_CONTROL_2:
+	case WM5100_DSP3_CONTROL_3:
+	case WM5100_DSP3_CONTROL_4:
+	case WM5100_DSP3_CONTROL_5:
+	case WM5100_DSP3_CONTROL_6:
+	case WM5100_DSP3_CONTROL_7:
+	case WM5100_DSP3_CONTROL_8:
+	case WM5100_DSP3_CONTROL_9:
+	case WM5100_DSP3_CONTROL_10:
+	case WM5100_DSP3_CONTROL_11:
+	case WM5100_DSP3_CONTROL_12:
+	case WM5100_DSP3_CONTROL_13:
+	case WM5100_DSP3_CONTROL_14:
+	case WM5100_DSP3_CONTROL_15:
+	case WM5100_DSP3_CONTROL_16:
+	case WM5100_DSP3_CONTROL_17:
+	case WM5100_DSP3_CONTROL_18:
+	case WM5100_DSP3_CONTROL_19:
+	case WM5100_DSP3_CONTROL_20:
+	case WM5100_DSP3_CONTROL_21:
+	case WM5100_DSP3_CONTROL_22:
+	case WM5100_DSP3_CONTROL_23:
+	case WM5100_DSP3_CONTROL_24:
+	case WM5100_DSP3_CONTROL_25:
+	case WM5100_DSP3_CONTROL_26:
+	case WM5100_DSP3_CONTROL_27:
+	case WM5100_DSP3_CONTROL_28:
+	case WM5100_DSP3_CONTROL_29:
+	case WM5100_DSP3_CONTROL_30:
 		return 1;
 	default:
-		return 0;
+		if ((reg >= WM5100_DSP1_PM_0 && reg <= WM5100_DSP1_PM_1535) ||
+		    (reg >= WM5100_DSP1_ZM_0 && reg <= WM5100_DSP1_ZM_2047) ||
+		    (reg >= WM5100_DSP1_DM_0 && reg <= WM5100_DSP1_DM_511) ||
+		    (reg >= WM5100_DSP2_PM_0 && reg <= WM5100_DSP2_PM_1535) ||
+		    (reg >= WM5100_DSP2_ZM_0 && reg <= WM5100_DSP2_ZM_2047) ||
+		    (reg >= WM5100_DSP2_DM_0 && reg <= WM5100_DSP2_DM_511) ||
+		    (reg >= WM5100_DSP3_PM_0 && reg <= WM5100_DSP3_PM_1535) ||
+		    (reg >= WM5100_DSP3_ZM_0 && reg <= WM5100_DSP3_ZM_2047) ||
+		    (reg >= WM5100_DSP3_DM_0 && reg <= WM5100_DSP3_DM_511))
+			return 1;
+		else
+			return 0;
 	}
 }
 
@@ -1361,4 +1473,13 @@ struct reg_default wm5100_reg_defaults[WM5100_REGISTER_COUNT] = {
 	{ 0x0EC9, 0x0000 },  /* R3785  - HPLPF3_2 */
 	{ 0x0ECC, 0x0000 },  /* R3788  - HPLPF4_1 */
 	{ 0x0ECD, 0x0000 },  /* R3789  - HPLPF4_2 */
+	{ 0x0F02, 0x0000 },  /* R3842  - DSP1 Control 2 */
+	{ 0x0F03, 0x0000 },  /* R3843  - DSP1 Control 3 */
+	{ 0x0F04, 0x0000 },  /* R3844  - DSP1 Control 4 */
+	{ 0x1002, 0x0000 },  /* R4098  - DSP2 Control 2 */
+	{ 0x1003, 0x0000 },  /* R4099  - DSP2 Control 3 */
+	{ 0x1004, 0x0000 },  /* R4100  - DSP2 Control 4 */
+	{ 0x1102, 0x0000 },  /* R4354  - DSP3 Control 2 */
+	{ 0x1103, 0x0000 },  /* R4355  - DSP3 Control 3 */
+	{ 0x1104, 0x0000 },  /* R4356  - DSP3 Control 4 */
 };
diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c
index b9c185ce64e4..cb6d5372103a 100644
--- a/sound/soc/codecs/wm5100.c
+++ b/sound/soc/codecs/wm5100.c
@@ -1265,29 +1265,12 @@ static const __devinitdata struct reg_default wm5100_reva_patches[] = {
 	{ WM5100_AUDIO_IF_3_19, 1 },
 };
 
-static int wm5100_dai_to_base(struct snd_soc_dai *dai)
-{
-	switch (dai->id) {
-	case 0:
-		return WM5100_AUDIO_IF_1_1 - 1;
-	case 1:
-		return WM5100_AUDIO_IF_2_1 - 1;
-	case 2:
-		return WM5100_AUDIO_IF_3_1 - 1;
-	default:
-		BUG();
-		return -EINVAL;
-	}
-}
-
 static int wm5100_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
 {
 	struct snd_soc_codec *codec = dai->codec;
 	int lrclk, bclk, mask, base;
 
-	base = wm5100_dai_to_base(dai);
-	if (base < 0)
-		return base;
+	base = dai->driver->base;
 
 	lrclk = 0;
 	bclk = 0;
@@ -1414,9 +1397,7 @@ static int wm5100_hw_params(struct snd_pcm_substream *substream,
 	int i, base, bclk, aif_rate, lrclk, wl, fl, sr;
 	int *bclk_rates;
 
-	base = wm5100_dai_to_base(dai);
-	if (base < 0)
-		return base;
+	base = dai->driver->base;
 
 	/* Data sizes if not using TDM */
 	wl = snd_pcm_format_width(params_format(params));
@@ -1897,6 +1878,7 @@ static int wm5100_set_fll(struct snd_soc_codec *codec, int fll_id, int source,
 static struct snd_soc_dai_driver wm5100_dai[] = {
 	{
 		.name = "wm5100-aif1",
+		.base = WM5100_AUDIO_IF_1_1 - 1,
 		.playback = {
 			.stream_name = "AIF1 Playback",
 			.channels_min = 2,
@@ -1916,6 +1898,7 @@ static struct snd_soc_dai_driver wm5100_dai[] = {
 	{
 		.name = "wm5100-aif2",
 		.id = 1,
+		.base = WM5100_AUDIO_IF_2_1 - 1,
 		.playback = {
 			.stream_name = "AIF2 Playback",
 			.channels_min = 2,
@@ -1935,6 +1918,7 @@ static struct snd_soc_dai_driver wm5100_dai[] = {
 	{
 		.name = "wm5100-aif3",
 		.id = 2,
+		.base = WM5100_AUDIO_IF_3_1 - 1,
 		.playback = {
 			.stream_name = "AIF3 Playback",
 			.channels_min = 2,
@@ -2454,7 +2438,7 @@ static __devinit int wm5100_i2c_probe(struct i2c_client *i2c,
 
 	wm5100->dev = &i2c->dev;
 
-	wm5100->regmap = regmap_init_i2c(i2c, &wm5100_regmap);
+	wm5100->regmap = devm_regmap_init_i2c(i2c, &wm5100_regmap);
 	if (IS_ERR(wm5100->regmap)) {
 		ret = PTR_ERR(wm5100->regmap);
 		dev_err(&i2c->dev, "Failed to allocate register map: %d\n",
@@ -2479,7 +2463,7 @@ static __devinit int wm5100_i2c_probe(struct i2c_client *i2c,
 	if (ret != 0) {
 		dev_err(&i2c->dev, "Failed to request core supplies: %d\n",
 			ret);
-		goto err_regmap;
+		goto err;
 	}
 
 	ret = regulator_bulk_enable(ARRAY_SIZE(wm5100->core_supplies),
@@ -2487,7 +2471,7 @@ static __devinit int wm5100_i2c_probe(struct i2c_client *i2c,
 	if (ret != 0) {
 		dev_err(&i2c->dev, "Failed to enable core supplies: %d\n",
 			ret);
-		goto err_regmap;
+		goto err;
 	}
 
 	if (wm5100->pdata.ldo_ena) {
@@ -2660,8 +2644,6 @@ err_ldo:
 err_enable:
 	regulator_bulk_disable(ARRAY_SIZE(wm5100->core_supplies),
 			       wm5100->core_supplies);
-err_regmap:
-	regmap_exit(wm5100->regmap);
 err:
 	return ret;
 }
@@ -2682,7 +2664,6 @@ static __devexit int wm5100_i2c_remove(struct i2c_client *i2c)
 		gpio_set_value_cansleep(wm5100->pdata.ldo_ena, 0);
 		gpio_free(wm5100->pdata.ldo_ena);
 	}
-	regmap_exit(wm5100->regmap);
 
 	return 0;
 }
@@ -2749,17 +2730,7 @@ static struct i2c_driver wm5100_i2c_driver = {
 	.id_table = wm5100_i2c_id,
 };
 
-static int __init wm5100_modinit(void)
-{
-	return i2c_add_driver(&wm5100_i2c_driver);
-}
-module_init(wm5100_modinit);
-
-static void __exit wm5100_exit(void)
-{
-	i2c_del_driver(&wm5100_i2c_driver);
-}
-module_exit(wm5100_exit);
+module_i2c_driver(wm5100_i2c_driver);
 
 MODULE_DESCRIPTION("ASoC WM5100 driver");
 MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>");
diff --git a/sound/soc/codecs/wm5100.h b/sound/soc/codecs/wm5100.h
index 25cb6016f9d7..935a9b7fb274 100644
--- a/sound/soc/codecs/wm5100.h
+++ b/sound/soc/codecs/wm5100.h
@@ -709,6 +709,96 @@ int wm5100_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack);
 #define WM5100_HPLPF3_2                         0xEC9
 #define WM5100_HPLPF4_1                         0xECC
 #define WM5100_HPLPF4_2                         0xECD
+#define WM5100_DSP1_CONTROL_1                   0xF00
+#define WM5100_DSP1_CONTROL_2                   0xF02
+#define WM5100_DSP1_CONTROL_3                   0xF03
+#define WM5100_DSP1_CONTROL_4                   0xF04
+#define WM5100_DSP1_CONTROL_5                   0xF06
+#define WM5100_DSP1_CONTROL_6                   0xF07
+#define WM5100_DSP1_CONTROL_7                   0xF08
+#define WM5100_DSP1_CONTROL_8                   0xF09
+#define WM5100_DSP1_CONTROL_9                   0xF0A
+#define WM5100_DSP1_CONTROL_10                  0xF0B
+#define WM5100_DSP1_CONTROL_11                  0xF0C
+#define WM5100_DSP1_CONTROL_12                  0xF0D
+#define WM5100_DSP1_CONTROL_13                  0xF0F
+#define WM5100_DSP1_CONTROL_14                  0xF10
+#define WM5100_DSP1_CONTROL_15                  0xF11
+#define WM5100_DSP1_CONTROL_16                  0xF12
+#define WM5100_DSP1_CONTROL_17                  0xF13
+#define WM5100_DSP1_CONTROL_18                  0xF14
+#define WM5100_DSP1_CONTROL_19                  0xF16
+#define WM5100_DSP1_CONTROL_20                  0xF17
+#define WM5100_DSP1_CONTROL_21                  0xF18
+#define WM5100_DSP1_CONTROL_22                  0xF1A
+#define WM5100_DSP1_CONTROL_23                  0xF1B
+#define WM5100_DSP1_CONTROL_24                  0xF1C
+#define WM5100_DSP1_CONTROL_25                  0xF1E
+#define WM5100_DSP1_CONTROL_26                  0xF20
+#define WM5100_DSP1_CONTROL_27                  0xF21
+#define WM5100_DSP1_CONTROL_28                  0xF22
+#define WM5100_DSP1_CONTROL_29                  0xF23
+#define WM5100_DSP1_CONTROL_30                  0xF24
+#define WM5100_DSP2_CONTROL_1                   0x1000
+#define WM5100_DSP2_CONTROL_2                   0x1002
+#define WM5100_DSP2_CONTROL_3                   0x1003
+#define WM5100_DSP2_CONTROL_4                   0x1004
+#define WM5100_DSP2_CONTROL_5                   0x1006
+#define WM5100_DSP2_CONTROL_6                   0x1007
+#define WM5100_DSP2_CONTROL_7                   0x1008
+#define WM5100_DSP2_CONTROL_8                   0x1009
+#define WM5100_DSP2_CONTROL_9                   0x100A
+#define WM5100_DSP2_CONTROL_10                  0x100B
+#define WM5100_DSP2_CONTROL_11                  0x100C
+#define WM5100_DSP2_CONTROL_12                  0x100D
+#define WM5100_DSP2_CONTROL_13                  0x100F
+#define WM5100_DSP2_CONTROL_14                  0x1010
+#define WM5100_DSP2_CONTROL_15                  0x1011
+#define WM5100_DSP2_CONTROL_16                  0x1012
+#define WM5100_DSP2_CONTROL_17                  0x1013
+#define WM5100_DSP2_CONTROL_18                  0x1014
+#define WM5100_DSP2_CONTROL_19                  0x1016
+#define WM5100_DSP2_CONTROL_20                  0x1017
+#define WM5100_DSP2_CONTROL_21                  0x1018
+#define WM5100_DSP2_CONTROL_22                  0x101A
+#define WM5100_DSP2_CONTROL_23                  0x101B
+#define WM5100_DSP2_CONTROL_24                  0x101C
+#define WM5100_DSP2_CONTROL_25                  0x101E
+#define WM5100_DSP2_CONTROL_26                  0x1020
+#define WM5100_DSP2_CONTROL_27                  0x1021
+#define WM5100_DSP2_CONTROL_28                  0x1022
+#define WM5100_DSP2_CONTROL_29                  0x1023
+#define WM5100_DSP2_CONTROL_30                  0x1024
+#define WM5100_DSP3_CONTROL_1                   0x1100
+#define WM5100_DSP3_CONTROL_2                   0x1102
+#define WM5100_DSP3_CONTROL_3                   0x1103
+#define WM5100_DSP3_CONTROL_4                   0x1104
+#define WM5100_DSP3_CONTROL_5                   0x1106
+#define WM5100_DSP3_CONTROL_6                   0x1107
+#define WM5100_DSP3_CONTROL_7                   0x1108
+#define WM5100_DSP3_CONTROL_8                   0x1109
+#define WM5100_DSP3_CONTROL_9                   0x110A
+#define WM5100_DSP3_CONTROL_10                  0x110B
+#define WM5100_DSP3_CONTROL_11                  0x110C
+#define WM5100_DSP3_CONTROL_12                  0x110D
+#define WM5100_DSP3_CONTROL_13                  0x110F
+#define WM5100_DSP3_CONTROL_14                  0x1110
+#define WM5100_DSP3_CONTROL_15                  0x1111
+#define WM5100_DSP3_CONTROL_16                  0x1112
+#define WM5100_DSP3_CONTROL_17                  0x1113
+#define WM5100_DSP3_CONTROL_18                  0x1114
+#define WM5100_DSP3_CONTROL_19                  0x1116
+#define WM5100_DSP3_CONTROL_20                  0x1117
+#define WM5100_DSP3_CONTROL_21                  0x1118
+#define WM5100_DSP3_CONTROL_22                  0x111A
+#define WM5100_DSP3_CONTROL_23                  0x111B
+#define WM5100_DSP3_CONTROL_24                  0x111C
+#define WM5100_DSP3_CONTROL_25                  0x111E
+#define WM5100_DSP3_CONTROL_26                  0x1120
+#define WM5100_DSP3_CONTROL_27                  0x1121
+#define WM5100_DSP3_CONTROL_28                  0x1122
+#define WM5100_DSP3_CONTROL_29                  0x1123
+#define WM5100_DSP3_CONTROL_30                  0x1124
 #define WM5100_DSP1_DM_0                        0x4000
 #define WM5100_DSP1_DM_1                        0x4001
 #define WM5100_DSP1_DM_2                        0x4002
@@ -4560,6 +4650,75 @@ int wm5100_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack);
 #define WM5100_LHPF4_COEFF_SHIFT                     0  /* LHPF4_COEFF - [15:0] */
 #define WM5100_LHPF4_COEFF_WIDTH                    16  /* LHPF4_COEFF - [15:0] */
 
+/*
+ * R4132 (0x1024) - DSP2 Control 30
+ */
+#define WM5100_DSP2_RATE_MASK                   0xC000  /* DSP2_RATE - [15:14] */
+#define WM5100_DSP2_RATE_SHIFT                      14  /* DSP2_RATE - [15:14] */
+#define WM5100_DSP2_RATE_WIDTH                       2  /* DSP2_RATE - [15:14] */
+#define WM5100_DSP2_DBG_CLK_ENA                 0x0008  /* DSP2_DBG_CLK_ENA */
+#define WM5100_DSP2_DBG_CLK_ENA_MASK            0x0008  /* DSP2_DBG_CLK_ENA */
+#define WM5100_DSP2_DBG_CLK_ENA_SHIFT                3  /* DSP2_DBG_CLK_ENA */
+#define WM5100_DSP2_DBG_CLK_ENA_WIDTH                1  /* DSP2_DBG_CLK_ENA */
+#define WM5100_DSP2_SYS_ENA                     0x0004  /* DSP2_SYS_ENA */
+#define WM5100_DSP2_SYS_ENA_MASK                0x0004  /* DSP2_SYS_ENA */
+#define WM5100_DSP2_SYS_ENA_SHIFT                    2  /* DSP2_SYS_ENA */
+#define WM5100_DSP2_SYS_ENA_WIDTH                    1  /* DSP2_SYS_ENA */
+#define WM5100_DSP2_CORE_ENA                    0x0002  /* DSP2_CORE_ENA */
+#define WM5100_DSP2_CORE_ENA_MASK               0x0002  /* DSP2_CORE_ENA */
+#define WM5100_DSP2_CORE_ENA_SHIFT                   1  /* DSP2_CORE_ENA */
+#define WM5100_DSP2_CORE_ENA_WIDTH                   1  /* DSP2_CORE_ENA */
+#define WM5100_DSP2_START                       0x0001  /* DSP2_START */
+#define WM5100_DSP2_START_MASK                  0x0001  /* DSP2_START */
+#define WM5100_DSP2_START_SHIFT                      0  /* DSP2_START */
+#define WM5100_DSP2_START_WIDTH                      1  /* DSP2_START */
+
+/*
+ * R3876 (0xF24) - DSP1 Control 30
+ */
+#define WM5100_DSP1_RATE_MASK                   0xC000  /* DSP1_RATE - [15:14] */
+#define WM5100_DSP1_RATE_SHIFT                      14  /* DSP1_RATE - [15:14] */
+#define WM5100_DSP1_RATE_WIDTH                       2  /* DSP1_RATE - [15:14] */
+#define WM5100_DSP1_DBG_CLK_ENA                 0x0008  /* DSP1_DBG_CLK_ENA */
+#define WM5100_DSP1_DBG_CLK_ENA_MASK            0x0008  /* DSP1_DBG_CLK_ENA */
+#define WM5100_DSP1_DBG_CLK_ENA_SHIFT                3  /* DSP1_DBG_CLK_ENA */
+#define WM5100_DSP1_DBG_CLK_ENA_WIDTH                1  /* DSP1_DBG_CLK_ENA */
+#define WM5100_DSP1_SYS_ENA                     0x0004  /* DSP1_SYS_ENA */
+#define WM5100_DSP1_SYS_ENA_MASK                0x0004  /* DSP1_SYS_ENA */
+#define WM5100_DSP1_SYS_ENA_SHIFT                    2  /* DSP1_SYS_ENA */
+#define WM5100_DSP1_SYS_ENA_WIDTH                    1  /* DSP1_SYS_ENA */
+#define WM5100_DSP1_CORE_ENA                    0x0002  /* DSP1_CORE_ENA */
+#define WM5100_DSP1_CORE_ENA_MASK               0x0002  /* DSP1_CORE_ENA */
+#define WM5100_DSP1_CORE_ENA_SHIFT                   1  /* DSP1_CORE_ENA */
+#define WM5100_DSP1_CORE_ENA_WIDTH                   1  /* DSP1_CORE_ENA */
+#define WM5100_DSP1_START                       0x0001  /* DSP1_START */
+#define WM5100_DSP1_START_MASK                  0x0001  /* DSP1_START */
+#define WM5100_DSP1_START_SHIFT                      0  /* DSP1_START */
+#define WM5100_DSP1_START_WIDTH                      1  /* DSP1_START */
+
+/*
+ * R4388 (0x1124) - DSP3 Control 30
+ */
+#define WM5100_DSP3_RATE_MASK                   0xC000  /* DSP3_RATE - [15:14] */
+#define WM5100_DSP3_RATE_SHIFT                      14  /* DSP3_RATE - [15:14] */
+#define WM5100_DSP3_RATE_WIDTH                       2  /* DSP3_RATE - [15:14] */
+#define WM5100_DSP3_DBG_CLK_ENA                 0x0008  /* DSP3_DBG_CLK_ENA */
+#define WM5100_DSP3_DBG_CLK_ENA_MASK            0x0008  /* DSP3_DBG_CLK_ENA */
+#define WM5100_DSP3_DBG_CLK_ENA_SHIFT                3  /* DSP3_DBG_CLK_ENA */
+#define WM5100_DSP3_DBG_CLK_ENA_WIDTH                1  /* DSP3_DBG_CLK_ENA */
+#define WM5100_DSP3_SYS_ENA                     0x0004  /* DSP3_SYS_ENA */
+#define WM5100_DSP3_SYS_ENA_MASK                0x0004  /* DSP3_SYS_ENA */
+#define WM5100_DSP3_SYS_ENA_SHIFT                    2  /* DSP3_SYS_ENA */
+#define WM5100_DSP3_SYS_ENA_WIDTH                    1  /* DSP3_SYS_ENA */
+#define WM5100_DSP3_CORE_ENA                    0x0002  /* DSP3_CORE_ENA */
+#define WM5100_DSP3_CORE_ENA_MASK               0x0002  /* DSP3_CORE_ENA */
+#define WM5100_DSP3_CORE_ENA_SHIFT                   1  /* DSP3_CORE_ENA */
+#define WM5100_DSP3_CORE_ENA_WIDTH                   1  /* DSP3_CORE_ENA */
+#define WM5100_DSP3_START                       0x0001  /* DSP3_START */
+#define WM5100_DSP3_START_MASK                  0x0001  /* DSP3_START */
+#define WM5100_DSP3_START_SHIFT                      0  /* DSP3_START */
+#define WM5100_DSP3_START_WIDTH                      1  /* DSP3_START */
+
 /*
  * R16384 (0x4000) - DSP1 DM 0
  */
diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c
index aa12c6b6beeb..555ee146ae0d 100644
--- a/sound/soc/codecs/wm8350.c
+++ b/sound/soc/codecs/wm8350.c
@@ -71,13 +71,6 @@ struct wm8350_data {
 	int fll_freq_in;
 };
 
-static unsigned int wm8350_codec_cache_read(struct snd_soc_codec *codec,
-					    unsigned int reg)
-{
-	struct wm8350 *wm8350 = codec->control_data;
-	return wm8350->reg_cache[reg];
-}
-
 static unsigned int wm8350_codec_read(struct snd_soc_codec *codec,
 				      unsigned int reg)
 {
@@ -99,7 +92,7 @@ static inline int wm8350_out1_ramp_step(struct snd_soc_codec *codec)
 {
 	struct wm8350_data *wm8350_data = snd_soc_codec_get_drvdata(codec);
 	struct wm8350_output *out1 = &wm8350_data->out1;
-	struct wm8350 *wm8350 = codec->control_data;
+	struct wm8350 *wm8350 = wm8350_data->wm8350;
 	int left_complete = 0, right_complete = 0;
 	u16 reg, val;
 
@@ -165,7 +158,7 @@ static inline int wm8350_out2_ramp_step(struct snd_soc_codec *codec)
 {
 	struct wm8350_data *wm8350_data = snd_soc_codec_get_drvdata(codec);
 	struct wm8350_output *out2 = &wm8350_data->out2;
-	struct wm8350 *wm8350 = codec->control_data;
+	struct wm8350 *wm8350 = wm8350_data->wm8350;
 	int left_complete = 0, right_complete = 0;
 	u16 reg, val;
 
@@ -360,8 +353,8 @@ static int wm8350_put_volsw_2r_vu(struct snd_kcontrol *kcontrol,
 		return ret;
 
 	/* now hit the volume update bits (always bit 8) */
-	val = wm8350_codec_read(codec, reg);
-	wm8350_codec_write(codec, reg, val | WM8350_OUT1_VU);
+	val = snd_soc_read(codec, reg);
+	snd_soc_write(codec, reg, val | WM8350_OUT1_VU);
 	return 1;
 }
 
@@ -781,7 +774,8 @@ static int wm8350_set_dai_sysclk(struct snd_soc_dai *codec_dai,
 				 int clk_id, unsigned int freq, int dir)
 {
 	struct snd_soc_codec *codec = codec_dai->codec;
-	struct wm8350 *wm8350 = codec->control_data;
+	struct wm8350_data *wm8350_data = snd_soc_codec_get_drvdata(codec);
+	struct wm8350 *wm8350 = wm8350_data->wm8350;
 	u16 fll_4;
 
 	switch (clk_id) {
@@ -795,9 +789,9 @@ static int wm8350_set_dai_sysclk(struct snd_soc_dai *codec_dai,
 	case WM8350_MCLK_SEL_PLL_32K:
 		wm8350_set_bits(wm8350, WM8350_CLOCK_CONTROL_1,
 				WM8350_MCLK_SEL);
-		fll_4 = wm8350_codec_read(codec, WM8350_FLL_CONTROL_4) &
+		fll_4 = snd_soc_read(codec, WM8350_FLL_CONTROL_4) &
 		    ~WM8350_FLL_CLK_SRC_MASK;
-		wm8350_codec_write(codec, WM8350_FLL_CONTROL_4, fll_4 | clk_id);
+		snd_soc_write(codec, WM8350_FLL_CONTROL_4, fll_4 | clk_id);
 		break;
 	}
 
@@ -819,39 +813,39 @@ static int wm8350_set_clkdiv(struct snd_soc_dai *codec_dai, int div_id, int div)
 
 	switch (div_id) {
 	case WM8350_ADC_CLKDIV:
-		val = wm8350_codec_read(codec, WM8350_ADC_DIVIDER) &
+		val = snd_soc_read(codec, WM8350_ADC_DIVIDER) &
 		    ~WM8350_ADC_CLKDIV_MASK;
-		wm8350_codec_write(codec, WM8350_ADC_DIVIDER, val | div);
+		snd_soc_write(codec, WM8350_ADC_DIVIDER, val | div);
 		break;
 	case WM8350_DAC_CLKDIV:
-		val = wm8350_codec_read(codec, WM8350_DAC_CLOCK_CONTROL) &
+		val = snd_soc_read(codec, WM8350_DAC_CLOCK_CONTROL) &
 		    ~WM8350_DAC_CLKDIV_MASK;
-		wm8350_codec_write(codec, WM8350_DAC_CLOCK_CONTROL, val | div);
+		snd_soc_write(codec, WM8350_DAC_CLOCK_CONTROL, val | div);
 		break;
 	case WM8350_BCLK_CLKDIV:
-		val = wm8350_codec_read(codec, WM8350_CLOCK_CONTROL_1) &
+		val = snd_soc_read(codec, WM8350_CLOCK_CONTROL_1) &
 		    ~WM8350_BCLK_DIV_MASK;
-		wm8350_codec_write(codec, WM8350_CLOCK_CONTROL_1, val | div);
+		snd_soc_write(codec, WM8350_CLOCK_CONTROL_1, val | div);
 		break;
 	case WM8350_OPCLK_CLKDIV:
-		val = wm8350_codec_read(codec, WM8350_CLOCK_CONTROL_1) &
+		val = snd_soc_read(codec, WM8350_CLOCK_CONTROL_1) &
 		    ~WM8350_OPCLK_DIV_MASK;
-		wm8350_codec_write(codec, WM8350_CLOCK_CONTROL_1, val | div);
+		snd_soc_write(codec, WM8350_CLOCK_CONTROL_1, val | div);
 		break;
 	case WM8350_SYS_CLKDIV:
-		val = wm8350_codec_read(codec, WM8350_CLOCK_CONTROL_1) &
+		val = snd_soc_read(codec, WM8350_CLOCK_CONTROL_1) &
 		    ~WM8350_MCLK_DIV_MASK;
-		wm8350_codec_write(codec, WM8350_CLOCK_CONTROL_1, val | div);
+		snd_soc_write(codec, WM8350_CLOCK_CONTROL_1, val | div);
 		break;
 	case WM8350_DACLR_CLKDIV:
-		val = wm8350_codec_read(codec, WM8350_DAC_LR_RATE) &
+		val = snd_soc_read(codec, WM8350_DAC_LR_RATE) &
 		    ~WM8350_DACLRC_RATE_MASK;
-		wm8350_codec_write(codec, WM8350_DAC_LR_RATE, val | div);
+		snd_soc_write(codec, WM8350_DAC_LR_RATE, val | div);
 		break;
 	case WM8350_ADCLR_CLKDIV:
-		val = wm8350_codec_read(codec, WM8350_ADC_LR_RATE) &
+		val = snd_soc_read(codec, WM8350_ADC_LR_RATE) &
 		    ~WM8350_ADCLRC_RATE_MASK;
-		wm8350_codec_write(codec, WM8350_ADC_LR_RATE, val | div);
+		snd_soc_write(codec, WM8350_ADC_LR_RATE, val | div);
 		break;
 	default:
 		return -EINVAL;
@@ -863,13 +857,13 @@ static int wm8350_set_clkdiv(struct snd_soc_dai *codec_dai, int div_id, int div)
 static int wm8350_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
 {
 	struct snd_soc_codec *codec = codec_dai->codec;
-	u16 iface = wm8350_codec_read(codec, WM8350_AI_FORMATING) &
+	u16 iface = snd_soc_read(codec, WM8350_AI_FORMATING) &
 	    ~(WM8350_AIF_BCLK_INV | WM8350_AIF_LRCLK_INV | WM8350_AIF_FMT_MASK);
-	u16 master = wm8350_codec_read(codec, WM8350_AI_DAC_CONTROL) &
+	u16 master = snd_soc_read(codec, WM8350_AI_DAC_CONTROL) &
 	    ~WM8350_BCLK_MSTR;
-	u16 dac_lrc = wm8350_codec_read(codec, WM8350_DAC_LR_RATE) &
+	u16 dac_lrc = snd_soc_read(codec, WM8350_DAC_LR_RATE) &
 	    ~WM8350_DACLRC_ENA;
-	u16 adc_lrc = wm8350_codec_read(codec, WM8350_ADC_LR_RATE) &
+	u16 adc_lrc = snd_soc_read(codec, WM8350_ADC_LR_RATE) &
 	    ~WM8350_ADCLRC_ENA;
 
 	/* set master/slave audio interface */
@@ -922,42 +916,10 @@ static int wm8350_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
 		return -EINVAL;
 	}
 
-	wm8350_codec_write(codec, WM8350_AI_FORMATING, iface);
-	wm8350_codec_write(codec, WM8350_AI_DAC_CONTROL, master);
-	wm8350_codec_write(codec, WM8350_DAC_LR_RATE, dac_lrc);
-	wm8350_codec_write(codec, WM8350_ADC_LR_RATE, adc_lrc);
-	return 0;
-}
-
-static int wm8350_pcm_trigger(struct snd_pcm_substream *substream,
-			      int cmd, struct snd_soc_dai *codec_dai)
-{
-	struct snd_soc_codec *codec = codec_dai->codec;
-	int master = wm8350_codec_cache_read(codec, WM8350_AI_DAC_CONTROL) &
-	    WM8350_BCLK_MSTR;
-	int enabled = 0;
-
-	/* Check that the DACs or ADCs are enabled since they are
-	 * required for LRC in master mode. The DACs or ADCs need a
-	 * valid audio path i.e. pin -> ADC or DAC -> pin before
-	 * the LRC will be enabled in master mode. */
-	if (!master || cmd != SNDRV_PCM_TRIGGER_START)
-		return 0;
-
-	if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
-		enabled = wm8350_codec_cache_read(codec, WM8350_POWER_MGMT_4) &
-		    (WM8350_ADCR_ENA | WM8350_ADCL_ENA);
-	} else {
-		enabled = wm8350_codec_cache_read(codec, WM8350_POWER_MGMT_4) &
-		    (WM8350_DACR_ENA | WM8350_DACL_ENA);
-	}
-
-	if (!enabled) {
-		dev_err(codec->dev,
-		       "%s: invalid audio path - no clocks available\n",
-		       __func__);
-		return -EINVAL;
-	}
+	snd_soc_write(codec, WM8350_AI_FORMATING, iface);
+	snd_soc_write(codec, WM8350_AI_DAC_CONTROL, master);
+	snd_soc_write(codec, WM8350_DAC_LR_RATE, dac_lrc);
+	snd_soc_write(codec, WM8350_ADC_LR_RATE, adc_lrc);
 	return 0;
 }
 
@@ -966,8 +928,9 @@ static int wm8350_pcm_hw_params(struct snd_pcm_substream *substream,
 				struct snd_soc_dai *codec_dai)
 {
 	struct snd_soc_codec *codec = codec_dai->codec;
-	struct wm8350 *wm8350 = codec->control_data;
-	u16 iface = wm8350_codec_read(codec, WM8350_AI_FORMATING) &
+	struct wm8350_data *wm8350_data = snd_soc_codec_get_drvdata(codec);
+	struct wm8350 *wm8350 = wm8350_data->wm8350;
+	u16 iface = snd_soc_read(codec, WM8350_AI_FORMATING) &
 	    ~WM8350_AIF_WL_MASK;
 
 	/* bit size */
@@ -985,7 +948,7 @@ static int wm8350_pcm_hw_params(struct snd_pcm_substream *substream,
 		break;
 	}
 
-	wm8350_codec_write(codec, WM8350_AI_FORMATING, iface);
+	snd_soc_write(codec, WM8350_AI_FORMATING, iface);
 
 	/* The sloping stopband filter is recommended for use with
 	 * lower sample rates to improve performance.
@@ -1005,12 +968,15 @@ static int wm8350_pcm_hw_params(struct snd_pcm_substream *substream,
 static int wm8350_mute(struct snd_soc_dai *dai, int mute)
 {
 	struct snd_soc_codec *codec = dai->codec;
-	struct wm8350 *wm8350 = codec->control_data;
+	unsigned int val;
 
 	if (mute)
-		wm8350_set_bits(wm8350, WM8350_DAC_MUTE, WM8350_DAC_MUTE_ENA);
+		val = WM8350_DAC_MUTE_ENA;
 	else
-		wm8350_clear_bits(wm8350, WM8350_DAC_MUTE, WM8350_DAC_MUTE_ENA);
+		val = 0;
+
+	snd_soc_update_bits(codec, WM8350_DAC_MUTE, WM8350_DAC_MUTE_ENA, val);
+
 	return 0;
 }
 
@@ -1079,8 +1045,8 @@ static int wm8350_set_fll(struct snd_soc_dai *codec_dai,
 			  unsigned int freq_out)
 {
 	struct snd_soc_codec *codec = codec_dai->codec;
-	struct wm8350 *wm8350 = codec->control_data;
 	struct wm8350_data *priv = snd_soc_codec_get_drvdata(codec);
+	struct wm8350 *wm8350 = priv->wm8350;
 	struct _fll_div fll_div;
 	int ret = 0;
 	u16 fll_1, fll_4;
@@ -1104,17 +1070,17 @@ static int wm8350_set_fll(struct snd_soc_dai *codec_dai,
 		fll_div.ratio);
 
 	/* set up N.K & dividers */
-	fll_1 = wm8350_codec_read(codec, WM8350_FLL_CONTROL_1) &
+	fll_1 = snd_soc_read(codec, WM8350_FLL_CONTROL_1) &
 	    ~(WM8350_FLL_OUTDIV_MASK | WM8350_FLL_RSP_RATE_MASK | 0xc000);
-	wm8350_codec_write(codec, WM8350_FLL_CONTROL_1,
+	snd_soc_write(codec, WM8350_FLL_CONTROL_1,
 			   fll_1 | (fll_div.div << 8) | 0x50);
-	wm8350_codec_write(codec, WM8350_FLL_CONTROL_2,
+	snd_soc_write(codec, WM8350_FLL_CONTROL_2,
 			   (fll_div.ratio << 11) | (fll_div.
 						    n & WM8350_FLL_N_MASK));
-	wm8350_codec_write(codec, WM8350_FLL_CONTROL_3, fll_div.k);
-	fll_4 = wm8350_codec_read(codec, WM8350_FLL_CONTROL_4) &
+	snd_soc_write(codec, WM8350_FLL_CONTROL_3, fll_div.k);
+	fll_4 = snd_soc_read(codec, WM8350_FLL_CONTROL_4) &
 	    ~(WM8350_FLL_FRAC | WM8350_FLL_SLOW_LOCK_REF);
-	wm8350_codec_write(codec, WM8350_FLL_CONTROL_4,
+	snd_soc_write(codec, WM8350_FLL_CONTROL_4,
 			   fll_4 | (fll_div.k ? WM8350_FLL_FRAC : 0) |
 			   (fll_div.ratio == 8 ? WM8350_FLL_SLOW_LOCK_REF : 0));
 
@@ -1131,8 +1097,8 @@ static int wm8350_set_fll(struct snd_soc_dai *codec_dai,
 static int wm8350_set_bias_level(struct snd_soc_codec *codec,
 				 enum snd_soc_bias_level level)
 {
-	struct wm8350 *wm8350 = codec->control_data;
 	struct wm8350_data *priv = snd_soc_codec_get_drvdata(codec);
+	struct wm8350 *wm8350 = priv->wm8350;
 	struct wm8350_audio_platform_data *platform =
 		wm8350->codec.platform_data;
 	u16 pm1;
@@ -1339,35 +1305,36 @@ static void wm8350_hpr_work(struct work_struct *work)
 	wm8350_hp_work(priv, &priv->hpr, WM8350_JACK_R_LVL);
 }
 
-static irqreturn_t wm8350_hp_jack_handler(int irq, void *data)
+static irqreturn_t wm8350_hpl_jack_handler(int irq, void *data)
 {
 	struct wm8350_data *priv = data;
 	struct wm8350 *wm8350 = priv->wm8350;
-	struct wm8350_jack_data *jack = NULL;
 
-	switch (irq - wm8350->irq_base) {
-	case WM8350_IRQ_CODEC_JCK_DET_L:
 #ifndef CONFIG_SND_SOC_WM8350_MODULE
-		trace_snd_soc_jack_irq("WM8350 HPL");
+	trace_snd_soc_jack_irq("WM8350 HPL");
 #endif
-		jack = &priv->hpl;
-		break;
-
-	case WM8350_IRQ_CODEC_JCK_DET_R:
-#ifndef CONFIG_SND_SOC_WM8350_MODULE
-		trace_snd_soc_jack_irq("WM8350 HPR");
-#endif
-		jack = &priv->hpr;
-		break;
-
-	default:
-		BUG();
-	}
 
 	if (device_may_wakeup(wm8350->dev))
 		pm_wakeup_event(wm8350->dev, 250);
 
-	schedule_delayed_work(&jack->work, 200);
+	schedule_delayed_work(&priv->hpl.work, 200);
+
+	return IRQ_HANDLED;
+}
+
+static irqreturn_t wm8350_hpr_jack_handler(int irq, void *data)
+{
+	struct wm8350_data *priv = data;
+	struct wm8350 *wm8350 = priv->wm8350;
+
+#ifndef CONFIG_SND_SOC_WM8350_MODULE
+	trace_snd_soc_jack_irq("WM8350 HPR");
+#endif
+
+	if (device_may_wakeup(wm8350->dev))
+		pm_wakeup_event(wm8350->dev, 250);
+
+	schedule_delayed_work(&priv->hpr.work, 200);
 
 	return IRQ_HANDLED;
 }
@@ -1387,7 +1354,7 @@ int wm8350_hp_jack_detect(struct snd_soc_codec *codec, enum wm8350_jack which,
 			  struct snd_soc_jack *jack, int report)
 {
 	struct wm8350_data *priv = snd_soc_codec_get_drvdata(codec);
-	struct wm8350 *wm8350 = codec->control_data;
+	struct wm8350 *wm8350 = priv->wm8350;
 	int irq;
 	int ena;
 
@@ -1418,7 +1385,14 @@ int wm8350_hp_jack_detect(struct snd_soc_codec *codec, enum wm8350_jack which,
 	}
 
 	/* Sync status */
-	wm8350_hp_jack_handler(irq + wm8350->irq_base, priv);
+	switch (which) {
+	case WM8350_JDL:
+		wm8350_hpl_jack_handler(0, priv);
+		break;
+	case WM8350_JDR:
+		wm8350_hpr_jack_handler(0, priv);
+		break;
+	}
 
 	return 0;
 }
@@ -1463,7 +1437,7 @@ int wm8350_mic_jack_detect(struct snd_soc_codec *codec,
 			   int detect_report, int short_report)
 {
 	struct wm8350_data *priv = snd_soc_codec_get_drvdata(codec);
-	struct wm8350 *wm8350 = codec->control_data;
+	struct wm8350 *wm8350 = priv->wm8350;
 
 	priv->mic.jack = jack;
 	priv->mic.report = detect_report;
@@ -1491,7 +1465,6 @@ EXPORT_SYMBOL_GPL(wm8350_mic_jack_detect);
 static const struct snd_soc_dai_ops wm8350_dai_ops = {
 	 .hw_params	= wm8350_pcm_hw_params,
 	 .digital_mute	= wm8350_mute,
-	 .trigger	= wm8350_pcm_trigger,
 	 .set_fmt	= wm8350_set_dai_fmt,
 	 .set_sysclk	= wm8350_set_dai_sysclk,
 	 .set_pll	= wm8350_set_fll,
@@ -1559,9 +1532,9 @@ static  int wm8350_codec_probe(struct snd_soc_codec *codec)
 	wm8350_set_bits(wm8350, WM8350_POWER_MGMT_5, WM8350_CODEC_ENA);
 
 	/* Enable robust clocking mode in ADC */
-	wm8350_codec_write(codec, WM8350_SECURITY, 0xa7);
-	wm8350_codec_write(codec, 0xde, 0x13);
-	wm8350_codec_write(codec, WM8350_SECURITY, 0);
+	snd_soc_write(codec, WM8350_SECURITY, 0xa7);
+	snd_soc_write(codec, 0xde, 0x13);
+	snd_soc_write(codec, WM8350_SECURITY, 0);
 
 	/* read OUT1 & OUT2 volumes */
 	out1 = &priv->out1;
@@ -1601,10 +1574,10 @@ static  int wm8350_codec_probe(struct snd_soc_codec *codec)
 			  WM8350_JDL_ENA | WM8350_JDR_ENA);
 
 	wm8350_register_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_L,
-			    wm8350_hp_jack_handler, 0, "Left jack detect",
+			    wm8350_hpl_jack_handler, 0, "Left jack detect",
 			    priv);
 	wm8350_register_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_R,
-			    wm8350_hp_jack_handler, 0, "Right jack detect",
+			    wm8350_hpr_jack_handler, 0, "Right jack detect",
 			    priv);
 	wm8350_register_irq(wm8350, WM8350_IRQ_CODEC_MICSCD,
 			    wm8350_mic_handler, 0, "Microphone short", priv);
diff --git a/sound/soc/codecs/wm8400.c b/sound/soc/codecs/wm8400.c
index 898979d23010..5dc31ebcd0e7 100644
--- a/sound/soc/codecs/wm8400.c
+++ b/sound/soc/codecs/wm8400.c
@@ -138,8 +138,8 @@ static int wm8400_outpga_put_volsw_vu(struct snd_kcontrol *kcontrol,
                 return ret;
 
         /* now hit the volume update bits (always bit 8) */
-        val = wm8400_read(codec, reg);
-        return wm8400_write(codec, reg, val | 0x0100);
+        val = snd_soc_read(codec, reg);
+        return snd_soc_write(codec, reg, val | 0x0100);
 }
 
 #define WM8400_OUTPGA_SINGLE_R_TLV(xname, reg, shift, max, invert, tlv_array) \
@@ -362,8 +362,8 @@ static int inmixer_event (struct snd_soc_dapm_widget *w,
 {
 	u16 reg, fakepower;
 
-	reg = wm8400_read(w->codec, WM8400_POWER_MANAGEMENT_2);
-	fakepower = wm8400_read(w->codec, WM8400_INTDRIVBITS);
+	reg = snd_soc_read(w->codec, WM8400_POWER_MANAGEMENT_2);
+	fakepower = snd_soc_read(w->codec, WM8400_INTDRIVBITS);
 
 	if (fakepower & ((1 << WM8400_INMIXL_PWR) |
 		(1 << WM8400_AINLMUX_PWR))) {
@@ -378,7 +378,7 @@ static int inmixer_event (struct snd_soc_dapm_widget *w,
 	} else {
 		reg &= ~WM8400_AINR_ENA;
 	}
-	wm8400_write(w->codec, WM8400_POWER_MANAGEMENT_2, reg);
+	snd_soc_write(w->codec, WM8400_POWER_MANAGEMENT_2, reg);
 
 	return 0;
 }
@@ -394,7 +394,7 @@ static int outmixer_event (struct snd_soc_dapm_widget *w,
 
 	switch (reg_shift) {
 	case WM8400_SPEAKER_MIXER | (WM8400_LDSPK << 8) :
-		reg = wm8400_read(w->codec, WM8400_OUTPUT_MIXER1);
+		reg = snd_soc_read(w->codec, WM8400_OUTPUT_MIXER1);
 		if (reg & WM8400_LDLO) {
 			printk(KERN_WARNING
 			"Cannot set as Output Mixer 1 LDLO Set\n");
@@ -402,7 +402,7 @@ static int outmixer_event (struct snd_soc_dapm_widget *w,
 		}
 		break;
 	case WM8400_SPEAKER_MIXER | (WM8400_RDSPK << 8):
-		reg = wm8400_read(w->codec, WM8400_OUTPUT_MIXER2);
+		reg = snd_soc_read(w->codec, WM8400_OUTPUT_MIXER2);
 		if (reg & WM8400_RDRO) {
 			printk(KERN_WARNING
 			"Cannot set as Output Mixer 2 RDRO Set\n");
@@ -410,7 +410,7 @@ static int outmixer_event (struct snd_soc_dapm_widget *w,
 		}
 		break;
 	case WM8400_OUTPUT_MIXER1 | (WM8400_LDLO << 8):
-		reg = wm8400_read(w->codec, WM8400_SPEAKER_MIXER);
+		reg = snd_soc_read(w->codec, WM8400_SPEAKER_MIXER);
 		if (reg & WM8400_LDSPK) {
 			printk(KERN_WARNING
 			"Cannot set as Speaker Mixer LDSPK Set\n");
@@ -418,7 +418,7 @@ static int outmixer_event (struct snd_soc_dapm_widget *w,
 		}
 		break;
 	case WM8400_OUTPUT_MIXER2 | (WM8400_RDRO << 8):
-		reg = wm8400_read(w->codec, WM8400_SPEAKER_MIXER);
+		reg = snd_soc_read(w->codec, WM8400_SPEAKER_MIXER);
 		if (reg & WM8400_RDSPK) {
 			printk(KERN_WARNING
 			"Cannot set as Speaker Mixer RDSPK Set\n");
@@ -1021,13 +1021,13 @@ static int wm8400_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id,
 	wm8400->fll_in = freq_in;
 
 	/* We *must* disable the FLL before any changes */
-	reg = wm8400_read(codec, WM8400_POWER_MANAGEMENT_2);
+	reg = snd_soc_read(codec, WM8400_POWER_MANAGEMENT_2);
 	reg &= ~WM8400_FLL_ENA;
-	wm8400_write(codec, WM8400_POWER_MANAGEMENT_2, reg);
+	snd_soc_write(codec, WM8400_POWER_MANAGEMENT_2, reg);
 
-	reg = wm8400_read(codec, WM8400_FLL_CONTROL_1);
+	reg = snd_soc_read(codec, WM8400_FLL_CONTROL_1);
 	reg &= ~WM8400_FLL_OSC_ENA;
-	wm8400_write(codec, WM8400_FLL_CONTROL_1, reg);
+	snd_soc_write(codec, WM8400_FLL_CONTROL_1, reg);
 
 	if (!freq_out)
 		return 0;
@@ -1035,15 +1035,15 @@ static int wm8400_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id,
 	reg &= ~(WM8400_FLL_REF_FREQ | WM8400_FLL_FRATIO_MASK);
 	reg |= WM8400_FLL_FRAC | factors.fratio;
 	reg |= factors.freq_ref << WM8400_FLL_REF_FREQ_SHIFT;
-	wm8400_write(codec, WM8400_FLL_CONTROL_1, reg);
+	snd_soc_write(codec, WM8400_FLL_CONTROL_1, reg);
 
-	wm8400_write(codec, WM8400_FLL_CONTROL_2, factors.k);
-	wm8400_write(codec, WM8400_FLL_CONTROL_3, factors.n);
+	snd_soc_write(codec, WM8400_FLL_CONTROL_2, factors.k);
+	snd_soc_write(codec, WM8400_FLL_CONTROL_3, factors.n);
 
-	reg = wm8400_read(codec, WM8400_FLL_CONTROL_4);
+	reg = snd_soc_read(codec, WM8400_FLL_CONTROL_4);
 	reg &= ~WM8400_FLL_OUTDIV_MASK;
 	reg |= factors.outdiv;
-	wm8400_write(codec, WM8400_FLL_CONTROL_4, reg);
+	snd_soc_write(codec, WM8400_FLL_CONTROL_4, reg);
 
 	return 0;
 }
@@ -1057,8 +1057,8 @@ static int wm8400_set_dai_fmt(struct snd_soc_dai *codec_dai,
 	struct snd_soc_codec *codec = codec_dai->codec;
 	u16 audio1, audio3;
 
-	audio1 = wm8400_read(codec, WM8400_AUDIO_INTERFACE_1);
-	audio3 = wm8400_read(codec, WM8400_AUDIO_INTERFACE_3);
+	audio1 = snd_soc_read(codec, WM8400_AUDIO_INTERFACE_1);
+	audio3 = snd_soc_read(codec, WM8400_AUDIO_INTERFACE_3);
 
 	/* set master/slave audio interface */
 	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
@@ -1099,8 +1099,8 @@ static int wm8400_set_dai_fmt(struct snd_soc_dai *codec_dai,
 		return -EINVAL;
 	}
 
-	wm8400_write(codec, WM8400_AUDIO_INTERFACE_1, audio1);
-	wm8400_write(codec, WM8400_AUDIO_INTERFACE_3, audio3);
+	snd_soc_write(codec, WM8400_AUDIO_INTERFACE_1, audio1);
+	snd_soc_write(codec, WM8400_AUDIO_INTERFACE_3, audio3);
 	return 0;
 }
 
@@ -1112,24 +1112,24 @@ static int wm8400_set_dai_clkdiv(struct snd_soc_dai *codec_dai,
 
 	switch (div_id) {
 	case WM8400_MCLK_DIV:
-		reg = wm8400_read(codec, WM8400_CLOCKING_2) &
+		reg = snd_soc_read(codec, WM8400_CLOCKING_2) &
 			~WM8400_MCLK_DIV_MASK;
-		wm8400_write(codec, WM8400_CLOCKING_2, reg | div);
+		snd_soc_write(codec, WM8400_CLOCKING_2, reg | div);
 		break;
 	case WM8400_DACCLK_DIV:
-		reg = wm8400_read(codec, WM8400_CLOCKING_2) &
+		reg = snd_soc_read(codec, WM8400_CLOCKING_2) &
 			~WM8400_DAC_CLKDIV_MASK;
-		wm8400_write(codec, WM8400_CLOCKING_2, reg | div);
+		snd_soc_write(codec, WM8400_CLOCKING_2, reg | div);
 		break;
 	case WM8400_ADCCLK_DIV:
-		reg = wm8400_read(codec, WM8400_CLOCKING_2) &
+		reg = snd_soc_read(codec, WM8400_CLOCKING_2) &
 			~WM8400_ADC_CLKDIV_MASK;
-		wm8400_write(codec, WM8400_CLOCKING_2, reg | div);
+		snd_soc_write(codec, WM8400_CLOCKING_2, reg | div);
 		break;
 	case WM8400_BCLK_DIV:
-		reg = wm8400_read(codec, WM8400_CLOCKING_1) &
+		reg = snd_soc_read(codec, WM8400_CLOCKING_1) &
 			~WM8400_BCLK_DIV_MASK;
-		wm8400_write(codec, WM8400_CLOCKING_1, reg | div);
+		snd_soc_write(codec, WM8400_CLOCKING_1, reg | div);
 		break;
 	default:
 		return -EINVAL;
@@ -1145,9 +1145,8 @@ static int wm8400_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;
-	u16 audio1 = wm8400_read(codec, WM8400_AUDIO_INTERFACE_1);
+	struct snd_soc_codec *codec = dai->codec;
+	u16 audio1 = snd_soc_read(codec, WM8400_AUDIO_INTERFACE_1);
 
 	audio1 &= ~WM8400_AIF_WL_MASK;
 	/* bit size */
@@ -1165,19 +1164,19 @@ static int wm8400_hw_params(struct snd_pcm_substream *substream,
 		break;
 	}
 
-	wm8400_write(codec, WM8400_AUDIO_INTERFACE_1, audio1);
+	snd_soc_write(codec, WM8400_AUDIO_INTERFACE_1, audio1);
 	return 0;
 }
 
 static int wm8400_mute(struct snd_soc_dai *dai, int mute)
 {
 	struct snd_soc_codec *codec = dai->codec;
-	u16 val = wm8400_read(codec, WM8400_DAC_CTRL) & ~WM8400_DAC_MUTE;
+	u16 val = snd_soc_read(codec, WM8400_DAC_CTRL) & ~WM8400_DAC_MUTE;
 
 	if (mute)
-		wm8400_write(codec, WM8400_DAC_CTRL, val | WM8400_DAC_MUTE);
+		snd_soc_write(codec, WM8400_DAC_CTRL, val | WM8400_DAC_MUTE);
 	else
-		wm8400_write(codec, WM8400_DAC_CTRL, val);
+		snd_soc_write(codec, WM8400_DAC_CTRL, val);
 
 	return 0;
 }
@@ -1196,9 +1195,9 @@ static int wm8400_set_bias_level(struct snd_soc_codec *codec,
 
 	case SND_SOC_BIAS_PREPARE:
 		/* VMID=2*50k */
-		val = wm8400_read(codec, WM8400_POWER_MANAGEMENT_1) &
+		val = snd_soc_read(codec, WM8400_POWER_MANAGEMENT_1) &
 			~WM8400_VMID_MODE_MASK;
-		wm8400_write(codec, WM8400_POWER_MANAGEMENT_1, val | 0x2);
+		snd_soc_write(codec, WM8400_POWER_MANAGEMENT_1, val | 0x2);
 		break;
 
 	case SND_SOC_BIAS_STANDBY:
@@ -1212,74 +1211,74 @@ static int wm8400_set_bias_level(struct snd_soc_codec *codec,
 				return ret;
 			}
 
-			wm8400_write(codec, WM8400_POWER_MANAGEMENT_1,
+			snd_soc_write(codec, WM8400_POWER_MANAGEMENT_1,
 				     WM8400_CODEC_ENA | WM8400_SYSCLK_ENA);
 
 			/* Enable POBCTRL, SOFT_ST, VMIDTOG and BUFDCOPEN */
-			wm8400_write(codec, WM8400_ANTIPOP2, WM8400_SOFTST |
+			snd_soc_write(codec, WM8400_ANTIPOP2, WM8400_SOFTST |
 				     WM8400_BUFDCOPEN | WM8400_POBCTRL);
 
 			msleep(50);
 
 			/* Enable VREF & VMID at 2x50k */
-			val = wm8400_read(codec, WM8400_POWER_MANAGEMENT_1);
+			val = snd_soc_read(codec, WM8400_POWER_MANAGEMENT_1);
 			val |= 0x2 | WM8400_VREF_ENA;
-			wm8400_write(codec, WM8400_POWER_MANAGEMENT_1, val);
+			snd_soc_write(codec, WM8400_POWER_MANAGEMENT_1, val);
 
 			/* Enable BUFIOEN */
-			wm8400_write(codec, WM8400_ANTIPOP2, WM8400_SOFTST |
+			snd_soc_write(codec, WM8400_ANTIPOP2, WM8400_SOFTST |
 				     WM8400_BUFDCOPEN | WM8400_POBCTRL |
 				     WM8400_BUFIOEN);
 
 			/* disable POBCTRL, SOFT_ST and BUFDCOPEN */
-			wm8400_write(codec, WM8400_ANTIPOP2, WM8400_BUFIOEN);
+			snd_soc_write(codec, WM8400_ANTIPOP2, WM8400_BUFIOEN);
 		}
 
 		/* VMID=2*300k */
-		val = wm8400_read(codec, WM8400_POWER_MANAGEMENT_1) &
+		val = snd_soc_read(codec, WM8400_POWER_MANAGEMENT_1) &
 			~WM8400_VMID_MODE_MASK;
-		wm8400_write(codec, WM8400_POWER_MANAGEMENT_1, val | 0x4);
+		snd_soc_write(codec, WM8400_POWER_MANAGEMENT_1, val | 0x4);
 		break;
 
 	case SND_SOC_BIAS_OFF:
 		/* Enable POBCTRL and SOFT_ST */
-		wm8400_write(codec, WM8400_ANTIPOP2, WM8400_SOFTST |
+		snd_soc_write(codec, WM8400_ANTIPOP2, WM8400_SOFTST |
 			WM8400_POBCTRL | WM8400_BUFIOEN);
 
 		/* Enable POBCTRL, SOFT_ST and BUFDCOPEN */
-		wm8400_write(codec, WM8400_ANTIPOP2, WM8400_SOFTST |
+		snd_soc_write(codec, WM8400_ANTIPOP2, WM8400_SOFTST |
 			WM8400_BUFDCOPEN | WM8400_POBCTRL |
 			WM8400_BUFIOEN);
 
 		/* mute DAC */
-		val = wm8400_read(codec, WM8400_DAC_CTRL);
-		wm8400_write(codec, WM8400_DAC_CTRL, val | WM8400_DAC_MUTE);
+		val = snd_soc_read(codec, WM8400_DAC_CTRL);
+		snd_soc_write(codec, WM8400_DAC_CTRL, val | WM8400_DAC_MUTE);
 
 		/* Enable any disabled outputs */
-		val = wm8400_read(codec, WM8400_POWER_MANAGEMENT_1);
+		val = snd_soc_read(codec, WM8400_POWER_MANAGEMENT_1);
 		val |= WM8400_SPK_ENA | WM8400_OUT3_ENA |
 			WM8400_OUT4_ENA | WM8400_LOUT_ENA |
 			WM8400_ROUT_ENA;
-		wm8400_write(codec, WM8400_POWER_MANAGEMENT_1, val);
+		snd_soc_write(codec, WM8400_POWER_MANAGEMENT_1, val);
 
 		/* Disable VMID */
 		val &= ~WM8400_VMID_MODE_MASK;
-		wm8400_write(codec, WM8400_POWER_MANAGEMENT_1, val);
+		snd_soc_write(codec, WM8400_POWER_MANAGEMENT_1, val);
 
 		msleep(300);
 
 		/* Enable all output discharge bits */
-		wm8400_write(codec, WM8400_ANTIPOP1, WM8400_DIS_LLINE |
+		snd_soc_write(codec, WM8400_ANTIPOP1, WM8400_DIS_LLINE |
 			WM8400_DIS_RLINE | WM8400_DIS_OUT3 |
 			WM8400_DIS_OUT4 | WM8400_DIS_LOUT |
 			WM8400_DIS_ROUT);
 
 		/* Disable VREF */
 		val &= ~WM8400_VREF_ENA;
-		wm8400_write(codec, WM8400_POWER_MANAGEMENT_1, val);
+		snd_soc_write(codec, WM8400_POWER_MANAGEMENT_1, val);
 
 		/* disable POBCTRL, SOFT_ST and BUFDCOPEN */
-		wm8400_write(codec, WM8400_ANTIPOP2, 0x0);
+		snd_soc_write(codec, WM8400_ANTIPOP2, 0x0);
 
 		ret = regulator_bulk_disable(ARRAY_SIZE(power),
 					     &power[0]);
@@ -1385,19 +1384,19 @@ static int wm8400_codec_probe(struct snd_soc_codec *codec)
 
 	wm8400_codec_reset(codec);
 
-	reg = wm8400_read(codec, WM8400_POWER_MANAGEMENT_1);
-	wm8400_write(codec, WM8400_POWER_MANAGEMENT_1, reg | WM8400_CODEC_ENA);
+	reg = snd_soc_read(codec, WM8400_POWER_MANAGEMENT_1);
+	snd_soc_write(codec, WM8400_POWER_MANAGEMENT_1, reg | WM8400_CODEC_ENA);
 
 	/* Latch volume update bits */
-	reg = wm8400_read(codec, WM8400_LEFT_LINE_INPUT_1_2_VOLUME);
-	wm8400_write(codec, WM8400_LEFT_LINE_INPUT_1_2_VOLUME,
+	reg = snd_soc_read(codec, WM8400_LEFT_LINE_INPUT_1_2_VOLUME);
+	snd_soc_write(codec, WM8400_LEFT_LINE_INPUT_1_2_VOLUME,
 		     reg & WM8400_IPVU);
-	reg = wm8400_read(codec, WM8400_RIGHT_LINE_INPUT_1_2_VOLUME);
-	wm8400_write(codec, WM8400_RIGHT_LINE_INPUT_1_2_VOLUME,
+	reg = snd_soc_read(codec, WM8400_RIGHT_LINE_INPUT_1_2_VOLUME);
+	snd_soc_write(codec, WM8400_RIGHT_LINE_INPUT_1_2_VOLUME,
 		     reg & WM8400_IPVU);
 
-	wm8400_write(codec, WM8400_LEFT_OUTPUT_VOLUME, 0x50 | (1<<8));
-	wm8400_write(codec, WM8400_RIGHT_OUTPUT_VOLUME, 0x50 | (1<<8));
+	snd_soc_write(codec, WM8400_LEFT_OUTPUT_VOLUME, 0x50 | (1<<8));
+	snd_soc_write(codec, WM8400_RIGHT_OUTPUT_VOLUME, 0x50 | (1<<8));
 
 	if (!schedule_work(&priv->work)) {
 		ret = -EINVAL;
@@ -1414,8 +1413,8 @@ static int  wm8400_codec_remove(struct snd_soc_codec *codec)
 {
 	u16 reg;
 
-	reg = wm8400_read(codec, WM8400_POWER_MANAGEMENT_1);
-	wm8400_write(codec, WM8400_POWER_MANAGEMENT_1,
+	reg = snd_soc_read(codec, WM8400_POWER_MANAGEMENT_1);
+	snd_soc_write(codec, WM8400_POWER_MANAGEMENT_1,
 		     reg & (~WM8400_CODEC_ENA));
 
 	regulator_bulk_free(ARRAY_SIZE(power), power);
@@ -1428,7 +1427,7 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8400 = {
 	.remove =	wm8400_codec_remove,
 	.suspend =	wm8400_suspend,
 	.resume =	wm8400_resume,
-	.read = wm8400_read,
+	.read = snd_soc_read,
 	.write = wm8400_write,
 	.set_bias_level = wm8400_set_bias_level,
 
diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c
index 9166126bd312..56a049555e2c 100644
--- a/sound/soc/codecs/wm8510.c
+++ b/sound/soc/codecs/wm8510.c
@@ -392,8 +392,7 @@ static int wm8510_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 snd_soc_codec *codec = dai->codec;
 	u16 iface = snd_soc_read(codec, WM8510_IFACE) & 0x19f;
 	u16 adn = snd_soc_read(codec, WM8510_ADD) & 0x1f1;
 
diff --git a/sound/soc/codecs/wm8523.c b/sound/soc/codecs/wm8523.c
index 7fea2c3bf7e7..1c3ffb290cdc 100644
--- a/sound/soc/codecs/wm8523.c
+++ b/sound/soc/codecs/wm8523.c
@@ -145,8 +145,7 @@ static int wm8523_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 snd_soc_codec *codec = dai->codec;
 	struct wm8523_priv *wm8523 = snd_soc_codec_get_drvdata(codec);
 	int i;
 	u16 aifctrl1 = snd_soc_read(codec, WM8523_AIF_CTRL1);
diff --git a/sound/soc/codecs/wm8728.c b/sound/soc/codecs/wm8728.c
index fc3d59e49084..1467f97dce21 100644
--- a/sound/soc/codecs/wm8728.c
+++ b/sound/soc/codecs/wm8728.c
@@ -88,8 +88,7 @@ static int wm8728_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 snd_soc_codec *codec = dai->codec;
 	u16 dac = snd_soc_read(codec, WM8728_DACCTL);
 
 	dac &= ~0x18;
diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c
index a32caa72bd7d..9d1b9b0271f1 100644
--- a/sound/soc/codecs/wm8731.c
+++ b/sound/soc/codecs/wm8731.c
@@ -635,16 +635,17 @@ static int __devinit wm8731_spi_probe(struct spi_device *spi)
 	struct wm8731_priv *wm8731;
 	int ret;
 
-	wm8731 = kzalloc(sizeof(struct wm8731_priv), GFP_KERNEL);
+	wm8731 = devm_kzalloc(&spi->dev, sizeof(struct wm8731_priv),
+			      GFP_KERNEL);
 	if (wm8731 == NULL)
 		return -ENOMEM;
 
-	wm8731->regmap = regmap_init_spi(spi, &wm8731_regmap);
+	wm8731->regmap = devm_regmap_init_spi(spi, &wm8731_regmap);
 	if (IS_ERR(wm8731->regmap)) {
 		ret = PTR_ERR(wm8731->regmap);
 		dev_err(&spi->dev, "Failed to allocate register map: %d\n",
 			ret);
-		goto err;
+		return ret;
 	}
 
 	spi_set_drvdata(spi, wm8731);
@@ -653,25 +654,15 @@ static int __devinit wm8731_spi_probe(struct spi_device *spi)
 			&soc_codec_dev_wm8731, &wm8731_dai, 1);
 	if (ret != 0) {
 		dev_err(&spi->dev, "Failed to register CODEC: %d\n", ret);
-		goto err_regmap;
+		return ret;
 	}
 
 	return 0;
-
-err_regmap:
-	regmap_exit(wm8731->regmap);
-err:
-	kfree(wm8731);
-	return ret;
 }
 
 static int __devexit wm8731_spi_remove(struct spi_device *spi)
 {
-	struct wm8731_priv *wm8731 = spi_get_drvdata(spi);
-
 	snd_soc_unregister_codec(&spi->dev);
-	regmap_exit(wm8731->regmap);
-	kfree(wm8731);
 	return 0;
 }
 
@@ -693,16 +684,17 @@ static __devinit int wm8731_i2c_probe(struct i2c_client *i2c,
 	struct wm8731_priv *wm8731;
 	int ret;
 
-	wm8731 = kzalloc(sizeof(struct wm8731_priv), GFP_KERNEL);
+	wm8731 = devm_kzalloc(&i2c->dev, sizeof(struct wm8731_priv),
+			      GFP_KERNEL);
 	if (wm8731 == NULL)
 		return -ENOMEM;
 
-	wm8731->regmap = regmap_init_i2c(i2c, &wm8731_regmap);
+	wm8731->regmap = devm_regmap_init_i2c(i2c, &wm8731_regmap);
 	if (IS_ERR(wm8731->regmap)) {
 		ret = PTR_ERR(wm8731->regmap);
 		dev_err(&i2c->dev, "Failed to allocate register map: %d\n",
 			ret);
-		goto err;
+		return ret;
 	}
 
 	i2c_set_clientdata(i2c, wm8731);
@@ -711,24 +703,15 @@ static __devinit int wm8731_i2c_probe(struct i2c_client *i2c,
 			&soc_codec_dev_wm8731, &wm8731_dai, 1);
 	if (ret != 0) {
 		dev_err(&i2c->dev, "Failed to register CODEC: %d\n", ret);
-		goto err_regmap;
+		return ret;
 	}
 
 	return 0;
-
-err_regmap:
-	regmap_exit(wm8731->regmap);
-err:
-	kfree(wm8731);
-	return ret;
 }
 
 static __devexit int wm8731_i2c_remove(struct i2c_client *client)
 {
-	struct wm8731_priv *wm8731 = i2c_get_clientdata(client);
 	snd_soc_unregister_codec(&client->dev);
-	regmap_exit(wm8731->regmap);
-	kfree(wm8731);
 	return 0;
 }
 
diff --git a/sound/soc/codecs/wm8737.c b/sound/soc/codecs/wm8737.c
index 4fe9d191e277..d0520124616d 100644
--- a/sound/soc/codecs/wm8737.c
+++ b/sound/soc/codecs/wm8737.c
@@ -329,8 +329,7 @@ static int wm8737_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 snd_soc_codec *codec = dai->codec;
 	struct wm8737_priv *wm8737 = snd_soc_codec_get_drvdata(codec);
 	int i;
 	u16 clocking = 0;
diff --git a/sound/soc/codecs/wm8741.c b/sound/soc/codecs/wm8741.c
index 3941f50bf187..6e849cb04243 100644
--- a/sound/soc/codecs/wm8741.c
+++ b/sound/soc/codecs/wm8741.c
@@ -203,8 +203,7 @@ static int wm8741_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 snd_soc_codec *codec = dai->codec;
 	struct wm8741_priv *wm8741 = snd_soc_codec_get_drvdata(codec);
 	u16 iface = snd_soc_read(codec, WM8741_FORMAT_CONTROL) & 0x1FC;
 	int i;
diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c
index e4c50ce7d9c0..89151ca5e776 100644
--- a/sound/soc/codecs/wm8750.c
+++ b/sound/soc/codecs/wm8750.c
@@ -547,8 +547,7 @@ static int wm8750_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 snd_soc_codec *codec = dai->codec;
 	struct wm8750_priv *wm8750 = snd_soc_codec_get_drvdata(codec);
 	u16 iface = snd_soc_read(codec, WM8750_IFACE) & 0x1f3;
 	u16 srate = snd_soc_read(codec, WM8750_SRATE) & 0x1c0;
diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c
index e27e7b62b365..a26482cd7654 100644
--- a/sound/soc/codecs/wm8753.c
+++ b/sound/soc/codecs/wm8753.c
@@ -931,8 +931,7 @@ static int wm8753_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 snd_soc_codec *codec = dai->codec;
 	struct wm8753_priv *wm8753 = snd_soc_codec_get_drvdata(codec);
 	u16 voice = snd_soc_read(codec, WM8753_PCM) & 0x01f3;
 	u16 srate = snd_soc_read(codec, WM8753_SRATE1) & 0x017f;
@@ -1161,8 +1160,7 @@ static int wm8753_i2s_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 snd_soc_codec *codec = dai->codec;
 	struct wm8753_priv *wm8753 = snd_soc_codec_get_drvdata(codec);
 	u16 srate = snd_soc_read(codec, WM8753_SRATE1) & 0x01c0;
 	u16 hifi = snd_soc_read(codec, WM8753_HIFI) & 0x01f3;
diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c
index f18c554efc98..077c9628c70d 100644
--- a/sound/soc/codecs/wm8900.c
+++ b/sound/soc/codecs/wm8900.c
@@ -610,8 +610,7 @@ static int wm8900_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 snd_soc_codec *codec = dai->codec;
 	u16 reg;
 
 	reg = snd_soc_read(codec, WM8900_REG_AUDIO1) & ~0x60;
diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c
index c91fb2f99c13..86b8a2926591 100644
--- a/sound/soc/codecs/wm8903.c
+++ b/sound/soc/codecs/wm8903.c
@@ -1432,8 +1432,7 @@ static int wm8903_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 snd_soc_codec *codec = dai->codec;
 	struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec);
 	int fs = params_rate(params);
 	int bclk;
diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c
index d2883affea3b..481a3d9cfe48 100644
--- a/sound/soc/codecs/wm8940.c
+++ b/sound/soc/codecs/wm8940.c
@@ -371,8 +371,7 @@ static int wm8940_i2s_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 snd_soc_codec *codec = dai->codec;
 	u16 iface = snd_soc_read(codec, WM8940_IFACE) & 0xFD9F;
 	u16 addcntrl = snd_soc_read(codec, WM8940_ADDCNTRL) & 0xFFF1;
 	u16 companding =  snd_soc_read(codec,
diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c
index 840d72086d04..8bc659d8dd2e 100644
--- a/sound/soc/codecs/wm8960.c
+++ b/sound/soc/codecs/wm8960.c
@@ -505,8 +505,7 @@ static int wm8960_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 snd_soc_codec *codec = dai->codec;
 	struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec);
 	u16 iface = snd_soc_read(codec, WM8960_IFACE1) & 0xfff3;
 	int i;
diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c
index 15d467ff91b4..0cfce9999c89 100644
--- a/sound/soc/codecs/wm8962.c
+++ b/sound/soc/codecs/wm8962.c
@@ -1478,7 +1478,8 @@ static const DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0);
 
 static int wm8962_dsp2_write_config(struct snd_soc_codec *codec)
 {
-	return 0;
+	return regcache_sync_region(codec->control_data,
+				    WM8962_HDBASS_AI_1, WM8962_MAX_REGISTER);
 }
 
 static int wm8962_dsp2_set_enable(struct snd_soc_codec *codec, u16 val)
@@ -1755,10 +1756,22 @@ SOC_DOUBLE_R_TLV("EQ4 Volume", WM8962_EQ3, WM8962_EQ23,
 SOC_DOUBLE_R_TLV("EQ5 Volume", WM8962_EQ3, WM8962_EQ23,
 		 WM8962_EQL_B5_GAIN_SHIFT, 31, 0, eq_tlv),
 
+SOC_SINGLE("3D Switch", WM8962_THREED1, 0, 1, 0),
+SND_SOC_BYTES_MASK("3D Coefficients", WM8962_THREED1, 4, WM8962_THREED_ENA),
+
+SOC_SINGLE("DF1 Switch", WM8962_DF1, 0, 1, 0),
+SND_SOC_BYTES_MASK("DF1 Coefficients", WM8962_DF1, 7, WM8962_DF1_ENA),
+
+SOC_SINGLE("DRC Switch", WM8962_DRC_1, 0, 1, 0),
+SND_SOC_BYTES_MASK("DRC Coefficients", WM8962_DRC_1, 5, WM8962_DRC_ENA),
+
 WM8962_DSP2_ENABLE("VSS Switch", WM8962_VSS_ENA_SHIFT),
+SND_SOC_BYTES("VSS Coefficients", WM8962_VSS_XHD2_1, 148),
 WM8962_DSP2_ENABLE("HPF1 Switch", WM8962_HPF1_ENA_SHIFT),
 WM8962_DSP2_ENABLE("HPF2 Switch", WM8962_HPF2_ENA_SHIFT),
+SND_SOC_BYTES("HPF Coefficients", WM8962_LHPF2, 1),
 WM8962_DSP2_ENABLE("HD Bass Switch", WM8962_HDBASS_ENA_SHIFT),
+SND_SOC_BYTES("HD Bass Coefficients", WM8962_HDBASS_AI_1, 30),
 };
 
 static const struct snd_kcontrol_new wm8962_spk_mono_controls[] = {
@@ -2519,8 +2532,7 @@ static int wm8962_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 snd_soc_codec *codec = dai->codec;
 	struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec);
 	int i;
 	int aif0 = 0;
diff --git a/sound/soc/codecs/wm8971.c b/sound/soc/codecs/wm8971.c
index 28fe59e3ce01..eef783f6b6d6 100644
--- a/sound/soc/codecs/wm8971.c
+++ b/sound/soc/codecs/wm8971.c
@@ -478,8 +478,7 @@ static int wm8971_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 snd_soc_codec *codec = dai->codec;
 	struct wm8971_priv *wm8971 = snd_soc_codec_get_drvdata(codec);
 	u16 iface = snd_soc_read(codec, WM8971_IFACE) & 0x1f3;
 	u16 srate = snd_soc_read(codec, WM8971_SRATE) & 0x1c0;
diff --git a/sound/soc/codecs/wm8978.c b/sound/soc/codecs/wm8978.c
index 72d5fdcd3cc2..a5be3adecf75 100644
--- a/sound/soc/codecs/wm8978.c
+++ b/sound/soc/codecs/wm8978.c
@@ -723,8 +723,7 @@ static int wm8978_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 snd_soc_codec *codec = dai->codec;
 	struct wm8978_priv *wm8978 = snd_soc_codec_get_drvdata(codec);
 	/* Word length mask = 0x60 */
 	u16 iface_ctl = snd_soc_read(codec, WM8978_AUDIO_INTERFACE) & ~0x60;
diff --git a/sound/soc/codecs/wm8988.c b/sound/soc/codecs/wm8988.c
index 6cdf6a2bc283..1d4c5cf47b06 100644
--- a/sound/soc/codecs/wm8988.c
+++ b/sound/soc/codecs/wm8988.c
@@ -668,8 +668,7 @@ static int wm8988_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 snd_soc_codec *codec = dai->codec;
 	struct wm8988_priv *wm8988 = snd_soc_codec_get_drvdata(codec);
 	u16 iface = snd_soc_read(codec, WM8988_IFACE) & 0x1f3;
 	u16 srate = snd_soc_read(codec, WM8988_SRATE) & 0x180;
diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c
index 9d242351e6e8..db63c97ddf51 100644
--- a/sound/soc/codecs/wm8990.c
+++ b/sound/soc/codecs/wm8990.c
@@ -1112,8 +1112,7 @@ static int wm8990_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 snd_soc_codec *codec = dai->codec;
 	u16 audio1 = snd_soc_read(codec, WM8990_AUDIO_INTERFACE_1);
 
 	audio1 &= ~WM8990_AIF_WL_MASK;
diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c
index d256a9340644..36acfccab999 100644
--- a/sound/soc/codecs/wm8993.c
+++ b/sound/soc/codecs/wm8993.c
@@ -218,7 +218,6 @@ struct wm8993_priv {
 	unsigned int sysclk_rate;
 	unsigned int fs;
 	unsigned int bclk;
-	int class_w_users;
 	unsigned int fll_fref;
 	unsigned int fll_fout;
 	int fll_src;
@@ -824,84 +823,6 @@ static int clk_sys_event(struct snd_soc_dapm_widget *w,
 	return 0;
 }
 
-/*
- * When used with DAC outputs only the WM8993 charge pump supports
- * operation in class W mode, providing very low power consumption
- * when used with digital sources.  Enable and disable this mode
- * automatically depending on the mixer configuration.
- *
- * Currently the only supported paths are the direct DAC->headphone
- * paths (which provide minimum power consumption anyway).
- */
-static int class_w_put(struct snd_kcontrol *kcontrol,
-		       struct snd_ctl_elem_value *ucontrol)
-{
-	struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol);
-	struct snd_soc_dapm_widget *widget = wlist->widgets[0];
-	struct snd_soc_codec *codec = widget->codec;
-	struct wm8993_priv *wm8993 = snd_soc_codec_get_drvdata(codec);
-	int ret;
-
-	/* Turn it off if we're using the main output mixer */
-	if (ucontrol->value.integer.value[0] == 0) {
-		if (wm8993->class_w_users == 0) {
-			dev_dbg(codec->dev, "Disabling Class W\n");
-			snd_soc_update_bits(codec, WM8993_CLASS_W_0,
-					    WM8993_CP_DYN_FREQ |
-					    WM8993_CP_DYN_V,
-					    0);
-		}
-		wm8993->class_w_users++;
-		wm8993->hubs_data.class_w = true;
-	}
-
-	/* Implement the change */
-	ret = snd_soc_dapm_put_enum_double(kcontrol, ucontrol);
-
-	/* Enable it if we're using the direct DAC path */
-	if (ucontrol->value.integer.value[0] == 1) {
-		if (wm8993->class_w_users == 1) {
-			dev_dbg(codec->dev, "Enabling Class W\n");
-			snd_soc_update_bits(codec, WM8993_CLASS_W_0,
-					    WM8993_CP_DYN_FREQ |
-					    WM8993_CP_DYN_V,
-					    WM8993_CP_DYN_FREQ |
-					    WM8993_CP_DYN_V);
-		}
-		wm8993->class_w_users--;
-		wm8993->hubs_data.class_w = false;
-	}
-
-	dev_dbg(codec->dev, "Indirect DAC use count now %d\n",
-		wm8993->class_w_users);
-
-	return ret;
-}
-
-#define SOC_DAPM_ENUM_W(xname, xenum) \
-{	.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
-	.info = snd_soc_info_enum_double, \
-	.get = snd_soc_dapm_get_enum_double, \
-	.put = class_w_put, \
-	.private_value = (unsigned long)&xenum }
-
-static const char *hp_mux_text[] = {
-	"Mixer",
-	"DAC",
-};
-
-static const struct soc_enum hpl_enum =
-	SOC_ENUM_SINGLE(WM8993_OUTPUT_MIXER1, 8, 2, hp_mux_text);
-
-static const struct snd_kcontrol_new hpl_mux =
-	SOC_DAPM_ENUM_W("Left Headphone Mux", hpl_enum);
-
-static const struct soc_enum hpr_enum =
-	SOC_ENUM_SINGLE(WM8993_OUTPUT_MIXER2, 8, 2, hp_mux_text);
-
-static const struct snd_kcontrol_new hpr_mux =
-	SOC_DAPM_ENUM_W("Right Headphone Mux", hpr_enum);
-
 static const struct snd_kcontrol_new left_speaker_mixer[] = {
 SOC_DAPM_SINGLE("Input Switch", WM8993_SPEAKER_MIXER, 7, 1, 0),
 SOC_DAPM_SINGLE("IN1LP Switch", WM8993_SPEAKER_MIXER, 5, 1, 0),
@@ -988,8 +909,8 @@ SND_SOC_DAPM_MUX("DACR Sidetone", SND_SOC_NOPM, 0, 0, &sidetoner_mux),
 SND_SOC_DAPM_DAC("DACL", NULL, WM8993_POWER_MANAGEMENT_3, 1, 0),
 SND_SOC_DAPM_DAC("DACR", NULL, WM8993_POWER_MANAGEMENT_3, 0, 0),
 
-SND_SOC_DAPM_MUX("Left Headphone Mux", SND_SOC_NOPM, 0, 0, &hpl_mux),
-SND_SOC_DAPM_MUX("Right Headphone Mux", SND_SOC_NOPM, 0, 0, &hpr_mux),
+SND_SOC_DAPM_MUX("Left Headphone Mux", SND_SOC_NOPM, 0, 0, &wm_hubs_hpl_mux),
+SND_SOC_DAPM_MUX("Right Headphone Mux", SND_SOC_NOPM, 0, 0, &wm_hubs_hpr_mux),
 
 SND_SOC_DAPM_MIXER("SPKL", WM8993_POWER_MANAGEMENT_3, 8, 0,
 		   left_speaker_mixer, ARRAY_SIZE(left_speaker_mixer)),
@@ -1579,9 +1500,6 @@ static int wm8993_probe(struct snd_soc_codec *codec)
 		return ret;
 	}
 
-	/* By default we're using the output mixers */
-	wm8993->class_w_users = 2;
-
 	/* Latch volume update bits and default ZC on */
 	snd_soc_update_bits(codec, WM8993_RIGHT_DAC_DIGITAL_VOLUME,
 			    WM8993_DAC_VU, WM8993_DAC_VU);
diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index 2de12ebe43b5..993639d694ce 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -70,8 +70,8 @@ static const struct wm8958_micd_rate micdet_rates[] = {
 static const struct wm8958_micd_rate jackdet_rates[] = {
 	{ 32768,       true,  0, 1 },
 	{ 32768,       false, 0, 1 },
-	{ 44100 * 256, true,  7, 10 },
-	{ 44100 * 256, false, 7, 10 },
+	{ 44100 * 256, true,  10, 10 },
+	{ 44100 * 256, false, 7, 8 },
 };
 
 static void wm8958_micd_set_rate(struct snd_soc_codec *codec)
@@ -82,7 +82,8 @@ static void wm8958_micd_set_rate(struct snd_soc_codec *codec)
 	const struct wm8958_micd_rate *rates;
 	int num_rates;
 
-	if (wm8994->jack_cb != wm8958_default_micdet)
+	if (!(wm8994->pdata && wm8994->pdata->micd_rates) &&
+	    wm8994->jack_cb != wm8958_default_micdet)
 		return;
 
 	idle = !wm8994->jack_mic;
@@ -118,6 +119,10 @@ static void wm8958_micd_set_rate(struct snd_soc_codec *codec)
 	val = rates[best].start << WM8958_MICD_BIAS_STARTTIME_SHIFT
 		| rates[best].rate << WM8958_MICD_RATE_SHIFT;
 
+	dev_dbg(codec->dev, "MICD rate %d,%d for %dHz %s\n",
+		rates[best].start, rates[best].rate, sysclk,
+		idle ? "idle" : "active");
+
 	snd_soc_update_bits(codec, WM8958_MIC_DETECT_1,
 			    WM8958_MICD_BIAS_STARTTIME_MASK |
 			    WM8958_MICD_RATE_MASK, val);
@@ -398,7 +403,7 @@ static void wm8994_set_retune_mobile(struct snd_soc_codec *codec, int block)
 		wm8994->dac_rates[iface]);
 
 	/* The EQ will be disabled while reconfiguring it, remember the
-	 * current configuration. 
+	 * current configuration.
 	 */
 	save = snd_soc_read(codec, base);
 	save &= WM8994_AIF1DAC1_EQ_ENA;
@@ -686,6 +691,9 @@ static void wm1811_jackdet_set_mode(struct snd_soc_codec *codec, u16 mode)
 {
 	struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
 
+	if (!wm8994->jackdet || !wm8994->jack_cb)
+		return;
+
 	if (!wm8994->jackdet || !wm8994->jack_cb)
 		return;
 
@@ -784,7 +792,7 @@ static void vmid_reference(struct snd_soc_codec *codec)
 
 		switch (wm8994->vmid_mode) {
 		default:
-			WARN_ON(0 == "Invalid VMID mode");
+			WARN_ON(NULL == "Invalid VMID mode");
 		case WM8994_VMID_NORMAL:
 			/* Startup bias, VMID ramp & buffer */
 			snd_soc_update_bits(codec, WM8994_ANTIPOP_2,
@@ -937,27 +945,12 @@ static int vmid_event(struct snd_soc_dapm_widget *w,
 	return 0;
 }
 
-static void wm8994_update_class_w(struct snd_soc_codec *codec)
+static bool wm8994_check_class_w_digital(struct snd_soc_codec *codec)
 {
-	struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
-	int enable = 1;
 	int source = 0;  /* GCC flow analysis can't track enable */
 	int reg, reg_r;
 
-	/* Only support direct DAC->headphone paths */
-	reg = snd_soc_read(codec, WM8994_OUTPUT_MIXER_1);
-	if (!(reg & WM8994_DAC1L_TO_HPOUT1L)) {
-		dev_vdbg(codec->dev, "HPL connected to output mixer\n");
-		enable = 0;
-	}
-
-	reg = snd_soc_read(codec, WM8994_OUTPUT_MIXER_2);
-	if (!(reg & WM8994_DAC1R_TO_HPOUT1R)) {
-		dev_vdbg(codec->dev, "HPR connected to output mixer\n");
-		enable = 0;
-	}
-
-	/* We also need the same setting for L/R and only one path */
+	/* We also need the same AIF source for L/R and only one path */
 	reg = snd_soc_read(codec, WM8994_DAC1_LEFT_MIXER_ROUTING);
 	switch (reg) {
 	case WM8994_AIF2DACL_TO_DAC1L:
@@ -974,30 +967,20 @@ static void wm8994_update_class_w(struct snd_soc_codec *codec)
 		break;
 	default:
 		dev_vdbg(codec->dev, "DAC mixer setting: %x\n", reg);
-		enable = 0;
-		break;
+		return false;
 	}
 
 	reg_r = snd_soc_read(codec, WM8994_DAC1_RIGHT_MIXER_ROUTING);
 	if (reg_r != reg) {
 		dev_vdbg(codec->dev, "Left and right DAC mixers different\n");
-		enable = 0;
+		return false;
 	}
 
-	if (enable) {
-		dev_dbg(codec->dev, "Class W enabled\n");
-		snd_soc_update_bits(codec, WM8994_CLASS_W_1,
-				    WM8994_CP_DYN_PWR |
-				    WM8994_CP_DYN_SRC_SEL_MASK,
-				    source | WM8994_CP_DYN_PWR);
-		wm8994->hubs.class_w = true;
-		
-	} else {
-		dev_dbg(codec->dev, "Class W disabled\n");
-		snd_soc_update_bits(codec, WM8994_CLASS_W_1,
-				    WM8994_CP_DYN_PWR, 0);
-		wm8994->hubs.class_w = false;
-	}
+	/* Set the source up */
+	snd_soc_update_bits(codec, WM8994_CLASS_W_1,
+			    WM8994_CP_DYN_SRC_SEL_MASK, source);
+
+	return true;
 }
 
 static int aif1clk_ev(struct snd_soc_dapm_widget *w,
@@ -1280,45 +1263,6 @@ static int dac_ev(struct snd_soc_dapm_widget *w,
 	return 0;
 }
 
-static const char *hp_mux_text[] = {
-	"Mixer",
-	"DAC",
-};
-
-#define WM8994_HP_ENUM(xname, xenum) \
-{	.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
-	.info = snd_soc_info_enum_double, \
- 	.get = snd_soc_dapm_get_enum_double, \
- 	.put = wm8994_put_hp_enum, \
-  	.private_value = (unsigned long)&xenum }
-
-static int wm8994_put_hp_enum(struct snd_kcontrol *kcontrol,
-			      struct snd_ctl_elem_value *ucontrol)
-{
-	struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol);
-	struct snd_soc_dapm_widget *w = wlist->widgets[0];
-	struct snd_soc_codec *codec = w->codec;
-	int ret;
-
-	ret = snd_soc_dapm_put_enum_double(kcontrol, ucontrol);
-
-	wm8994_update_class_w(codec);
-
-	return ret;
-}
-
-static const struct soc_enum hpl_enum =
-	SOC_ENUM_SINGLE(WM8994_OUTPUT_MIXER_1, 8, 2, hp_mux_text);
-
-static const struct snd_kcontrol_new hpl_mux =
-	WM8994_HP_ENUM("Left Headphone Mux", hpl_enum);
-
-static const struct soc_enum hpr_enum =
-	SOC_ENUM_SINGLE(WM8994_OUTPUT_MIXER_2, 8, 2, hp_mux_text);
-
-static const struct snd_kcontrol_new hpr_mux =
-	WM8994_HP_ENUM("Right Headphone Mux", hpr_enum);
-
 static const char *adc_mux_text[] = {
 	"ADC",
 	"DMIC",
@@ -1430,7 +1374,7 @@ static int wm8994_put_class_w(struct snd_kcontrol *kcontrol,
 
 	ret = snd_soc_dapm_put_volsw(kcontrol, ucontrol);
 
-	wm8994_update_class_w(codec);
+	wm_hubs_update_class_w(codec);
 
 	return ret;
 }
@@ -1524,7 +1468,7 @@ static const struct snd_kcontrol_new wm8958_aif3adc_mux =
 	SOC_DAPM_ENUM("AIF3ADC Mux", wm8958_aif3adc_enum);
 
 static const char *mono_pcm_out_text[] = {
-	"None", "AIF2ADCL", "AIF2ADCR", 
+	"None", "AIF2ADCL", "AIF2ADCR",
 };
 
 static const struct soc_enum mono_pcm_out_enum =
@@ -1573,9 +1517,9 @@ SND_SOC_DAPM_MIXER_E("SPKL", WM8994_POWER_MANAGEMENT_3, 8, 0,
 SND_SOC_DAPM_MIXER_E("SPKR", WM8994_POWER_MANAGEMENT_3, 9, 0,
 		     right_speaker_mixer, ARRAY_SIZE(right_speaker_mixer),
 		     late_enable_ev, SND_SOC_DAPM_PRE_PMU),
-SND_SOC_DAPM_MUX_E("Left Headphone Mux", SND_SOC_NOPM, 0, 0, &hpl_mux,
+SND_SOC_DAPM_MUX_E("Left Headphone Mux", SND_SOC_NOPM, 0, 0, &wm_hubs_hpl_mux,
 		   late_enable_ev, SND_SOC_DAPM_PRE_PMU),
-SND_SOC_DAPM_MUX_E("Right Headphone Mux", SND_SOC_NOPM, 0, 0, &hpr_mux,
+SND_SOC_DAPM_MUX_E("Right Headphone Mux", SND_SOC_NOPM, 0, 0, &wm_hubs_hpr_mux,
 		   late_enable_ev, SND_SOC_DAPM_PRE_PMU),
 
 SND_SOC_DAPM_POST("Late Disable PGA", late_disable_ev)
@@ -1591,8 +1535,8 @@ SND_SOC_DAPM_MIXER("SPKL", WM8994_POWER_MANAGEMENT_3, 8, 0,
 		   left_speaker_mixer, ARRAY_SIZE(left_speaker_mixer)),
 SND_SOC_DAPM_MIXER("SPKR", WM8994_POWER_MANAGEMENT_3, 9, 0,
 		   right_speaker_mixer, ARRAY_SIZE(right_speaker_mixer)),
-SND_SOC_DAPM_MUX("Left Headphone Mux", SND_SOC_NOPM, 0, 0, &hpl_mux),
-SND_SOC_DAPM_MUX("Right Headphone Mux", SND_SOC_NOPM, 0, 0, &hpr_mux),
+SND_SOC_DAPM_MUX("Left Headphone Mux", SND_SOC_NOPM, 0, 0, &wm_hubs_hpl_mux),
+SND_SOC_DAPM_MUX("Right Headphone Mux", SND_SOC_NOPM, 0, 0, &wm_hubs_hpr_mux),
 };
 
 static const struct snd_soc_dapm_widget wm8994_dac_revd_widgets[] = {
@@ -1732,6 +1676,7 @@ SND_SOC_DAPM_MUX("AIF3ADC Mux", SND_SOC_NOPM, 0, 0, &wm8994_aif3adc_mux),
 };
 
 static const struct snd_soc_dapm_widget wm8958_dapm_widgets[] = {
+SND_SOC_DAPM_SUPPLY("AIF3", WM8994_POWER_MANAGEMENT_6, 5, 1, NULL, 0),
 SND_SOC_DAPM_MUX("Mono PCM Out Mux", SND_SOC_NOPM, 0, 0, &mono_pcm_out_mux),
 SND_SOC_DAPM_MUX("AIF2DACL Mux", SND_SOC_NOPM, 0, 0, &aif2dacl_src_mux),
 SND_SOC_DAPM_MUX("AIF2DACR Mux", SND_SOC_NOPM, 0, 0, &aif2dacr_src_mux),
@@ -1972,6 +1917,9 @@ static const struct snd_soc_dapm_route wm8958_intercon[] = {
 	{ "AIF2DACR Mux", "AIF2", "AIF2DAC Mux" },
 	{ "AIF2DACR Mux", "AIF3", "AIF3DACDAT" },
 
+	{ "AIF3DACDAT", NULL, "AIF3" },
+	{ "AIF3ADCDAT", NULL, "AIF3" },
+
 	{ "Mono PCM Out Mux", "AIF2ADCL", "AIF2ADCL" },
 	{ "Mono PCM Out Mux", "AIF2ADCR", "AIF2ADCR" },
 
@@ -2068,24 +2016,20 @@ static int _wm8994_set_fll(struct snd_soc_codec *codec, int id, int src,
 	struct wm8994 *control = wm8994->wm8994;
 	int reg_offset, ret;
 	struct fll_div fll;
-	u16 reg, aif1, aif2;
+	u16 reg, clk1, aif_reg, aif_src;
 	unsigned long timeout;
 	bool was_enabled;
 
-	aif1 = snd_soc_read(codec, WM8994_AIF1_CLOCKING_1)
-		& WM8994_AIF1CLK_ENA;
-
-	aif2 = snd_soc_read(codec, WM8994_AIF2_CLOCKING_1)
-		& WM8994_AIF2CLK_ENA;
-
 	switch (id) {
 	case WM8994_FLL1:
 		reg_offset = 0;
 		id = 0;
+		aif_src = 0x10;
 		break;
 	case WM8994_FLL2:
 		reg_offset = 0x20;
 		id = 1;
+		aif_src = 0x18;
 		break;
 	default:
 		return -EINVAL;
@@ -2127,16 +2071,33 @@ static int _wm8994_set_fll(struct snd_soc_codec *codec, int id, int src,
 	if (ret < 0)
 		return ret;
 
-	/* Gate the AIF clocks while we reclock */
-	snd_soc_update_bits(codec, WM8994_AIF1_CLOCKING_1,
-			    WM8994_AIF1CLK_ENA, 0);
-	snd_soc_update_bits(codec, WM8994_AIF2_CLOCKING_1,
-			    WM8994_AIF2CLK_ENA, 0);
+	/* Make sure that we're not providing SYSCLK right now */
+	clk1 = snd_soc_read(codec, WM8994_CLOCKING_1);
+	if (clk1 & WM8994_SYSCLK_SRC)
+		aif_reg = WM8994_AIF2_CLOCKING_1;
+	else
+		aif_reg = WM8994_AIF1_CLOCKING_1;
+	reg = snd_soc_read(codec, aif_reg);
+
+	if ((reg & WM8994_AIF1CLK_ENA) &&
+	    (reg & WM8994_AIF1CLK_SRC_MASK) == aif_src) {
+		dev_err(codec->dev, "FLL%d is currently providing SYSCLK\n",
+			id + 1);
+		return -EBUSY;
+	}
 
 	/* We always need to disable the FLL while reconfiguring */
 	snd_soc_update_bits(codec, WM8994_FLL1_CONTROL_1 + reg_offset,
 			    WM8994_FLL1_ENA, 0);
 
+	if (wm8994->fll_byp && src == WM8994_FLL_SRC_BCLK &&
+	    freq_in == freq_out && freq_out) {
+		dev_dbg(codec->dev, "Bypassing FLL%d\n", id + 1);
+		snd_soc_update_bits(codec, WM8994_FLL1_CONTROL_5 + reg_offset,
+				    WM8958_FLL1_BYP, WM8958_FLL1_BYP);
+		goto out;
+	}
+
 	reg = (fll.outdiv << WM8994_FLL1_OUTDIV_SHIFT) |
 		(fll.fll_fratio << WM8994_FLL1_FRATIO_SHIFT);
 	snd_soc_update_bits(codec, WM8994_FLL1_CONTROL_2 + reg_offset,
@@ -2151,6 +2112,7 @@ static int _wm8994_set_fll(struct snd_soc_codec *codec, int id, int src,
 				    fll.n << WM8994_FLL1_N_SHIFT);
 
 	snd_soc_update_bits(codec, WM8994_FLL1_CONTROL_5 + reg_offset,
+			    WM8958_FLL1_BYP |
 			    WM8994_FLL1_REFCLK_DIV_MASK |
 			    WM8994_FLL1_REFCLK_SRC_MASK,
 			    (fll.clk_ref_div << WM8994_FLL1_REFCLK_DIV_SHIFT) |
@@ -2213,16 +2175,11 @@ static int _wm8994_set_fll(struct snd_soc_codec *codec, int id, int src,
 		}
 	}
 
+out:
 	wm8994->fll[id].in = freq_in;
 	wm8994->fll[id].out = freq_out;
 	wm8994->fll[id].src = src;
 
-	/* Enable any gated AIF clocks */
-	snd_soc_update_bits(codec, WM8994_AIF1_CLOCKING_1,
-			    WM8994_AIF1CLK_ENA, aif1);
-	snd_soc_update_bits(codec, WM8994_AIF2_CLOCKING_1,
-			    WM8994_AIF2CLK_ENA, aif2);
-
 	configure_clock(codec);
 
 	return 0;
@@ -2290,7 +2247,7 @@ static int wm8994_set_dai_sysclk(struct snd_soc_dai *dai,
 
 	case WM8994_SYSCLK_OPCLK:
 		/* Special case - a division (times 10) is given and
-		 * no effect on main clocking. 
+		 * no effect on main clocking.
 		 */
 		if (freq) {
 			for (i = 0; i < ARRAY_SIZE(opclk_divs); i++)
@@ -2792,33 +2749,6 @@ static int wm8994_aif3_hw_params(struct snd_pcm_substream *substream,
 	return snd_soc_update_bits(codec, aif1_reg, WM8994_AIF1_WL_MASK, aif1);
 }
 
-static void wm8994_aif_shutdown(struct snd_pcm_substream *substream,
-				struct snd_soc_dai *dai)
-{
-	struct snd_soc_codec *codec = dai->codec;
-	int rate_reg = 0;
-
-	switch (dai->id) {
-	case 1:
-		rate_reg = WM8994_AIF1_RATE;
-		break;
-	case 2:
-		rate_reg = WM8994_AIF2_RATE;
-		break;
-	default:
-		break;
-	}
-
-	/* If the DAI is idle then configure the divider tree for the
-	 * lowest output rate to save a little power if the clock is
-	 * still active (eg, because it is system clock).
-	 */
-	if (rate_reg && !dai->playback_active && !dai->capture_active)
-		snd_soc_update_bits(codec, rate_reg,
-				    WM8994_AIF1_SR_MASK |
-				    WM8994_AIF1CLK_RATE_MASK, 0x9);
-}
-
 static int wm8994_aif_mute(struct snd_soc_dai *codec_dai, int mute)
 {
 	struct snd_soc_codec *codec = codec_dai->codec;
@@ -2860,10 +2790,6 @@ static int wm8994_set_tristate(struct snd_soc_dai *codec_dai, int tristate)
 		reg = WM8994_AIF2_MASTER_SLAVE;
 		mask = WM8994_AIF2_TRI;
 		break;
-	case 3:
-		reg = WM8994_POWER_MANAGEMENT_6;
-		mask = WM8994_AIF3_TRI;
-		break;
 	default:
 		return -EINVAL;
 	}
@@ -2900,7 +2826,6 @@ static const struct snd_soc_dai_ops wm8994_aif1_dai_ops = {
 	.set_sysclk	= wm8994_set_dai_sysclk,
 	.set_fmt	= wm8994_set_dai_fmt,
 	.hw_params	= wm8994_hw_params,
-	.shutdown	= wm8994_aif_shutdown,
 	.digital_mute	= wm8994_aif_mute,
 	.set_pll	= wm8994_set_fll,
 	.set_tristate	= wm8994_set_tristate,
@@ -2910,7 +2835,6 @@ static const struct snd_soc_dai_ops wm8994_aif2_dai_ops = {
 	.set_sysclk	= wm8994_set_dai_sysclk,
 	.set_fmt	= wm8994_set_dai_fmt,
 	.hw_params	= wm8994_hw_params,
-	.shutdown	= wm8994_aif_shutdown,
 	.digital_mute   = wm8994_aif_mute,
 	.set_pll	= wm8994_set_fll,
 	.set_tristate	= wm8994_set_tristate,
@@ -2918,7 +2842,6 @@ static const struct snd_soc_dai_ops wm8994_aif2_dai_ops = {
 
 static const struct snd_soc_dai_ops wm8994_aif3_dai_ops = {
 	.hw_params	= wm8994_aif3_hw_params,
-	.set_tristate	= wm8994_set_tristate,
 };
 
 static struct snd_soc_dai_driver wm8994_dai[] = {
@@ -3126,14 +3049,14 @@ static void wm8994_handle_retune_mobile_pdata(struct wm8994_priv *wm8994)
 
 		/* Expand the array... */
 		t = krealloc(wm8994->retune_mobile_texts,
-			     sizeof(char *) * 
+			     sizeof(char *) *
 			     (wm8994->num_retune_mobile_texts + 1),
 			     GFP_KERNEL);
 		if (t == NULL)
 			continue;
 
 		/* ...store the new entry... */
-		t[wm8994->num_retune_mobile_texts] = 
+		t[wm8994->num_retune_mobile_texts] =
 			pdata->retune_mobile_cfgs[i].name;
 
 		/* ...and remember the new version. */
@@ -3304,25 +3227,25 @@ int wm8994_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack,
 }
 EXPORT_SYMBOL_GPL(wm8994_mic_detect);
 
-static irqreturn_t wm8994_mic_irq(int irq, void *data)
+static void wm8994_mic_work(struct work_struct *work)
 {
-	struct wm8994_priv *priv = data;
-	struct snd_soc_codec *codec = priv->codec;
-	int reg;
+	struct wm8994_priv *priv = container_of(work,
+						struct wm8994_priv,
+						mic_work.work);
+	struct regmap *regmap = priv->wm8994->regmap;
+	struct device *dev = priv->wm8994->dev;
+	unsigned int reg;
+	int ret;
 	int report;
 
-#ifndef CONFIG_SND_SOC_WM8994_MODULE
-	trace_snd_soc_jack_irq(dev_name(codec->dev));
-#endif
-
-	reg = snd_soc_read(codec, WM8994_INTERRUPT_RAW_STATUS_2);
-	if (reg < 0) {
-		dev_err(codec->dev, "Failed to read microphone status: %d\n",
-			reg);
-		return IRQ_HANDLED;
+	ret = regmap_read(regmap, WM8994_INTERRUPT_RAW_STATUS_2, &reg);
+	if (ret < 0) {
+		dev_err(dev, "Failed to read microphone status: %d\n",
+			ret);
+		return;
 	}
 
-	dev_dbg(codec->dev, "Microphone status: %x\n", reg);
+	dev_dbg(dev, "Microphone status: %x\n", reg);
 
 	report = 0;
 	if (reg & WM8994_MIC1_DET_STS) {
@@ -3361,6 +3284,20 @@ static irqreturn_t wm8994_mic_irq(int irq, void *data)
 
 	snd_soc_jack_report(priv->micdet[1].jack, report,
 			    SND_JACK_HEADSET | SND_JACK_BTN_0);
+}
+
+static irqreturn_t wm8994_mic_irq(int irq, void *data)
+{
+	struct wm8994_priv *priv = data;
+	struct snd_soc_codec *codec = priv->codec;
+
+#ifndef CONFIG_SND_SOC_WM8994_MODULE
+	trace_snd_soc_jack_irq(dev_name(codec->dev));
+#endif
+
+	pm_wakeup_event(codec->dev, 300);
+
+	schedule_delayed_work(&priv->mic_work, msecs_to_jiffies(250));
 
 	return IRQ_HANDLED;
 }
@@ -3415,9 +3352,6 @@ static void wm8958_default_micdet(u16 status, void *data)
 
 		wm8958_micd_set_rate(codec);
 
-		snd_soc_jack_report(wm8994->micdet[0].jack, SND_JACK_HEADPHONE,
-				    SND_JACK_HEADSET);
-
 		/* If we have jackdet that will detect removal */
 		if (wm8994->jackdet) {
 			mutex_lock(&wm8994->accdet_lock);
@@ -3430,14 +3364,13 @@ static void wm8958_default_micdet(u16 status, void *data)
 
 			mutex_unlock(&wm8994->accdet_lock);
 
-			if (wm8994->pdata->jd_ext_cap) {
-				mutex_lock(&codec->mutex);
+			if (wm8994->pdata->jd_ext_cap)
 				snd_soc_dapm_disable_pin(&codec->dapm,
 							 "MICBIAS2");
-				snd_soc_dapm_sync(&codec->dapm);
-				mutex_unlock(&codec->mutex);
-			}
 		}
+
+		snd_soc_jack_report(wm8994->micdet[0].jack, SND_JACK_HEADPHONE,
+				    SND_JACK_HEADSET);
 	}
 
 	/* Report short circuit as a button */
@@ -3489,6 +3422,8 @@ static irqreturn_t wm1811_jackdet_irq(int irq, void *data)
 	if (present) {
 		dev_dbg(codec->dev, "Jack detected\n");
 
+		wm8958_micd_set_rate(codec);
+
 		snd_soc_update_bits(codec, WM8958_MICBIAS2,
 				    WM8958_MICB2_DISCH, 0);
 
@@ -3526,16 +3461,11 @@ static irqreturn_t wm1811_jackdet_irq(int irq, void *data)
 
 	/* If required for an external cap force MICBIAS on */
 	if (wm8994->pdata->jd_ext_cap) {
-		mutex_lock(&codec->mutex);
-
 		if (present)
 			snd_soc_dapm_force_enable_pin(&codec->dapm,
 						      "MICBIAS2");
 		else
 			snd_soc_dapm_disable_pin(&codec->dapm, "MICBIAS2");
-
-		snd_soc_dapm_sync(&codec->dapm);
-		mutex_unlock(&codec->mutex);
 	}
 
 	if (present)
@@ -3740,6 +3670,7 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec)
 	wm8994->codec = codec;
 
 	mutex_init(&wm8994->accdet_lock);
+	INIT_DELAYED_WORK(&wm8994->mic_work, wm8994_mic_work);
 
 	for (i = 0; i < ARRAY_SIZE(wm8994->fll_locked); i++)
 		init_completion(&wm8994->fll_locked[i]);
@@ -3783,13 +3714,22 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec)
 	case WM8958:
 		wm8994->hubs.dcs_readback_mode = 1;
 		wm8994->hubs.hp_startup_mode = 1;
+
+		switch (wm8994->revision) {
+		case 0:
+			break;
+		default:
+			wm8994->fll_byp = true;
+			break;
+		}
 		break;
 
 	case WM1811:
 		wm8994->hubs.dcs_readback_mode = 2;
 		wm8994->hubs.no_series_update = 1;
 		wm8994->hubs.hp_startup_mode = 1;
-		wm8994->hubs.no_cache_class_w = true;
+		wm8994->hubs.no_cache_dac_hp_direct = true;
+		wm8994->fll_byp = true;
 
 		switch (wm8994->revision) {
 		case 0:
@@ -4010,7 +3950,8 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec)
 		break;
 	}
 
-	wm8994_update_class_w(codec);
+	wm8994->hubs.check_class_w_digital = wm8994_check_class_w_digital;
+	wm_hubs_update_class_w(codec);
 
 	wm8994_handle_pdata(wm8994);
 
@@ -4075,7 +4016,6 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec)
 					  ARRAY_SIZE(wm8994_dac_widgets));
 		break;
 	}
-		
 
 	wm_hubs_add_analogue_routes(codec, 0, 0);
 	snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon));
@@ -4140,7 +4080,7 @@ err_irq:
 	return ret;
 }
 
-static int  wm8994_codec_remove(struct snd_soc_codec *codec)
+static int wm8994_codec_remove(struct snd_soc_codec *codec)
 {
 	struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
 	struct wm8994 *control = wm8994->wm8994;
@@ -4181,14 +4121,10 @@ static int  wm8994_codec_remove(struct snd_soc_codec *codec)
 			free_irq(wm8994->micdet_irq, wm8994);
 		break;
 	}
-	if (wm8994->mbc)
-		release_firmware(wm8994->mbc);
-	if (wm8994->mbc_vss)
-		release_firmware(wm8994->mbc_vss);
-	if (wm8994->enh_eq)
-		release_firmware(wm8994->enh_eq);
+	release_firmware(wm8994->mbc);
+	release_firmware(wm8994->mbc_vss);
+	release_firmware(wm8994->enh_eq);
 	kfree(wm8994->retune_mobile_texts);
-
 	return 0;
 }
 
diff --git a/sound/soc/codecs/wm8994.h b/sound/soc/codecs/wm8994.h
index c724112998d8..d77e06f0a675 100644
--- a/sound/soc/codecs/wm8994.h
+++ b/sound/soc/codecs/wm8994.h
@@ -12,6 +12,7 @@
 #include <sound/soc.h>
 #include <linux/firmware.h>
 #include <linux/completion.h>
+#include <linux/workqueue.h>
 
 #include "wm_hubs.h"
 
@@ -79,6 +80,7 @@ struct wm8994_priv {
 	struct wm8994_fll_config fll[2], fll_suspend[2];
 	struct completion fll_locked[2];
 	bool fll_locked_irq;
+	bool fll_byp;
 
 	int vmid_refcount;
 	int active_refcount;
@@ -126,6 +128,7 @@ struct wm8994_priv {
 
 	struct mutex accdet_lock;
 	struct wm8994_micdet micdet[2];
+	struct delayed_work mic_work;
 	bool mic_detecting;
 	bool jack_mic;
 	int btn_mask;
diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c
index 1fd635494045..8af422e38fd0 100644
--- a/sound/soc/codecs/wm8996.c
+++ b/sound/soc/codecs/wm8996.c
@@ -1770,7 +1770,13 @@ static int wm8996_set_bias_level(struct snd_soc_codec *codec,
 
 	switch (level) {
 	case SND_SOC_BIAS_ON:
+		break;
 	case SND_SOC_BIAS_PREPARE:
+		/* Put the MICBIASes into regulating mode */
+		snd_soc_update_bits(codec, WM8996_MICBIAS_1,
+				    WM8996_MICB1_MODE, 0);
+		snd_soc_update_bits(codec, WM8996_MICBIAS_2,
+				    WM8996_MICB2_MODE, 0);
 		break;
 
 	case SND_SOC_BIAS_STANDBY:
@@ -1793,6 +1799,12 @@ static int wm8996_set_bias_level(struct snd_soc_codec *codec,
 			regcache_cache_only(codec->control_data, false);
 			regcache_sync(codec->control_data);
 		}
+
+		/* Bypass the MICBIASes for lowest power */
+		snd_soc_update_bits(codec, WM8996_MICBIAS_1,
+				    WM8996_MICB1_MODE, WM8996_MICB1_MODE);
+		snd_soc_update_bits(codec, WM8996_MICBIAS_2,
+				    WM8996_MICB2_MODE, WM8996_MICB2_MODE);
 		break;
 
 	case SND_SOC_BIAS_OFF:
diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c
index 076c126ed9b1..9328270df16c 100644
--- a/sound/soc/codecs/wm9081.c
+++ b/sound/soc/codecs/wm9081.c
@@ -774,7 +774,7 @@ static const struct snd_soc_dapm_widget wm9081_dapm_widgets[] = {
 SND_SOC_DAPM_INPUT("IN1"),
 SND_SOC_DAPM_INPUT("IN2"),
 
-SND_SOC_DAPM_DAC("DAC", "HiFi Playback", WM9081_POWER_MANAGEMENT, 0, 0),
+SND_SOC_DAPM_DAC("DAC", NULL, WM9081_POWER_MANAGEMENT, 0, 0),
 
 SND_SOC_DAPM_MIXER_NAMED_CTL("Mixer", SND_SOC_NOPM, 0, 0,
 			     mixer, ARRAY_SIZE(mixer)),
@@ -799,6 +799,7 @@ SND_SOC_DAPM_SUPPLY("TSENSE", WM9081_POWER_MANAGEMENT, 7, 0, NULL, 0),
 static const struct snd_soc_dapm_route wm9081_audio_paths[] = {
 	{ "DAC", NULL, "CLK_SYS" },
 	{ "DAC", NULL, "CLK_DSP" },
+	{ "DAC", NULL, "AIF" },
 
 	{ "Mixer", "IN1 Switch", "IN1" },
 	{ "Mixer", "IN2 Switch", "IN2" },
@@ -1252,7 +1253,7 @@ static const struct snd_soc_dai_ops wm9081_dai_ops = {
 static struct snd_soc_dai_driver wm9081_dai = {
 	.name = "wm9081-hifi",
 	.playback = {
-		.stream_name = "HiFi Playback",
+		.stream_name = "AIF",
 		.channels_min = 1,
 		.channels_max = 2,
 		.rates = WM9081_RATES,
diff --git a/sound/soc/codecs/wm9705.c b/sound/soc/codecs/wm9705.c
index cacc6a86b46f..e8e782a0c78d 100644
--- a/sound/soc/codecs/wm9705.c
+++ b/sound/soc/codecs/wm9705.c
@@ -236,9 +236,7 @@ static int ac97_write(struct snd_soc_codec *codec, unsigned int reg,
 static int ac97_prepare(struct snd_pcm_substream *substream,
 			struct snd_soc_dai *dai)
 {
-	struct snd_pcm_runtime *runtime = substream->runtime;
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_codec *codec = rtd->codec;
+	struct snd_soc_codec *codec = dai->codec;
 	int reg;
 	u16 vra;
 
@@ -250,7 +248,7 @@ static int ac97_prepare(struct snd_pcm_substream *substream,
 	else
 		reg = AC97_PCM_LR_ADC_RATE;
 
-	return ac97_write(codec, reg, runtime->rate);
+	return ac97_write(codec, reg, substream->runtime->rate);
 }
 
 #define WM9705_AC97_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | \
diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c
index b342ae50bcd6..a1541414d904 100644
--- a/sound/soc/codecs/wm9712.c
+++ b/sound/soc/codecs/wm9712.c
@@ -467,11 +467,10 @@ static int ac97_write(struct snd_soc_codec *codec, unsigned int reg,
 static int ac97_prepare(struct snd_pcm_substream *substream,
 			struct snd_soc_dai *dai)
 {
-	struct snd_pcm_runtime *runtime = substream->runtime;
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_codec *codec =rtd->codec;
+	struct snd_soc_codec *codec = dai->codec;
 	int reg;
 	u16 vra;
+	struct snd_pcm_runtime *runtime = substream->runtime;
 
 	vra = ac97_read(codec, AC97_EXTENDED_STATUS);
 	ac97_write(codec, AC97_EXTENDED_STATUS, vra | 0x1);
@@ -487,10 +486,9 @@ static int ac97_prepare(struct snd_pcm_substream *substream,
 static int ac97_aux_prepare(struct snd_pcm_substream *substream,
 			    struct snd_soc_dai *dai)
 {
-	struct snd_pcm_runtime *runtime = substream->runtime;
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_codec *codec = rtd->codec;
+	struct snd_soc_codec *codec = dai->codec;
 	u16 vra, xsle;
+	struct snd_pcm_runtime *runtime = substream->runtime;
 
 	vra = ac97_read(codec, AC97_EXTENDED_STATUS);
 	ac97_write(codec, AC97_EXTENDED_STATUS, vra | 0x1);
diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c
index 6c028c470601..dfe957a47f29 100644
--- a/sound/soc/codecs/wm_hubs.c
+++ b/sound/soc/codecs/wm_hubs.c
@@ -109,12 +109,103 @@ irqreturn_t wm_hubs_dcs_done(int irq, void *data)
 }
 EXPORT_SYMBOL_GPL(wm_hubs_dcs_done);
 
+static bool wm_hubs_dac_hp_direct(struct snd_soc_codec *codec)
+{
+	int reg;
+
+	/* If we're going via the mixer we'll need to do additional checks */
+	reg = snd_soc_read(codec, WM8993_OUTPUT_MIXER1);
+	if (!(reg & WM8993_DACL_TO_HPOUT1L)) {
+		if (reg & ~WM8993_DACL_TO_MIXOUTL) {
+			dev_vdbg(codec->dev, "Analogue paths connected: %x\n",
+				 reg & ~WM8993_DACL_TO_HPOUT1L);
+			return false;
+		} else {
+			dev_vdbg(codec->dev, "HPL connected to mixer\n");
+		}
+	} else {
+		dev_vdbg(codec->dev, "HPL connected to DAC\n");
+	}
+
+	reg = snd_soc_read(codec, WM8993_OUTPUT_MIXER2);
+	if (!(reg & WM8993_DACR_TO_HPOUT1R)) {
+		if (reg & ~WM8993_DACR_TO_MIXOUTR) {
+			dev_vdbg(codec->dev, "Analogue paths connected: %x\n",
+				 reg & ~WM8993_DACR_TO_HPOUT1R);
+			return false;
+		} else {
+			dev_vdbg(codec->dev, "HPR connected to mixer\n");
+		}
+	} else {
+		dev_vdbg(codec->dev, "HPR connected to DAC\n");
+	}
+
+	return true;
+}
+
+struct wm_hubs_dcs_cache {
+	struct list_head list;
+	unsigned int left;
+	unsigned int right;
+	u16 dcs_cfg;
+};
+
+static bool wm_hubs_dcs_cache_get(struct snd_soc_codec *codec,
+				  struct wm_hubs_dcs_cache **entry)
+{
+	struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec);
+	struct wm_hubs_dcs_cache *cache;
+	unsigned int left, right;
+
+	left = snd_soc_read(codec, WM8993_LEFT_OUTPUT_VOLUME);
+	left &= WM8993_HPOUT1L_VOL_MASK;
+
+	right = snd_soc_read(codec, WM8993_RIGHT_OUTPUT_VOLUME);
+	right &= WM8993_HPOUT1R_VOL_MASK;
+
+	list_for_each_entry(cache, &hubs->dcs_cache, list) {
+		if (cache->left != left || cache->right != right)
+			continue;
+
+		*entry = cache;
+		return true;
+	}
+
+	return false;
+}
+
+static void wm_hubs_dcs_cache_set(struct snd_soc_codec *codec, u16 dcs_cfg)
+{
+	struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec);
+	struct wm_hubs_dcs_cache *cache;
+
+	if (hubs->no_cache_dac_hp_direct)
+		return;
+
+	cache = devm_kzalloc(codec->dev, sizeof(*cache), GFP_KERNEL);
+	if (!cache) {
+		dev_err(codec->dev, "Failed to allocate DCS cache entry\n");
+		return;
+	}
+
+	cache->left = snd_soc_read(codec, WM8993_LEFT_OUTPUT_VOLUME);
+	cache->left &= WM8993_HPOUT1L_VOL_MASK;
+
+	cache->right = snd_soc_read(codec, WM8993_RIGHT_OUTPUT_VOLUME);
+	cache->right &= WM8993_HPOUT1R_VOL_MASK;
+
+	cache->dcs_cfg = dcs_cfg;
+
+	list_add_tail(&cache->list, &hubs->dcs_cache);
+}
+
 /*
  * Startup calibration of the DC servo
  */
 static void calibrate_dc_servo(struct snd_soc_codec *codec)
 {
 	struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec);
+	struct wm_hubs_dcs_cache *cache;
 	s8 offset;
 	u16 reg, reg_l, reg_r, dcs_cfg, dcs_reg;
 
@@ -129,10 +220,11 @@ static void calibrate_dc_servo(struct snd_soc_codec *codec)
 
 	/* If we're using a digital only path and have a previously
 	 * callibrated DC servo offset stored then use that. */
-	if (hubs->class_w && hubs->class_w_dcs) {
-		dev_dbg(codec->dev, "Using cached DC servo offset %x\n",
-			hubs->class_w_dcs);
-		snd_soc_write(codec, dcs_reg, hubs->class_w_dcs);
+	if (wm_hubs_dac_hp_direct(codec) &&
+	    wm_hubs_dcs_cache_get(codec, &cache)) {
+		dev_dbg(codec->dev, "Using cached DCS offset %x for %d,%d\n",
+			cache->dcs_cfg, cache->left, cache->right);
+		snd_soc_write(codec, dcs_reg, cache->dcs_cfg);
 		wait_for_dc_servo(codec,
 				  WM8993_DCS_TRIG_DAC_WR_0 |
 				  WM8993_DCS_TRIG_DAC_WR_1);
@@ -207,8 +299,8 @@ static void calibrate_dc_servo(struct snd_soc_codec *codec)
 
 	/* Save the callibrated offset if we're in class W mode and
 	 * therefore don't have any analogue signal mixed in. */
-	if (hubs->class_w && !hubs->no_cache_class_w)
-		hubs->class_w_dcs = dcs_cfg;
+	if (wm_hubs_dac_hp_direct(codec))
+		wm_hubs_dcs_cache_set(codec, dcs_cfg);
 }
 
 /*
@@ -223,9 +315,6 @@ static int wm8993_put_dc_servo(struct snd_kcontrol *kcontrol,
 
 	ret = snd_soc_put_volsw(kcontrol, ucontrol);
 
-	/* Updating the analogue gains invalidates the DC servo cache */
-	hubs->class_w_dcs = 0;
-
 	/* If we're applying an offset correction then updating the
 	 * callibration would be likely to introduce further offsets. */
 	if (hubs->dcs_codes_l || hubs->dcs_codes_r || hubs->no_series_update)
@@ -530,6 +619,86 @@ static int lineout_event(struct snd_soc_dapm_widget *w,
 	return 0;
 }
 
+void wm_hubs_update_class_w(struct snd_soc_codec *codec)
+{
+	struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec);
+	int enable = WM8993_CP_DYN_V | WM8993_CP_DYN_FREQ;
+
+	if (!wm_hubs_dac_hp_direct(codec))
+		enable = false;
+
+	if (hubs->check_class_w_digital && !hubs->check_class_w_digital(codec))
+		enable = false;
+
+	dev_vdbg(codec->dev, "Class W %s\n", enable ? "enabled" : "disabled");
+
+	snd_soc_update_bits(codec, WM8993_CLASS_W_0,
+			    WM8993_CP_DYN_V | WM8993_CP_DYN_FREQ, enable);
+}
+EXPORT_SYMBOL_GPL(wm_hubs_update_class_w);
+
+#define WM_HUBS_SINGLE_W(xname, reg, shift, max, invert) \
+{	.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
+	.info = snd_soc_info_volsw, \
+	.get = snd_soc_dapm_get_volsw, .put = class_w_put_volsw, \
+	.private_value =  SOC_SINGLE_VALUE(reg, shift, max, invert) }
+
+static int class_w_put_volsw(struct snd_kcontrol *kcontrol,
+			      struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol);
+	struct snd_soc_dapm_widget *widget = wlist->widgets[0];
+	struct snd_soc_codec *codec = widget->codec;
+	int ret;
+
+	ret = snd_soc_dapm_put_volsw(kcontrol, ucontrol);
+
+	wm_hubs_update_class_w(codec);
+
+	return ret;
+}
+
+#define WM_HUBS_ENUM_W(xname, xenum) \
+{	.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
+	.info = snd_soc_info_enum_double, \
+	.get = snd_soc_dapm_get_enum_double, \
+	.put = class_w_put_double, \
+	.private_value = (unsigned long)&xenum }
+
+static int class_w_put_double(struct snd_kcontrol *kcontrol,
+			      struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol);
+	struct snd_soc_dapm_widget *widget = wlist->widgets[0];
+	struct snd_soc_codec *codec = widget->codec;
+	int ret;
+
+	ret = snd_soc_dapm_put_enum_double(kcontrol, ucontrol);
+
+	wm_hubs_update_class_w(codec);
+
+	return ret;
+}
+
+static const char *hp_mux_text[] = {
+	"Mixer",
+	"DAC",
+};
+
+static const struct soc_enum hpl_enum =
+	SOC_ENUM_SINGLE(WM8993_OUTPUT_MIXER1, 8, 2, hp_mux_text);
+
+const struct snd_kcontrol_new wm_hubs_hpl_mux =
+	WM_HUBS_ENUM_W("Left Headphone Mux", hpl_enum);
+EXPORT_SYMBOL_GPL(wm_hubs_hpl_mux);
+
+static const struct soc_enum hpr_enum =
+	SOC_ENUM_SINGLE(WM8993_OUTPUT_MIXER2, 8, 2, hp_mux_text);
+
+const struct snd_kcontrol_new wm_hubs_hpr_mux =
+	WM_HUBS_ENUM_W("Right Headphone Mux", hpr_enum);
+EXPORT_SYMBOL_GPL(wm_hubs_hpr_mux);
+
 static const struct snd_kcontrol_new in1l_pga[] = {
 SOC_DAPM_SINGLE("IN1LP Switch", WM8993_INPUT_MIXER2, 5, 1, 0),
 SOC_DAPM_SINGLE("IN1LN Switch", WM8993_INPUT_MIXER2, 4, 1, 0),
@@ -561,25 +730,25 @@ SOC_DAPM_SINGLE("IN1R Switch", WM8993_INPUT_MIXER4, 5, 1, 0),
 };
 
 static const struct snd_kcontrol_new left_output_mixer[] = {
-SOC_DAPM_SINGLE("Right Input Switch", WM8993_OUTPUT_MIXER1, 7, 1, 0),
-SOC_DAPM_SINGLE("Left Input Switch", WM8993_OUTPUT_MIXER1, 6, 1, 0),
-SOC_DAPM_SINGLE("IN2RN Switch", WM8993_OUTPUT_MIXER1, 5, 1, 0),
-SOC_DAPM_SINGLE("IN2LN Switch", WM8993_OUTPUT_MIXER1, 4, 1, 0),
-SOC_DAPM_SINGLE("IN2LP Switch", WM8993_OUTPUT_MIXER1, 1, 1, 0),
-SOC_DAPM_SINGLE("IN1R Switch", WM8993_OUTPUT_MIXER1, 3, 1, 0),
-SOC_DAPM_SINGLE("IN1L Switch", WM8993_OUTPUT_MIXER1, 2, 1, 0),
-SOC_DAPM_SINGLE("DAC Switch", WM8993_OUTPUT_MIXER1, 0, 1, 0),
+WM_HUBS_SINGLE_W("Right Input Switch", WM8993_OUTPUT_MIXER1, 7, 1, 0),
+WM_HUBS_SINGLE_W("Left Input Switch", WM8993_OUTPUT_MIXER1, 6, 1, 0),
+WM_HUBS_SINGLE_W("IN2RN Switch", WM8993_OUTPUT_MIXER1, 5, 1, 0),
+WM_HUBS_SINGLE_W("IN2LN Switch", WM8993_OUTPUT_MIXER1, 4, 1, 0),
+WM_HUBS_SINGLE_W("IN2LP Switch", WM8993_OUTPUT_MIXER1, 1, 1, 0),
+WM_HUBS_SINGLE_W("IN1R Switch", WM8993_OUTPUT_MIXER1, 3, 1, 0),
+WM_HUBS_SINGLE_W("IN1L Switch", WM8993_OUTPUT_MIXER1, 2, 1, 0),
+WM_HUBS_SINGLE_W("DAC Switch", WM8993_OUTPUT_MIXER1, 0, 1, 0),
 };
 
 static const struct snd_kcontrol_new right_output_mixer[] = {
-SOC_DAPM_SINGLE("Left Input Switch", WM8993_OUTPUT_MIXER2, 7, 1, 0),
-SOC_DAPM_SINGLE("Right Input Switch", WM8993_OUTPUT_MIXER2, 6, 1, 0),
-SOC_DAPM_SINGLE("IN2LN Switch", WM8993_OUTPUT_MIXER2, 5, 1, 0),
-SOC_DAPM_SINGLE("IN2RN Switch", WM8993_OUTPUT_MIXER2, 4, 1, 0),
-SOC_DAPM_SINGLE("IN1L Switch", WM8993_OUTPUT_MIXER2, 3, 1, 0),
-SOC_DAPM_SINGLE("IN1R Switch", WM8993_OUTPUT_MIXER2, 2, 1, 0),
-SOC_DAPM_SINGLE("IN2RP Switch", WM8993_OUTPUT_MIXER2, 1, 1, 0),
-SOC_DAPM_SINGLE("DAC Switch", WM8993_OUTPUT_MIXER2, 0, 1, 0),
+WM_HUBS_SINGLE_W("Left Input Switch", WM8993_OUTPUT_MIXER2, 7, 1, 0),
+WM_HUBS_SINGLE_W("Right Input Switch", WM8993_OUTPUT_MIXER2, 6, 1, 0),
+WM_HUBS_SINGLE_W("IN2LN Switch", WM8993_OUTPUT_MIXER2, 5, 1, 0),
+WM_HUBS_SINGLE_W("IN2RN Switch", WM8993_OUTPUT_MIXER2, 4, 1, 0),
+WM_HUBS_SINGLE_W("IN1L Switch", WM8993_OUTPUT_MIXER2, 3, 1, 0),
+WM_HUBS_SINGLE_W("IN1R Switch", WM8993_OUTPUT_MIXER2, 2, 1, 0),
+WM_HUBS_SINGLE_W("IN2RP Switch", WM8993_OUTPUT_MIXER2, 1, 1, 0),
+WM_HUBS_SINGLE_W("DAC Switch", WM8993_OUTPUT_MIXER2, 0, 1, 0),
 };
 
 static const struct snd_kcontrol_new earpiece_mixer[] = {
@@ -943,6 +1112,7 @@ int wm_hubs_add_analogue_routes(struct snd_soc_codec *codec,
 	struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec);
 	struct snd_soc_dapm_context *dapm = &codec->dapm;
 
+	INIT_LIST_HEAD(&hubs->dcs_cache);
 	init_completion(&hubs->dcs_done);
 
 	snd_soc_dapm_add_routes(dapm, analogue_routes,
diff --git a/sound/soc/codecs/wm_hubs.h b/sound/soc/codecs/wm_hubs.h
index 5705276f4943..da2dc899ce6d 100644
--- a/sound/soc/codecs/wm_hubs.h
+++ b/sound/soc/codecs/wm_hubs.h
@@ -16,6 +16,8 @@
 
 #include <linux/completion.h>
 #include <linux/interrupt.h>
+#include <linux/list.h>
+#include <sound/control.h>
 
 struct snd_soc_codec;
 
@@ -30,9 +32,9 @@ struct wm_hubs_data {
 	int series_startup;
 	int no_series_update;
 
-	bool no_cache_class_w;
-	bool class_w;
-	u16 class_w_dcs;
+	bool no_cache_dac_hp_direct;
+	struct list_head dcs_cache;
+	bool (*check_class_w_digital)(struct snd_soc_codec *);
 
 	bool lineout1_se;
 	bool lineout1n_ena;
@@ -58,5 +60,9 @@ extern irqreturn_t wm_hubs_dcs_done(int irq, void *data);
 extern void wm_hubs_vmid_ena(struct snd_soc_codec *codec);
 extern void wm_hubs_set_bias_level(struct snd_soc_codec *codec,
 				   enum snd_soc_bias_level level);
+extern void wm_hubs_update_class_w(struct snd_soc_codec *codec);
+
+extern const struct snd_kcontrol_new wm_hubs_hpl_mux;
+extern const struct snd_kcontrol_new wm_hubs_hpr_mux;
 
 #endif
diff --git a/sound/soc/ep93xx/ep93xx-ac97.c b/sound/soc/ep93xx/ep93xx-ac97.c
index 0678637abd66..bdffab33e160 100644
--- a/sound/soc/ep93xx/ep93xx-ac97.c
+++ b/sound/soc/ep93xx/ep93xx-ac97.c
@@ -87,17 +87,13 @@
  * struct ep93xx_ac97_info - EP93xx AC97 controller info structure
  * @lock: mutex serializing access to the bus (slot 1 & 2 ops)
  * @dev: pointer to the platform device dev structure
- * @mem: physical memory resource for the registers
  * @regs: mapped AC97 controller registers
- * @irq: AC97 interrupt number
  * @done: bus ops wait here for an interrupt
  */
 struct ep93xx_ac97_info {
 	struct mutex		lock;
 	struct device		*dev;
-	struct resource		*mem;
 	void __iomem		*regs;
-	int			irq;
 	struct completion	done;
 };
 
@@ -359,66 +355,50 @@ static struct snd_soc_dai_driver ep93xx_ac97_dai = {
 static int __devinit ep93xx_ac97_probe(struct platform_device *pdev)
 {
 	struct ep93xx_ac97_info *info;
+	struct resource *res;
+	unsigned int irq;
 	int ret;
 
-	info = kzalloc(sizeof(struct ep93xx_ac97_info), GFP_KERNEL);
+	info = devm_kzalloc(&pdev->dev, sizeof(*info), GFP_KERNEL);
 	if (!info)
 		return -ENOMEM;
 
+	res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+	if (!res)
+		return -ENODEV;
+
+	info->regs = devm_request_and_ioremap(&pdev->dev, res);
+	if (!info->regs)
+		return -ENXIO;
+
+	irq = platform_get_irq(pdev, 0);
+	if (!irq)
+		return -ENODEV;
+
+	ret = devm_request_irq(&pdev->dev, irq, ep93xx_ac97_interrupt,
+			       IRQF_TRIGGER_HIGH, pdev->name, info);
+	if (ret)
+		goto fail;
+
 	dev_set_drvdata(&pdev->dev, info);
 
 	mutex_init(&info->lock);
 	init_completion(&info->done);
 	info->dev = &pdev->dev;
 
-	info->mem = platform_get_resource(pdev, IORESOURCE_MEM, 0);
-	if (!info->mem) {
-		ret = -ENXIO;
-		goto fail_free_info;
-	}
-
-	info->irq = platform_get_irq(pdev, 0);
-	if (!info->irq) {
-		ret = -ENXIO;
-		goto fail_free_info;
-	}
-
-	if (!request_mem_region(info->mem->start, resource_size(info->mem),
-				pdev->name)) {
-		ret = -EBUSY;
-		goto fail_free_info;
-	}
-
-	info->regs = ioremap(info->mem->start, resource_size(info->mem));
-	if (!info->regs) {
-		ret = -ENOMEM;
-		goto fail_release_mem;
-	}
-
-	ret = request_irq(info->irq, ep93xx_ac97_interrupt, IRQF_TRIGGER_HIGH,
-			  pdev->name, info);
-	if (ret)
-		goto fail_unmap_mem;
-
 	ep93xx_ac97_info = info;
 	platform_set_drvdata(pdev, info);
 
 	ret = snd_soc_register_dai(&pdev->dev, &ep93xx_ac97_dai);
 	if (ret)
-		goto fail_free_irq;
+		goto fail;
 
 	return 0;
 
-fail_free_irq:
+fail:
 	platform_set_drvdata(pdev, NULL);
-	free_irq(info->irq, info);
-fail_unmap_mem:
-	iounmap(info->regs);
-fail_release_mem:
-	release_mem_region(info->mem->start, resource_size(info->mem));
-fail_free_info:
-	kfree(info);
-
+	ep93xx_ac97_info = NULL;
+	dev_set_drvdata(&pdev->dev, NULL);
 	return ret;
 }
 
@@ -431,11 +411,9 @@ static int __devexit ep93xx_ac97_remove(struct platform_device *pdev)
 	/* disable the AC97 controller */
 	ep93xx_ac97_write_reg(info, AC97GCR, 0);
 
-	free_irq(info->irq, info);
-	iounmap(info->regs);
-	release_mem_region(info->mem->start, resource_size(info->mem));
 	platform_set_drvdata(pdev, NULL);
-	kfree(info);
+	ep93xx_ac97_info = NULL;
+	dev_set_drvdata(&pdev->dev, NULL);
 
 	return 0;
 }
diff --git a/sound/soc/ep93xx/ep93xx-i2s.c b/sound/soc/ep93xx/ep93xx-i2s.c
index f7a62348e3fe..8df8f6dc474f 100644
--- a/sound/soc/ep93xx/ep93xx-i2s.c
+++ b/sound/soc/ep93xx/ep93xx-i2s.c
@@ -63,7 +63,6 @@ struct ep93xx_i2s_info {
 	struct clk			*sclk;
 	struct clk			*lrclk;
 	struct ep93xx_pcm_dma_params	*dma_params;
-	struct resource			*mem;
 	void __iomem			*regs;
 };
 
@@ -373,38 +372,22 @@ static int ep93xx_i2s_probe(struct platform_device *pdev)
 	struct resource *res;
 	int err;
 
-	info = kzalloc(sizeof(struct ep93xx_i2s_info), GFP_KERNEL);
-	if (!info) {
-		err = -ENOMEM;
-		goto fail;
-	}
-
-	dev_set_drvdata(&pdev->dev, info);
-	info->dma_params = ep93xx_i2s_dma_params;
+	info = devm_kzalloc(&pdev->dev, sizeof(*info), GFP_KERNEL);
+	if (!info)
+		return -ENOMEM;
 
 	res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
-	if (!res) {
-		err = -ENODEV;
-		goto fail_free_info;
-	}
+	if (!res)
+		return -ENODEV;
 
-	info->mem = request_mem_region(res->start, resource_size(res),
-				       pdev->name);
-	if (!info->mem) {
-		err = -EBUSY;
-		goto fail_free_info;
-	}
-
-	info->regs = ioremap(info->mem->start, resource_size(info->mem));
-	if (!info->regs) {
-		err = -ENXIO;
-		goto fail_release_mem;
-	}
+	info->regs = devm_request_and_ioremap(&pdev->dev, res);
+	if (!info->regs)
+		return -ENXIO;
 
 	info->mclk = clk_get(&pdev->dev, "mclk");
 	if (IS_ERR(info->mclk)) {
 		err = PTR_ERR(info->mclk);
-		goto fail_unmap_mem;
+		goto fail;
 	}
 
 	info->sclk = clk_get(&pdev->dev, "sclk");
@@ -419,6 +402,9 @@ static int ep93xx_i2s_probe(struct platform_device *pdev)
 		goto fail_put_sclk;
 	}
 
+	dev_set_drvdata(&pdev->dev, info);
+	info->dma_params = ep93xx_i2s_dma_params;
+
 	err = snd_soc_register_dai(&pdev->dev, &ep93xx_i2s_dai);
 	if (err)
 		goto fail_put_lrclk;
@@ -426,17 +412,12 @@ static int ep93xx_i2s_probe(struct platform_device *pdev)
 	return 0;
 
 fail_put_lrclk:
+	dev_set_drvdata(&pdev->dev, NULL);
 	clk_put(info->lrclk);
 fail_put_sclk:
 	clk_put(info->sclk);
 fail_put_mclk:
 	clk_put(info->mclk);
-fail_unmap_mem:
-	iounmap(info->regs);
-fail_release_mem:
-	release_mem_region(info->mem->start, resource_size(info->mem));
-fail_free_info:
-	kfree(info);
 fail:
 	return err;
 }
@@ -446,12 +427,10 @@ static int __devexit ep93xx_i2s_remove(struct platform_device *pdev)
 	struct ep93xx_i2s_info *info = dev_get_drvdata(&pdev->dev);
 
 	snd_soc_unregister_dai(&pdev->dev);
+	dev_set_drvdata(&pdev->dev, NULL);
 	clk_put(info->lrclk);
 	clk_put(info->sclk);
 	clk_put(info->mclk);
-	iounmap(info->regs);
-	release_mem_region(info->mem->start, resource_size(info->mem));
-	kfree(info);
 	return 0;
 }
 
diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig
index d754d34d68a6..d70133086ac3 100644
--- a/sound/soc/fsl/Kconfig
+++ b/sound/soc/fsl/Kconfig
@@ -1,18 +1,31 @@
+config SND_SOC_FSL_SSI
+	tristate
+
+config SND_SOC_FSL_UTILS
+	tristate
+
+menuconfig SND_POWERPC_SOC
+	tristate "SoC Audio for Freescale PowerPC CPUs"
+	depends on FSL_SOC
+	help
+	  Say Y or M if you want to add support for codecs attached to
+	  the PowerPC CPUs.
+
+if SND_POWERPC_SOC
+
 config SND_MPC52xx_DMA
 	tristate
 
-# ASoC platform support for the Freescale PowerPC SOCs that have an SSI and
-# an Elo DMA controller, such as the MPC8610 and P1022.  You will still need to
-# select a platform driver and a codec driver.
-config SND_SOC_POWERPC_SSI
+config SND_SOC_POWERPC_DMA
 	tristate
-	depends on FSL_SOC
 
 config SND_SOC_MPC8610_HPCD
 	tristate "ALSA SoC support for the Freescale MPC8610 HPCD board"
 	# I2C is necessary for the CS4270 driver
 	depends on MPC8610_HPCD && I2C
-	select SND_SOC_POWERPC_SSI
+	select SND_SOC_FSL_SSI
+	select SND_SOC_FSL_UTILS
+	select SND_SOC_POWERPC_DMA
 	select SND_SOC_CS4270
 	select SND_SOC_CS4270_VD33_ERRATA
 	default y if MPC8610_HPCD
@@ -23,7 +36,9 @@ config SND_SOC_P1022_DS
 	tristate "ALSA SoC support for the Freescale P1022 DS board"
 	# I2C is necessary for the WM8776 driver
 	depends on P1022_DS && I2C
-	select SND_SOC_POWERPC_SSI
+	select SND_SOC_FSL_SSI
+	select SND_SOC_FSL_UTILS
+	select SND_SOC_POWERPC_DMA
 	select SND_SOC_WM8776
 	default y if P1022_DS
 	help
@@ -65,3 +80,103 @@ config SND_MPC52xx_SOC_EFIKA
 	help
 	  Say Y if you want to add support for sound on the Efika.
 
+endif # SND_POWERPC_SOC
+
+menuconfig SND_IMX_SOC
+	tristate "SoC Audio for Freescale i.MX CPUs"
+	depends on ARCH_MXC
+	help
+	  Say Y or M if you want to add support for codecs attached to
+	  the i.MX CPUs.
+
+if SND_IMX_SOC
+
+config SND_SOC_IMX_SSI
+	tristate
+
+config SND_SOC_IMX_PCM
+	tristate
+
+config SND_SOC_IMX_PCM_FIQ
+	tristate
+	select FIQ
+	select SND_SOC_IMX_PCM
+
+config SND_SOC_IMX_PCM_DMA
+	tristate
+	select SND_SOC_DMAENGINE_PCM
+	select SND_SOC_IMX_PCM
+
+config SND_SOC_IMX_AUDMUX
+	tristate
+
+config SND_MXC_SOC_WM1133_EV1
+	tristate "Audio on the i.MX31ADS with WM1133-EV1 fitted"
+	depends on MACH_MX31ADS_WM1133_EV1 && EXPERIMENTAL
+	select SND_SOC_WM8350
+	select SND_SOC_IMX_PCM_FIQ
+	select SND_SOC_IMX_AUDMUX
+	select SND_SOC_IMX_SSI
+	help
+	  Enable support for audio on the i.MX31ADS with the WM1133-EV1
+	  PMIC board with WM8835x fitted.
+
+config SND_SOC_MX27VIS_AIC32X4
+	tristate "SoC audio support for Visstrim M10 boards"
+	depends on MACH_IMX27_VISSTRIM_M10 && I2C
+	select SND_SOC_TLV320AIC32X4
+	select SND_SOC_IMX_PCM_DMA
+	select SND_SOC_IMX_AUDMUX
+	select SND_SOC_IMX_SSI
+	help
+	  Say Y if you want to add support for SoC audio on Visstrim SM10
+	  board with TLV320AIC32X4 codec.
+
+config SND_SOC_PHYCORE_AC97
+	tristate "SoC Audio support for Phytec phyCORE (and phyCARD) boards"
+	depends on MACH_PCM043 || MACH_PCA100
+	select SND_SOC_AC97_BUS
+	select SND_SOC_WM9712
+	select SND_SOC_IMX_PCM_FIQ
+	select SND_SOC_IMX_AUDMUX
+	select SND_SOC_IMX_SSI
+	help
+	  Say Y if you want to add support for SoC audio on Phytec phyCORE
+	  and phyCARD boards in AC97 mode
+
+config SND_SOC_EUKREA_TLV320
+	tristate "Eukrea TLV320"
+	depends on MACH_EUKREA_MBIMX27_BASEBOARD \
+		|| MACH_EUKREA_MBIMXSD25_BASEBOARD \
+		|| MACH_EUKREA_MBIMXSD35_BASEBOARD \
+		|| MACH_EUKREA_MBIMXSD51_BASEBOARD
+	depends on I2C
+	select SND_SOC_TLV320AIC23
+	select SND_SOC_IMX_PCM_FIQ
+	select SND_SOC_IMX_AUDMUX
+	select SND_SOC_IMX_SSI
+	help
+	  Enable I2S based access to the TLV320AIC23B codec attached
+	  to the SSI interface
+
+config SND_SOC_IMX_SGTL5000
+	tristate "SoC Audio support for i.MX boards with sgtl5000"
+	depends on OF && I2C
+	select SND_SOC_SGTL5000
+	select SND_SOC_IMX_PCM_DMA
+	select SND_SOC_IMX_AUDMUX
+	select SND_SOC_FSL_SSI
+	select SND_SOC_FSL_UTILS
+	help
+	  Say Y if you want to add support for SoC audio on an i.MX board with
+	  a sgtl5000 codec.
+
+config SND_SOC_IMX_MC13783
+	tristate "SoC Audio support for I.MX boards with mc13783"
+	depends on MFD_MC13783
+	select SND_SOC_IMX_SSI
+	select SND_SOC_IMX_AUDMUX
+	select SND_SOC_MC13783
+	select SND_SOC_IMX_PCM_DMA
+
+endif # SND_IMX_SOC
diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile
index b4a38c0ac58c..5f3cf3f52ea0 100644
--- a/sound/soc/fsl/Makefile
+++ b/sound/soc/fsl/Makefile
@@ -8,8 +8,11 @@ obj-$(CONFIG_SND_SOC_P1022_DS) += snd-soc-p1022-ds.o
 
 # Freescale PowerPC SSI/DMA Platform Support
 snd-soc-fsl-ssi-objs := fsl_ssi.o
+snd-soc-fsl-utils-objs := fsl_utils.o
 snd-soc-fsl-dma-objs := fsl_dma.o
-obj-$(CONFIG_SND_SOC_POWERPC_SSI) += snd-soc-fsl-ssi.o snd-soc-fsl-dma.o
+obj-$(CONFIG_SND_SOC_FSL_SSI) += snd-soc-fsl-ssi.o
+obj-$(CONFIG_SND_SOC_FSL_UTILS) += snd-soc-fsl-utils.o
+obj-$(CONFIG_SND_SOC_POWERPC_DMA) += snd-soc-fsl-dma.o
 
 # MPC5200 Platform Support
 obj-$(CONFIG_SND_MPC52xx_DMA) += mpc5200_dma.o
@@ -20,3 +23,29 @@ obj-$(CONFIG_SND_SOC_MPC5200_AC97) += mpc5200_psc_ac97.o
 obj-$(CONFIG_SND_MPC52xx_SOC_PCM030) += pcm030-audio-fabric.o
 obj-$(CONFIG_SND_MPC52xx_SOC_EFIKA) += efika-audio-fabric.o
 
+# i.MX Platform Support
+snd-soc-imx-ssi-objs := imx-ssi.o
+snd-soc-imx-audmux-objs := imx-audmux.o
+
+obj-$(CONFIG_SND_SOC_IMX_SSI) += snd-soc-imx-ssi.o
+obj-$(CONFIG_SND_SOC_IMX_AUDMUX) += snd-soc-imx-audmux.o
+
+obj-$(CONFIG_SND_SOC_IMX_PCM) += snd-soc-imx-pcm.o
+snd-soc-imx-pcm-y := imx-pcm.o
+snd-soc-imx-pcm-$(CONFIG_SND_SOC_IMX_PCM_FIQ) += imx-pcm-fiq.o
+snd-soc-imx-pcm-$(CONFIG_SND_SOC_IMX_PCM_DMA) += imx-pcm-dma.o
+
+# i.MX Machine Support
+snd-soc-eukrea-tlv320-objs := eukrea-tlv320.o
+snd-soc-phycore-ac97-objs := phycore-ac97.o
+snd-soc-mx27vis-aic32x4-objs := mx27vis-aic32x4.o
+snd-soc-wm1133-ev1-objs := wm1133-ev1.o
+snd-soc-imx-sgtl5000-objs := imx-sgtl5000.o
+snd-soc-imx-mc13783-objs := imx-mc13783.o
+
+obj-$(CONFIG_SND_SOC_EUKREA_TLV320) += snd-soc-eukrea-tlv320.o
+obj-$(CONFIG_SND_SOC_PHYCORE_AC97) += snd-soc-phycore-ac97.o
+obj-$(CONFIG_SND_SOC_MX27VIS_AIC32X4) += snd-soc-mx27vis-aic32x4.o
+obj-$(CONFIG_SND_MXC_SOC_WM1133_EV1) += snd-soc-wm1133-ev1.o
+obj-$(CONFIG_SND_SOC_IMX_SGTL5000) += snd-soc-imx-sgtl5000.o
+obj-$(CONFIG_SND_SOC_IMX_MC13783) += snd-soc-imx-mc13783.o
diff --git a/sound/soc/imx/eukrea-tlv320.c b/sound/soc/fsl/eukrea-tlv320.c
similarity index 99%
rename from sound/soc/imx/eukrea-tlv320.c
rename to sound/soc/fsl/eukrea-tlv320.c
index 7d4475cfdb24..efb9ede01208 100644
--- a/sound/soc/imx/eukrea-tlv320.c
+++ b/sound/soc/fsl/eukrea-tlv320.c
@@ -7,7 +7,7 @@
  * which is Copyright 2009 Simtec Electronics
  * and on sound/soc/imx/phycore-ac97.c which is
  * Copyright 2009 Sascha Hauer, Pengutronix <s.hauer@pengutronix.de>
- * 
+ *
  *  This program is free software; you can redistribute  it and/or modify it
  *  under  the terms of  the GNU General  Public License as published by the
  *  Free Software Foundation;  either version 2 of the  License, or (at your
diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index 2eb407fa3b48..4ed2afd47782 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -11,11 +11,15 @@
  */
 
 #include <linux/init.h>
+#include <linux/io.h>
 #include <linux/module.h>
 #include <linux/interrupt.h>
+#include <linux/clk.h>
 #include <linux/device.h>
 #include <linux/delay.h>
 #include <linux/slab.h>
+#include <linux/of_address.h>
+#include <linux/of_irq.h>
 #include <linux/of_platform.h>
 
 #include <sound/core.h>
@@ -25,6 +29,26 @@
 #include <sound/soc.h>
 
 #include "fsl_ssi.h"
+#include "imx-pcm.h"
+
+#ifdef PPC
+#define read_ssi(addr)			 in_be32(addr)
+#define write_ssi(val, addr)		 out_be32(addr, val)
+#define write_ssi_mask(addr, clear, set) clrsetbits_be32(addr, clear, set)
+#elif defined ARM
+#define read_ssi(addr)			 readl(addr)
+#define write_ssi(val, addr)		 writel(val, addr)
+/*
+ * FIXME: Proper locking should be added at write_ssi_mask caller level
+ * to ensure this register read/modify/write sequence is race free.
+ */
+static inline void write_ssi_mask(u32 __iomem *addr, u32 clear, u32 set)
+{
+	u32 val = readl(addr);
+	val = (val & ~clear) | set;
+	writel(val, addr);
+}
+#endif
 
 /**
  * FSLSSI_I2S_RATES: sample rates supported by the I2S
@@ -94,6 +118,13 @@ struct fsl_ssi_private {
 	struct device_attribute dev_attr;
 	struct platform_device *pdev;
 
+	bool new_binding;
+	bool ssi_on_imx;
+	struct clk *clk;
+	struct platform_device *imx_pcm_pdev;
+	struct imx_pcm_dma_params dma_params_tx;
+	struct imx_pcm_dma_params dma_params_rx;
+
 	struct {
 		unsigned int rfrc;
 		unsigned int tfrc;
@@ -145,7 +176,7 @@ static irqreturn_t fsl_ssi_isr(int irq, void *dev_id)
 	   were interrupted for.  We mask it with the Interrupt Enable register
 	   so that we only check for events that we're interested in.
 	 */
-	sisr = in_be32(&ssi->sisr) & SIER_FLAGS;
+	sisr = read_ssi(&ssi->sisr) & SIER_FLAGS;
 
 	if (sisr & CCSR_SSI_SISR_RFRC) {
 		ssi_private->stats.rfrc++;
@@ -260,7 +291,7 @@ static irqreturn_t fsl_ssi_isr(int irq, void *dev_id)
 
 	/* Clear the bits that we set */
 	if (sisr2)
-		out_be32(&ssi->sisr, sisr2);
+		write_ssi(sisr2, &ssi->sisr);
 
 	return ret;
 }
@@ -295,7 +326,7 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream,
 		 * SSI needs to be disabled before updating the registers we set
 		 * here.
 		 */
-		clrbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN);
+		write_ssi_mask(&ssi->scr, CCSR_SSI_SCR_SSIEN, 0);
 
 		/*
 		 * Program the SSI into I2S Slave Non-Network Synchronous mode.
@@ -303,20 +334,18 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream,
 		 *
 		 * FIXME: Little-endian samples require a different shift dir
 		 */
-		clrsetbits_be32(&ssi->scr,
+		write_ssi_mask(&ssi->scr,
 			CCSR_SSI_SCR_I2S_MODE_MASK | CCSR_SSI_SCR_SYN,
 			CCSR_SSI_SCR_TFR_CLK_DIS | CCSR_SSI_SCR_I2S_MODE_SLAVE
 			| (synchronous ? CCSR_SSI_SCR_SYN : 0));
 
-		out_be32(&ssi->stcr,
-			 CCSR_SSI_STCR_TXBIT0 | CCSR_SSI_STCR_TFEN0 |
+		write_ssi(CCSR_SSI_STCR_TXBIT0 | CCSR_SSI_STCR_TFEN0 |
 			 CCSR_SSI_STCR_TFSI | CCSR_SSI_STCR_TEFS |
-			 CCSR_SSI_STCR_TSCKP);
+			 CCSR_SSI_STCR_TSCKP, &ssi->stcr);
 
-		out_be32(&ssi->srcr,
-			 CCSR_SSI_SRCR_RXBIT0 | CCSR_SSI_SRCR_RFEN0 |
+		write_ssi(CCSR_SSI_SRCR_RXBIT0 | CCSR_SSI_SRCR_RFEN0 |
 			 CCSR_SSI_SRCR_RFSI | CCSR_SSI_SRCR_REFS |
-			 CCSR_SSI_SRCR_RSCKP);
+			 CCSR_SSI_SRCR_RSCKP, &ssi->srcr);
 
 		/*
 		 * The DC and PM bits are only used if the SSI is the clock
@@ -324,7 +353,7 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream,
 		 */
 
 		/* Enable the interrupts and DMA requests */
-		out_be32(&ssi->sier, SIER_FLAGS);
+		write_ssi(SIER_FLAGS, &ssi->sier);
 
 		/*
 		 * Set the watermark for transmit FIFI 0 and receive FIFO 0. We
@@ -339,9 +368,9 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream,
 		 * make this value larger (and maybe we should), but this way
 		 * data will be written to memory as soon as it's available.
 		 */
-		out_be32(&ssi->sfcsr,
-			CCSR_SSI_SFCSR_TFWM0(ssi_private->fifo_depth - 2) |
-			CCSR_SSI_SFCSR_RFWM0(ssi_private->fifo_depth - 2));
+		write_ssi(CCSR_SSI_SFCSR_TFWM0(ssi_private->fifo_depth - 2) |
+			CCSR_SSI_SFCSR_RFWM0(ssi_private->fifo_depth - 2),
+			&ssi->sfcsr);
 
 		/*
 		 * We keep the SSI disabled because if we enable it, then the
@@ -393,6 +422,12 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream,
 		ssi_private->second_stream = substream;
 	}
 
+	if (ssi_private->ssi_on_imx)
+		snd_soc_dai_set_dma_data(dai, substream,
+			(substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ?
+				&ssi_private->dma_params_tx :
+				&ssi_private->dma_params_rx);
+
 	return 0;
 }
 
@@ -417,7 +452,7 @@ static int fsl_ssi_hw_params(struct snd_pcm_substream *substream,
 	unsigned int sample_size =
 		snd_pcm_format_width(params_format(hw_params));
 	u32 wl = CCSR_SSI_SxCCR_WL(sample_size);
-	int enabled = in_be32(&ssi->scr) & CCSR_SSI_SCR_SSIEN;
+	int enabled = read_ssi(&ssi->scr) & CCSR_SSI_SCR_SSIEN;
 
 	/*
 	 * If we're in synchronous mode, and the SSI is already enabled,
@@ -439,9 +474,9 @@ static int fsl_ssi_hw_params(struct snd_pcm_substream *substream,
 	/* In synchronous mode, the SSI uses STCCR for capture */
 	if ((substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ||
 	    ssi_private->cpu_dai_drv.symmetric_rates)
-		clrsetbits_be32(&ssi->stccr, CCSR_SSI_SxCCR_WL_MASK, wl);
+		write_ssi_mask(&ssi->stccr, CCSR_SSI_SxCCR_WL_MASK, wl);
 	else
-		clrsetbits_be32(&ssi->srccr, CCSR_SSI_SxCCR_WL_MASK, wl);
+		write_ssi_mask(&ssi->srccr, CCSR_SSI_SxCCR_WL_MASK, wl);
 
 	return 0;
 }
@@ -466,19 +501,19 @@ static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd,
 	case SNDRV_PCM_TRIGGER_START:
 	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
 		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
-			setbits32(&ssi->scr,
+			write_ssi_mask(&ssi->scr, 0,
 				CCSR_SSI_SCR_SSIEN | CCSR_SSI_SCR_TE);
 		else
-			setbits32(&ssi->scr,
+			write_ssi_mask(&ssi->scr, 0,
 				CCSR_SSI_SCR_SSIEN | CCSR_SSI_SCR_RE);
 		break;
 
 	case SNDRV_PCM_TRIGGER_STOP:
 	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
 		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
-			clrbits32(&ssi->scr, CCSR_SSI_SCR_TE);
+			write_ssi_mask(&ssi->scr, CCSR_SSI_SCR_TE, 0);
 		else
-			clrbits32(&ssi->scr, CCSR_SSI_SCR_RE);
+			write_ssi_mask(&ssi->scr, CCSR_SSI_SCR_RE, 0);
 		break;
 
 	default:
@@ -510,7 +545,7 @@ static void fsl_ssi_shutdown(struct snd_pcm_substream *substream,
 	if (!ssi_private->first_stream) {
 		struct ccsr_ssi __iomem *ssi = ssi_private->ssi;
 
-		clrbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN);
+		write_ssi_mask(&ssi->scr, CCSR_SSI_SCR_SSIEN, 0);
 	}
 }
 
@@ -622,12 +657,6 @@ static int __devinit fsl_ssi_probe(struct platform_device *pdev)
 	if (!of_device_is_available(np))
 		return -ENODEV;
 
-	/* Check for a codec-handle property. */
-	if (!of_get_property(np, "codec-handle", NULL)) {
-		dev_err(&pdev->dev, "missing codec-handle property\n");
-		return -ENODEV;
-	}
-
 	/* We only support the SSI in "I2S Slave" mode */
 	sprop = of_get_property(np, "fsl,mode", NULL);
 	if (!sprop || strcmp(sprop, "i2s-slave")) {
@@ -692,6 +721,50 @@ static int __devinit fsl_ssi_probe(struct platform_device *pdev)
                 /* Older 8610 DTs didn't have the fifo-depth property */
 		ssi_private->fifo_depth = 8;
 
+	if (of_device_is_compatible(pdev->dev.of_node, "fsl,imx21-ssi")) {
+		u32 dma_events[2];
+		ssi_private->ssi_on_imx = true;
+
+		ssi_private->clk = clk_get(&pdev->dev, NULL);
+		if (IS_ERR(ssi_private->clk)) {
+			ret = PTR_ERR(ssi_private->clk);
+			dev_err(&pdev->dev, "could not get clock: %d\n", ret);
+			goto error_irq;
+		}
+		clk_prepare_enable(ssi_private->clk);
+
+		/*
+		 * We have burstsize be "fifo_depth - 2" to match the SSI
+		 * watermark setting in fsl_ssi_startup().
+		 */
+		ssi_private->dma_params_tx.burstsize =
+			ssi_private->fifo_depth - 2;
+		ssi_private->dma_params_rx.burstsize =
+			ssi_private->fifo_depth - 2;
+		ssi_private->dma_params_tx.dma_addr =
+			ssi_private->ssi_phys + offsetof(struct ccsr_ssi, stx0);
+		ssi_private->dma_params_rx.dma_addr =
+			ssi_private->ssi_phys + offsetof(struct ccsr_ssi, srx0);
+		/*
+		 * TODO: This is a temporary solution and should be changed
+		 * to use generic DMA binding later when the helplers get in.
+		 */
+		ret = of_property_read_u32_array(pdev->dev.of_node,
+					"fsl,ssi-dma-events", dma_events, 2);
+		if (ret) {
+			dev_err(&pdev->dev, "could not get dma events\n");
+			goto error_clk;
+		}
+		ssi_private->dma_params_tx.dma = dma_events[0];
+		ssi_private->dma_params_rx.dma = dma_events[1];
+
+		ssi_private->dma_params_tx.shared_peripheral =
+				of_device_is_compatible(of_get_parent(np),
+							"fsl,spba-bus");
+		ssi_private->dma_params_rx.shared_peripheral =
+				ssi_private->dma_params_tx.shared_peripheral;
+	}
+
 	/* Initialize the the device_attribute structure */
 	dev_attr = &ssi_private->dev_attr;
 	sysfs_attr_init(&dev_attr->attr);
@@ -715,6 +788,26 @@ static int __devinit fsl_ssi_probe(struct platform_device *pdev)
 		goto error_dev;
 	}
 
+	if (ssi_private->ssi_on_imx) {
+		ssi_private->imx_pcm_pdev =
+			platform_device_register_simple("imx-pcm-audio",
+							-1, NULL, 0);
+		if (IS_ERR(ssi_private->imx_pcm_pdev)) {
+			ret = PTR_ERR(ssi_private->imx_pcm_pdev);
+			goto error_dev;
+		}
+	}
+
+	/*
+	 * If codec-handle property is missing from SSI node, we assume
+	 * that the machine driver uses new binding which does not require
+	 * SSI driver to trigger machine driver's probe.
+	 */
+	if (!of_get_property(np, "codec-handle", NULL)) {
+		ssi_private->new_binding = true;
+		goto done;
+	}
+
 	/* Trigger the machine driver's probe function.  The platform driver
 	 * name of the machine driver is taken from /compatible property of the
 	 * device tree.  We also pass the address of the CPU DAI driver
@@ -736,15 +829,24 @@ static int __devinit fsl_ssi_probe(struct platform_device *pdev)
 		goto error_dai;
 	}
 
+done:
 	return 0;
 
 error_dai:
+	if (ssi_private->ssi_on_imx)
+		platform_device_unregister(ssi_private->imx_pcm_pdev);
 	snd_soc_unregister_dai(&pdev->dev);
 
 error_dev:
 	dev_set_drvdata(&pdev->dev, NULL);
 	device_remove_file(&pdev->dev, dev_attr);
 
+error_clk:
+	if (ssi_private->ssi_on_imx) {
+		clk_disable_unprepare(ssi_private->clk);
+		clk_put(ssi_private->clk);
+	}
+
 error_irq:
 	free_irq(ssi_private->irq, ssi_private);
 
@@ -764,7 +866,13 @@ static int fsl_ssi_remove(struct platform_device *pdev)
 {
 	struct fsl_ssi_private *ssi_private = dev_get_drvdata(&pdev->dev);
 
-	platform_device_unregister(ssi_private->pdev);
+	if (!ssi_private->new_binding)
+		platform_device_unregister(ssi_private->pdev);
+	if (ssi_private->ssi_on_imx) {
+		platform_device_unregister(ssi_private->imx_pcm_pdev);
+		clk_disable_unprepare(ssi_private->clk);
+		clk_put(ssi_private->clk);
+	}
 	snd_soc_unregister_dai(&pdev->dev);
 	device_remove_file(&pdev->dev, &ssi_private->dev_attr);
 
@@ -779,6 +887,7 @@ static int fsl_ssi_remove(struct platform_device *pdev)
 
 static const struct of_device_id fsl_ssi_ids[] = {
 	{ .compatible = "fsl,mpc8610-ssi", },
+	{ .compatible = "fsl,imx21-ssi", },
 	{}
 };
 MODULE_DEVICE_TABLE(of, fsl_ssi_ids);
diff --git a/sound/soc/fsl/fsl_utils.c b/sound/soc/fsl/fsl_utils.c
new file mode 100644
index 000000000000..b9e42b503a37
--- /dev/null
+++ b/sound/soc/fsl/fsl_utils.c
@@ -0,0 +1,91 @@
+/**
+ * Freescale ALSA SoC Machine driver utility
+ *
+ * Author: Timur Tabi <timur@freescale.com>
+ *
+ * Copyright 2010 Freescale Semiconductor, Inc.
+ *
+ * This file is licensed under the terms of the GNU General Public License
+ * version 2.  This program is licensed "as is" without any warranty of any
+ * kind, whether express or implied.
+ */
+
+#include <linux/module.h>
+#include <linux/of_address.h>
+#include <sound/soc.h>
+
+#include "fsl_utils.h"
+
+/**
+ * fsl_asoc_get_dma_channel - determine the dma channel for a SSI node
+ *
+ * @ssi_np: pointer to the SSI device tree node
+ * @name: name of the phandle pointing to the dma channel
+ * @dai: ASoC DAI link pointer to be filled with platform_name
+ * @dma_channel_id: dma channel id to be returned
+ * @dma_id: dma id to be returned
+ *
+ * This function determines the dma and channel id for given SSI node.  It
+ * also discovers the platform_name for the ASoC DAI link.
+ */
+int fsl_asoc_get_dma_channel(struct device_node *ssi_np,
+			     const char *name,
+			     struct snd_soc_dai_link *dai,
+			     unsigned int *dma_channel_id,
+			     unsigned int *dma_id)
+{
+	struct resource res;
+	struct device_node *dma_channel_np, *dma_np;
+	const u32 *iprop;
+	int ret;
+
+	dma_channel_np = of_parse_phandle(ssi_np, name, 0);
+	if (!dma_channel_np)
+		return -EINVAL;
+
+	if (!of_device_is_compatible(dma_channel_np, "fsl,ssi-dma-channel")) {
+		of_node_put(dma_channel_np);
+		return -EINVAL;
+	}
+
+	/* Determine the dev_name for the device_node.  This code mimics the
+	 * behavior of of_device_make_bus_id(). We need this because ASoC uses
+	 * the dev_name() of the device to match the platform (DMA) device with
+	 * the CPU (SSI) device.  It's all ugly and hackish, but it works (for
+	 * now).
+	 *
+	 * dai->platform name should already point to an allocated buffer.
+	 */
+	ret = of_address_to_resource(dma_channel_np, 0, &res);
+	if (ret) {
+		of_node_put(dma_channel_np);
+		return ret;
+	}
+	snprintf((char *)dai->platform_name, DAI_NAME_SIZE, "%llx.%s",
+		 (unsigned long long) res.start, dma_channel_np->name);
+
+	iprop = of_get_property(dma_channel_np, "cell-index", NULL);
+	if (!iprop) {
+		of_node_put(dma_channel_np);
+		return -EINVAL;
+	}
+	*dma_channel_id = be32_to_cpup(iprop);
+
+	dma_np = of_get_parent(dma_channel_np);
+	iprop = of_get_property(dma_np, "cell-index", NULL);
+	if (!iprop) {
+		of_node_put(dma_np);
+		return -EINVAL;
+	}
+	*dma_id = be32_to_cpup(iprop);
+
+	of_node_put(dma_np);
+	of_node_put(dma_channel_np);
+
+	return 0;
+}
+EXPORT_SYMBOL(fsl_asoc_get_dma_channel);
+
+MODULE_AUTHOR("Timur Tabi <timur@freescale.com>");
+MODULE_DESCRIPTION("Freescale ASoC utility code");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/fsl/fsl_utils.h b/sound/soc/fsl/fsl_utils.h
new file mode 100644
index 000000000000..b2951126527c
--- /dev/null
+++ b/sound/soc/fsl/fsl_utils.h
@@ -0,0 +1,26 @@
+/**
+ * Freescale ALSA SoC Machine driver utility
+ *
+ * Author: Timur Tabi <timur@freescale.com>
+ *
+ * Copyright 2010 Freescale Semiconductor, Inc.
+ *
+ * This file is licensed under the terms of the GNU General Public License
+ * version 2.  This program is licensed "as is" without any warranty of any
+ * kind, whether express or implied.
+ */
+
+#ifndef _FSL_UTILS_H
+#define _FSL_UTILS_H
+
+#define DAI_NAME_SIZE	32
+
+struct snd_soc_dai_link;
+struct device_node;
+
+int fsl_asoc_get_dma_channel(struct device_node *ssi_np, const char *name,
+			     struct snd_soc_dai_link *dai,
+			     unsigned int *dma_channel_id,
+			     unsigned int *dma_id);
+
+#endif /* _FSL_UTILS_H */
diff --git a/sound/soc/imx/imx-audmux.c b/sound/soc/fsl/imx-audmux.c
similarity index 100%
rename from sound/soc/imx/imx-audmux.c
rename to sound/soc/fsl/imx-audmux.c
diff --git a/sound/soc/imx/imx-audmux.h b/sound/soc/fsl/imx-audmux.h
similarity index 100%
rename from sound/soc/imx/imx-audmux.h
rename to sound/soc/fsl/imx-audmux.h
diff --git a/sound/soc/fsl/imx-mc13783.c b/sound/soc/fsl/imx-mc13783.c
new file mode 100644
index 000000000000..f59c34943662
--- /dev/null
+++ b/sound/soc/fsl/imx-mc13783.c
@@ -0,0 +1,156 @@
+/*
+ * imx-mc13783.c  --  SoC audio for imx based boards with mc13783 codec
+ *
+ * Copyright 2012 Philippe Retornaz, <philippe.retornaz@epfl.ch>
+ *
+ * Heavly based on phycore-mc13783:
+ * Copyright 2009 Sascha Hauer, Pengutronix <s.hauer@pengutronix.de>
+ *
+ *  This program is free software; you can redistribute  it and/or modify it
+ *  under  the terms of  the GNU General  Public License as published by the
+ *  Free Software Foundation;  either version 2 of the  License, or (at your
+ *  option) any later version.
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <asm/mach-types.h>
+
+#include "../codecs/mc13783.h"
+#include "imx-ssi.h"
+#include "imx-audmux.h"
+
+#define FMT_SSI (SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_NB_NF | \
+		SND_SOC_DAIFMT_CBM_CFM)
+
+static int imx_mc13783_hifi_hw_params(struct snd_pcm_substream *substream,
+	struct snd_pcm_hw_params *params)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	int ret;
+
+	ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xfffffffc, 0xfffffffc,
+					4, 16);
+	if (ret)
+		return ret;
+
+	ret = snd_soc_dai_set_sysclk(codec_dai, MC13783_CLK_CLIA, 26000000, 0);
+	if (ret)
+		return ret;
+
+	ret = snd_soc_dai_set_tdm_slot(cpu_dai, 0x0, 0xfffffffc, 2, 16);
+	if (ret)
+		return ret;
+
+	return 0;
+}
+
+static struct snd_soc_ops imx_mc13783_hifi_ops = {
+	.hw_params = imx_mc13783_hifi_hw_params,
+};
+
+static struct snd_soc_dai_link imx_mc13783_dai_mc13783[] = {
+	{
+		.name = "MC13783",
+		.stream_name	 = "Sound",
+		.codec_dai_name	 = "mc13783-hifi",
+		.codec_name	 = "mc13783-codec",
+		.cpu_dai_name	 = "imx-ssi.0",
+		.platform_name	 = "imx-pcm-audio.0",
+		.ops		 = &imx_mc13783_hifi_ops,
+		.symmetric_rates = 1,
+		.dai_fmt 	 = FMT_SSI,
+	},
+};
+
+static const struct snd_soc_dapm_widget imx_mc13783_widget[] = {
+	SND_SOC_DAPM_MIC("Mic", NULL),
+	SND_SOC_DAPM_HP("Headphone", NULL),
+	SND_SOC_DAPM_SPK("Speaker", NULL),
+};
+
+static const struct snd_soc_dapm_route imx_mc13783_routes[] = {
+	{"Speaker", NULL, "LSP"},
+	{"Headphone", NULL, "HSL"},
+	{"Headphone", NULL, "HSR"},
+
+	{"MC1LIN", NULL, "MC1 Bias"},
+	{"MC2IN", NULL, "MC2 Bias"},
+	{"MC1 Bias", NULL, "Mic"},
+	{"MC2 Bias", NULL, "Mic"},
+};
+
+static struct snd_soc_card imx_mc13783 = {
+	.name		= "imx_mc13783",
+	.dai_link	= imx_mc13783_dai_mc13783,
+	.num_links	= ARRAY_SIZE(imx_mc13783_dai_mc13783),
+	.dapm_widgets	= imx_mc13783_widget,
+	.num_dapm_widgets = ARRAY_SIZE(imx_mc13783_widget),
+	.dapm_routes	= imx_mc13783_routes,
+	.num_dapm_routes = ARRAY_SIZE(imx_mc13783_routes),
+};
+
+static int __devinit imx_mc13783_probe(struct platform_device *pdev)
+{
+	int ret;
+
+	imx_mc13783.dev = &pdev->dev;
+
+	ret = snd_soc_register_card(&imx_mc13783);
+	if (ret) {
+		dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n",
+			ret);
+		return ret;
+	}
+
+	imx_audmux_v2_configure_port(MX31_AUDMUX_PORT4_SSI_PINS_4,
+		IMX_AUDMUX_V2_PTCR_SYN,
+		IMX_AUDMUX_V2_PDCR_RXDSEL(MX31_AUDMUX_PORT1_SSI0) |
+		IMX_AUDMUX_V2_PDCR_MODE(1) |
+		IMX_AUDMUX_V2_PDCR_INMMASK(0xfc));
+	imx_audmux_v2_configure_port(MX31_AUDMUX_PORT1_SSI0,
+		IMX_AUDMUX_V2_PTCR_SYN |
+		IMX_AUDMUX_V2_PTCR_TFSDIR |
+		IMX_AUDMUX_V2_PTCR_TFSEL(MX31_AUDMUX_PORT4_SSI_PINS_4) |
+		IMX_AUDMUX_V2_PTCR_TCLKDIR |
+		IMX_AUDMUX_V2_PTCR_TCSEL(MX31_AUDMUX_PORT4_SSI_PINS_4) |
+		IMX_AUDMUX_V2_PTCR_RFSDIR |
+		IMX_AUDMUX_V2_PTCR_RFSEL(MX31_AUDMUX_PORT4_SSI_PINS_4) |
+		IMX_AUDMUX_V2_PTCR_RCLKDIR |
+		IMX_AUDMUX_V2_PTCR_RCSEL(MX31_AUDMUX_PORT4_SSI_PINS_4),
+		IMX_AUDMUX_V2_PDCR_RXDSEL(MX31_AUDMUX_PORT4_SSI_PINS_4));
+
+	return ret;
+}
+
+static int __devexit imx_mc13783_remove(struct platform_device *pdev)
+{
+	snd_soc_unregister_card(&imx_mc13783);
+
+	return 0;
+}
+
+static struct platform_driver imx_mc13783_audio_driver = {
+	.driver = {
+		.name = "imx_mc13783",
+		.owner = THIS_MODULE,
+	},
+	.probe = imx_mc13783_probe,
+	.remove = __devexit_p(imx_mc13783_remove)
+};
+
+module_platform_driver(imx_mc13783_audio_driver);
+
+MODULE_AUTHOR("Sascha Hauer <s.hauer@pengutronix.de>");
+MODULE_AUTHOR("Philippe Retornaz <philippe.retornaz@epfl.ch");
+MODULE_DESCRIPTION("imx with mc13783 codec ALSA SoC driver");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:imx_mc13783");
diff --git a/sound/soc/imx/imx-pcm-dma-mx2.c b/sound/soc/fsl/imx-pcm-dma.c
similarity index 97%
rename from sound/soc/imx/imx-pcm-dma-mx2.c
rename to sound/soc/fsl/imx-pcm-dma.c
index 6b818de2fc03..f3c0a5ef35c8 100644
--- a/sound/soc/imx/imx-pcm-dma-mx2.c
+++ b/sound/soc/fsl/imx-pcm-dma.c
@@ -109,7 +109,8 @@ static int snd_imx_open(struct snd_pcm_substream *substream)
 	dma_params = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
 
 	dma_data = kzalloc(sizeof(*dma_data), GFP_KERNEL);
-	dma_data->peripheral_type = IMX_DMATYPE_SSI;
+	dma_data->peripheral_type = dma_params->shared_peripheral ?
+					IMX_DMATYPE_SSI_SP : IMX_DMATYPE_SSI;
 	dma_data->priority = DMA_PRIO_HIGH;
 	dma_data->dma_request = dma_params->dma;
 
diff --git a/sound/soc/imx/imx-pcm-fiq.c b/sound/soc/fsl/imx-pcm-fiq.c
similarity index 100%
rename from sound/soc/imx/imx-pcm-fiq.c
rename to sound/soc/fsl/imx-pcm-fiq.c
diff --git a/sound/soc/imx/imx-pcm.c b/sound/soc/fsl/imx-pcm.c
similarity index 100%
rename from sound/soc/imx/imx-pcm.c
rename to sound/soc/fsl/imx-pcm.c
diff --git a/sound/soc/imx/imx-pcm.h b/sound/soc/fsl/imx-pcm.h
similarity index 93%
rename from sound/soc/imx/imx-pcm.h
rename to sound/soc/fsl/imx-pcm.h
index b5f5c3acf34d..83c0ed7d55c9 100644
--- a/sound/soc/imx/imx-pcm.h
+++ b/sound/soc/fsl/imx-pcm.h
@@ -22,6 +22,7 @@ struct imx_pcm_dma_params {
 	int dma;
 	unsigned long dma_addr;
 	int burstsize;
+	bool shared_peripheral;	/* The peripheral is on SPBA bus */
 };
 
 int snd_imx_pcm_mmap(struct snd_pcm_substream *substream,
diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c
new file mode 100644
index 000000000000..3a729caeb8c8
--- /dev/null
+++ b/sound/soc/fsl/imx-sgtl5000.c
@@ -0,0 +1,221 @@
+/*
+ * Copyright 2012 Freescale Semiconductor, Inc.
+ * Copyright 2012 Linaro Ltd.
+ *
+ * The code contained herein is licensed under the GNU General Public
+ * License. You may obtain a copy of the GNU General Public License
+ * Version 2 or later at the following locations:
+ *
+ * http://www.opensource.org/licenses/gpl-license.html
+ * http://www.gnu.org/copyleft/gpl.html
+ */
+
+#include <linux/module.h>
+#include <linux/of.h>
+#include <linux/of_platform.h>
+#include <linux/of_i2c.h>
+#include <linux/clk.h>
+#include <sound/soc.h>
+
+#include "../codecs/sgtl5000.h"
+#include "imx-audmux.h"
+
+#define DAI_NAME_SIZE	32
+
+struct imx_sgtl5000_data {
+	struct snd_soc_dai_link dai;
+	struct snd_soc_card card;
+	char codec_dai_name[DAI_NAME_SIZE];
+	char platform_name[DAI_NAME_SIZE];
+	struct clk *codec_clk;
+	unsigned int clk_frequency;
+};
+
+static int imx_sgtl5000_dai_init(struct snd_soc_pcm_runtime *rtd)
+{
+	struct imx_sgtl5000_data *data = container_of(rtd->card,
+					struct imx_sgtl5000_data, card);
+	struct device *dev = rtd->card->dev;
+	int ret;
+
+	ret = snd_soc_dai_set_sysclk(rtd->codec_dai, SGTL5000_SYSCLK,
+				     data->clk_frequency, SND_SOC_CLOCK_IN);
+	if (ret) {
+		dev_err(dev, "could not set codec driver clock params\n");
+		return ret;
+	}
+
+	return 0;
+}
+
+static const struct snd_soc_dapm_widget imx_sgtl5000_dapm_widgets[] = {
+	SND_SOC_DAPM_MIC("Mic Jack", NULL),
+	SND_SOC_DAPM_LINE("Line In Jack", NULL),
+	SND_SOC_DAPM_HP("Headphone Jack", NULL),
+	SND_SOC_DAPM_SPK("Line Out Jack", NULL),
+	SND_SOC_DAPM_SPK("Ext Spk", NULL),
+};
+
+static int __devinit imx_sgtl5000_probe(struct platform_device *pdev)
+{
+	struct device_node *np = pdev->dev.of_node;
+	struct device_node *ssi_np, *codec_np;
+	struct platform_device *ssi_pdev;
+	struct i2c_client *codec_dev;
+	struct imx_sgtl5000_data *data;
+	int int_port, ext_port;
+	int ret;
+
+	ret = of_property_read_u32(np, "mux-int-port", &int_port);
+	if (ret) {
+		dev_err(&pdev->dev, "mux-int-port missing or invalid\n");
+		return ret;
+	}
+	ret = of_property_read_u32(np, "mux-ext-port", &ext_port);
+	if (ret) {
+		dev_err(&pdev->dev, "mux-ext-port missing or invalid\n");
+		return ret;
+	}
+
+	/*
+	 * The port numbering in the hardware manual starts at 1, while
+	 * the audmux API expects it starts at 0.
+	 */
+	int_port--;
+	ext_port--;
+	ret = imx_audmux_v2_configure_port(int_port,
+			IMX_AUDMUX_V2_PTCR_SYN |
+			IMX_AUDMUX_V2_PTCR_TFSEL(ext_port) |
+			IMX_AUDMUX_V2_PTCR_TCSEL(ext_port) |
+			IMX_AUDMUX_V2_PTCR_TFSDIR |
+			IMX_AUDMUX_V2_PTCR_TCLKDIR,
+			IMX_AUDMUX_V2_PDCR_RXDSEL(ext_port));
+	if (ret) {
+		dev_err(&pdev->dev, "audmux internal port setup failed\n");
+		return ret;
+	}
+	imx_audmux_v2_configure_port(ext_port,
+			IMX_AUDMUX_V2_PTCR_SYN |
+			IMX_AUDMUX_V2_PTCR_TCSEL(int_port),
+			IMX_AUDMUX_V2_PDCR_RXDSEL(int_port));
+	if (ret) {
+		dev_err(&pdev->dev, "audmux external port setup failed\n");
+		return ret;
+	}
+
+	ssi_np = of_parse_phandle(pdev->dev.of_node, "ssi-controller", 0);
+	codec_np = of_parse_phandle(pdev->dev.of_node, "audio-codec", 0);
+	if (!ssi_np || !codec_np) {
+		dev_err(&pdev->dev, "phandle missing or invalid\n");
+		ret = -EINVAL;
+		goto fail;
+	}
+
+	ssi_pdev = of_find_device_by_node(ssi_np);
+	if (!ssi_pdev) {
+		dev_err(&pdev->dev, "failed to find SSI platform device\n");
+		ret = -EINVAL;
+		goto fail;
+	}
+	codec_dev = of_find_i2c_device_by_node(codec_np);
+	if (!codec_dev) {
+		dev_err(&pdev->dev, "failed to find codec platform device\n");
+		return -EINVAL;
+	}
+
+	data = devm_kzalloc(&pdev->dev, sizeof(*data), GFP_KERNEL);
+	if (!data) {
+		ret = -ENOMEM;
+		goto fail;
+	}
+
+	data->codec_clk = clk_get(&codec_dev->dev, NULL);
+	if (IS_ERR(data->codec_clk)) {
+		/* assuming clock enabled by default */
+		data->codec_clk = NULL;
+		ret = of_property_read_u32(codec_np, "clock-frequency",
+					&data->clk_frequency);
+		if (ret) {
+			dev_err(&codec_dev->dev,
+				"clock-frequency missing or invalid\n");
+			goto fail;
+		}
+	} else {
+		data->clk_frequency = clk_get_rate(data->codec_clk);
+		clk_prepare_enable(data->codec_clk);
+	}
+
+	data->dai.name = "HiFi";
+	data->dai.stream_name = "HiFi";
+	data->dai.codec_dai_name = "sgtl5000";
+	data->dai.codec_of_node = codec_np;
+	data->dai.cpu_dai_name = dev_name(&ssi_pdev->dev);
+	data->dai.platform_name = "imx-pcm-audio";
+	data->dai.init = &imx_sgtl5000_dai_init;
+	data->dai.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+			    SND_SOC_DAIFMT_CBM_CFM;
+
+	data->card.dev = &pdev->dev;
+	ret = snd_soc_of_parse_card_name(&data->card, "model");
+	if (ret)
+		goto clk_fail;
+	ret = snd_soc_of_parse_audio_routing(&data->card, "audio-routing");
+	if (ret)
+		goto clk_fail;
+	data->card.num_links = 1;
+	data->card.dai_link = &data->dai;
+	data->card.dapm_widgets = imx_sgtl5000_dapm_widgets;
+	data->card.num_dapm_widgets = ARRAY_SIZE(imx_sgtl5000_dapm_widgets);
+
+	ret = snd_soc_register_card(&data->card);
+	if (ret) {
+		dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret);
+		goto clk_fail;
+	}
+
+	platform_set_drvdata(pdev, data);
+clk_fail:
+	clk_put(data->codec_clk);
+fail:
+	if (ssi_np)
+		of_node_put(ssi_np);
+	if (codec_np)
+		of_node_put(codec_np);
+
+	return ret;
+}
+
+static int __devexit imx_sgtl5000_remove(struct platform_device *pdev)
+{
+	struct imx_sgtl5000_data *data = platform_get_drvdata(pdev);
+
+	if (data->codec_clk) {
+		clk_disable_unprepare(data->codec_clk);
+		clk_put(data->codec_clk);
+	}
+	snd_soc_unregister_card(&data->card);
+
+	return 0;
+}
+
+static const struct of_device_id imx_sgtl5000_dt_ids[] = {
+	{ .compatible = "fsl,imx-audio-sgtl5000", },
+	{ /* sentinel */ }
+};
+MODULE_DEVICE_TABLE(of, imx_sgtl5000_dt_ids);
+
+static struct platform_driver imx_sgtl5000_driver = {
+	.driver = {
+		.name = "imx-sgtl5000",
+		.owner = THIS_MODULE,
+		.of_match_table = imx_sgtl5000_dt_ids,
+	},
+	.probe = imx_sgtl5000_probe,
+	.remove = __devexit_p(imx_sgtl5000_remove),
+};
+module_platform_driver(imx_sgtl5000_driver);
+
+MODULE_AUTHOR("Shawn Guo <shawn.guo@linaro.org>");
+MODULE_DESCRIPTION("Freescale i.MX SGTL5000 ASoC machine driver");
+MODULE_LICENSE("GPL v2");
+MODULE_ALIAS("platform:imx-sgtl5000");
diff --git a/sound/soc/imx/imx-ssi.c b/sound/soc/fsl/imx-ssi.c
similarity index 99%
rename from sound/soc/imx/imx-ssi.c
rename to sound/soc/fsl/imx-ssi.c
index 4f81ed456325..cf3ed0362c9c 100644
--- a/sound/soc/imx/imx-ssi.c
+++ b/sound/soc/fsl/imx-ssi.c
@@ -28,7 +28,7 @@
  * value. When we read the same register two times (and the register still
  * contains the same value) these status bits are not set. We work
  * around this by not polling these bits but only wait a fixed delay.
- * 
+ *
  */
 
 #include <linux/clk.h>
diff --git a/sound/soc/imx/imx-ssi.h b/sound/soc/fsl/imx-ssi.h
similarity index 100%
rename from sound/soc/imx/imx-ssi.h
rename to sound/soc/fsl/imx-ssi.h
diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c
index 3fea5a15ffe8..60bcba1bc30e 100644
--- a/sound/soc/fsl/mpc8610_hpcd.c
+++ b/sound/soc/fsl/mpc8610_hpcd.c
@@ -14,18 +14,16 @@
 #include <linux/interrupt.h>
 #include <linux/of_device.h>
 #include <linux/slab.h>
-#include <linux/of_i2c.h>
 #include <sound/soc.h>
 #include <asm/fsl_guts.h>
 
 #include "fsl_dma.h"
 #include "fsl_ssi.h"
+#include "fsl_utils.h"
 
 /* There's only one global utilities register */
 static phys_addr_t guts_phys;
 
-#define DAI_NAME_SIZE	32
-
 /**
  * mpc8610_hpcd_data: machine-specific ASoC device data
  *
@@ -43,7 +41,6 @@ struct mpc8610_hpcd_data {
 	unsigned int dma_id[2];		/* 0 = DMA1, 1 = DMA2, etc */
 	unsigned int dma_channel_id[2]; /* 0 = ch 0, 1 = ch 1, etc*/
 	char codec_dai_name[DAI_NAME_SIZE];
-	char codec_name[DAI_NAME_SIZE];
 	char platform_name[2][DAI_NAME_SIZE]; /* One for each DMA channel */
 };
 
@@ -180,141 +177,6 @@ static struct snd_soc_ops mpc8610_hpcd_ops = {
 	.startup = mpc8610_hpcd_startup,
 };
 
-/**
- * get_node_by_phandle_name - get a node by its phandle name
- *
- * This function takes a node, the name of a property in that node, and a
- * compatible string.  Assuming the property is a phandle to another node,
- * it returns that node, (optionally) if that node is compatible.
- *
- * If the property is not a phandle, or the node it points to is not compatible
- * with the specific string, then NULL is returned.
- */
-static struct device_node *get_node_by_phandle_name(struct device_node *np,
-					       const char *name,
-					       const char *compatible)
-{
-	const phandle *ph;
-	int len;
-
-	ph = of_get_property(np, name, &len);
-	if (!ph || (len != sizeof(phandle)))
-		return NULL;
-
-	np = of_find_node_by_phandle(*ph);
-	if (!np)
-		return NULL;
-
-	if (compatible && !of_device_is_compatible(np, compatible)) {
-		of_node_put(np);
-		return NULL;
-	}
-
-	return np;
-}
-
-/**
- * get_parent_cell_index -- return the cell-index of the parent of a node
- *
- * Return the value of the cell-index property of the parent of the given
- * node.  This is used for DMA channel nodes that need to know the DMA ID
- * of the controller they are on.
- */
-static int get_parent_cell_index(struct device_node *np)
-{
-	struct device_node *parent = of_get_parent(np);
-	const u32 *iprop;
-
-	if (!parent)
-		return -1;
-
-	iprop = of_get_property(parent, "cell-index", NULL);
-	of_node_put(parent);
-
-	if (!iprop)
-		return -1;
-
-	return be32_to_cpup(iprop);
-}
-
-/**
- * codec_node_dev_name - determine the dev_name for a codec node
- *
- * This function determines the dev_name for an I2C node.  This is the name
- * that would be returned by dev_name() if this device_node were part of a
- * 'struct device'  It's ugly and hackish, but it works.
- *
- * The dev_name for such devices include the bus number and I2C address. For
- * example, "cs4270.0-004f".
- */
-static int codec_node_dev_name(struct device_node *np, char *buf, size_t len)
-{
-	const u32 *iprop;
-	int addr;
-	char temp[DAI_NAME_SIZE];
-	struct i2c_client *i2c;
-
-	of_modalias_node(np, temp, DAI_NAME_SIZE);
-
-	iprop = of_get_property(np, "reg", NULL);
-	if (!iprop)
-		return -EINVAL;
-
-	addr = be32_to_cpup(iprop);
-
-	/* We need the adapter number */
-	i2c = of_find_i2c_device_by_node(np);
-	if (!i2c)
-		return -ENODEV;
-
-	snprintf(buf, len, "%s.%u-%04x", temp, i2c->adapter->nr, addr);
-
-	return 0;
-}
-
-static int get_dma_channel(struct device_node *ssi_np,
-			   const char *name,
-			   struct snd_soc_dai_link *dai,
-			   unsigned int *dma_channel_id,
-			   unsigned int *dma_id)
-{
-	struct resource res;
-	struct device_node *dma_channel_np;
-	const u32 *iprop;
-	int ret;
-
-	dma_channel_np = get_node_by_phandle_name(ssi_np, name,
-						  "fsl,ssi-dma-channel");
-	if (!dma_channel_np)
-		return -EINVAL;
-
-	/* Determine the dev_name for the device_node.  This code mimics the
-	 * behavior of of_device_make_bus_id(). We need this because ASoC uses
-	 * the dev_name() of the device to match the platform (DMA) device with
-	 * the CPU (SSI) device.  It's all ugly and hackish, but it works (for
-	 * now).
-	 *
-	 * dai->platform name should already point to an allocated buffer.
-	 */
-	ret = of_address_to_resource(dma_channel_np, 0, &res);
-	if (ret)
-		return ret;
-	snprintf((char *)dai->platform_name, DAI_NAME_SIZE, "%llx.%s",
-		 (unsigned long long) res.start, dma_channel_np->name);
-
-	iprop = of_get_property(dma_channel_np, "cell-index", NULL);
-	if (!iprop) {
-		of_node_put(dma_channel_np);
-		return -EINVAL;
-	}
-
-	*dma_channel_id = be32_to_cpup(iprop);
-	*dma_id = get_parent_cell_index(dma_channel_np);
-	of_node_put(dma_channel_np);
-
-	return 0;
-}
-
 /**
  * mpc8610_hpcd_probe: platform probe function for the machine driver
  *
@@ -352,16 +214,8 @@ static int mpc8610_hpcd_probe(struct platform_device *pdev)
 	machine_data->dai[0].cpu_dai_name = dev_name(&ssi_pdev->dev);
 	machine_data->dai[0].ops = &mpc8610_hpcd_ops;
 
-	/* Determine the codec name, it will be used as the codec DAI name */
-	ret = codec_node_dev_name(codec_np, machine_data->codec_name,
-				  DAI_NAME_SIZE);
-	if (ret) {
-		dev_err(&pdev->dev, "invalid codec node %s\n",
-			codec_np->full_name);
-		ret = -EINVAL;
-		goto error;
-	}
-	machine_data->dai[0].codec_name = machine_data->codec_name;
+	/* ASoC core can match codec with device node */
+	machine_data->dai[0].codec_of_node = codec_np;
 
 	/* The DAI name from the codec (snd_soc_dai_driver.name) */
 	machine_data->dai[0].codec_dai_name = "cs4270-hifi";
@@ -458,9 +312,10 @@ static int mpc8610_hpcd_probe(struct platform_device *pdev)
 
 	/* Find the playback DMA channel to use. */
 	machine_data->dai[0].platform_name = machine_data->platform_name[0];
-	ret = get_dma_channel(np, "fsl,playback-dma", &machine_data->dai[0],
-			      &machine_data->dma_channel_id[0],
-			      &machine_data->dma_id[0]);
+	ret = fsl_asoc_get_dma_channel(np, "fsl,playback-dma",
+				       &machine_data->dai[0],
+				       &machine_data->dma_channel_id[0],
+				       &machine_data->dma_id[0]);
 	if (ret) {
 		dev_err(&pdev->dev, "missing/invalid playback DMA phandle\n");
 		goto error;
@@ -468,9 +323,10 @@ static int mpc8610_hpcd_probe(struct platform_device *pdev)
 
 	/* Find the capture DMA channel to use. */
 	machine_data->dai[1].platform_name = machine_data->platform_name[1];
-	ret = get_dma_channel(np, "fsl,capture-dma", &machine_data->dai[1],
-			      &machine_data->dma_channel_id[1],
-			      &machine_data->dma_id[1]);
+	ret = fsl_asoc_get_dma_channel(np, "fsl,capture-dma",
+				       &machine_data->dai[1],
+				       &machine_data->dma_channel_id[1],
+				       &machine_data->dma_id[1]);
 	if (ret) {
 		dev_err(&pdev->dev, "missing/invalid capture DMA phandle\n");
 		goto error;
diff --git a/sound/soc/imx/mx27vis-aic32x4.c b/sound/soc/fsl/mx27vis-aic32x4.c
similarity index 100%
rename from sound/soc/imx/mx27vis-aic32x4.c
rename to sound/soc/fsl/mx27vis-aic32x4.c
diff --git a/sound/soc/fsl/p1022_ds.c b/sound/soc/fsl/p1022_ds.c
index 982a1c944983..50adf4032bcc 100644
--- a/sound/soc/fsl/p1022_ds.c
+++ b/sound/soc/fsl/p1022_ds.c
@@ -14,12 +14,12 @@
 #include <linux/interrupt.h>
 #include <linux/of_device.h>
 #include <linux/slab.h>
-#include <linux/of_i2c.h>
 #include <sound/soc.h>
 #include <asm/fsl_guts.h>
 
 #include "fsl_dma.h"
 #include "fsl_ssi.h"
+#include "fsl_utils.h"
 
 /* P1022-specific PMUXCR and DMUXCR bit definitions */
 
@@ -57,8 +57,6 @@ static inline void guts_set_dmuxcr(struct ccsr_guts __iomem *guts,
 /* There's only one global utilities register */
 static phys_addr_t guts_phys;
 
-#define DAI_NAME_SIZE	32
-
 /**
  * machine_data: machine-specific ASoC device data
  *
@@ -75,7 +73,6 @@ struct machine_data {
 	unsigned int ssi_id;		/* 0 = SSI1, 1 = SSI2, etc */
 	unsigned int dma_id[2];		/* 0 = DMA1, 1 = DMA2, etc */
 	unsigned int dma_channel_id[2]; /* 0 = ch 0, 1 = ch 1, etc*/
-	char codec_name[DAI_NAME_SIZE];
 	char platform_name[2][DAI_NAME_SIZE]; /* One for each DMA channel */
 };
 
@@ -190,136 +187,6 @@ static struct snd_soc_ops p1022_ds_ops = {
 	.startup = p1022_ds_startup,
 };
 
-/**
- * get_node_by_phandle_name - get a node by its phandle name
- *
- * This function takes a node, the name of a property in that node, and a
- * compatible string.  Assuming the property is a phandle to another node,
- * it returns that node, (optionally) if that node is compatible.
- *
- * If the property is not a phandle, or the node it points to is not compatible
- * with the specific string, then NULL is returned.
- */
-static struct device_node *get_node_by_phandle_name(struct device_node *np,
-	const char *name, const char *compatible)
-{
-	np = of_parse_phandle(np, name, 0);
-	if (!np)
-		return NULL;
-
-	if (!of_device_is_compatible(np, compatible)) {
-		of_node_put(np);
-		return NULL;
-	}
-
-	return np;
-}
-
-/**
- * get_parent_cell_index -- return the cell-index of the parent of a node
- *
- * Return the value of the cell-index property of the parent of the given
- * node.  This is used for DMA channel nodes that need to know the DMA ID
- * of the controller they are on.
- */
-static int get_parent_cell_index(struct device_node *np)
-{
-	struct device_node *parent = of_get_parent(np);
-	const u32 *iprop;
-	int ret = -1;
-
-	if (!parent)
-		return -1;
-
-	iprop = of_get_property(parent, "cell-index", NULL);
-	if (iprop)
-		ret = be32_to_cpup(iprop);
-
-	of_node_put(parent);
-
-	return ret;
-}
-
-/**
- * codec_node_dev_name - determine the dev_name for a codec node
- *
- * This function determines the dev_name for an I2C node.  This is the name
- * that would be returned by dev_name() if this device_node were part of a
- * 'struct device'  It's ugly and hackish, but it works.
- *
- * The dev_name for such devices include the bus number and I2C address. For
- * example, "cs4270-codec.0-004f".
- */
-static int codec_node_dev_name(struct device_node *np, char *buf, size_t len)
-{
-	const u32 *iprop;
-	int addr;
-	char temp[DAI_NAME_SIZE];
-	struct i2c_client *i2c;
-
-	of_modalias_node(np, temp, DAI_NAME_SIZE);
-
-	iprop = of_get_property(np, "reg", NULL);
-	if (!iprop)
-		return -EINVAL;
-
-	addr = be32_to_cpup(iprop);
-
-	/* We need the adapter number */
-	i2c = of_find_i2c_device_by_node(np);
-	if (!i2c)
-		return -ENODEV;
-
-	snprintf(buf, len, "%s.%u-%04x", temp, i2c->adapter->nr, addr);
-
-	return 0;
-}
-
-static int get_dma_channel(struct device_node *ssi_np,
-			   const char *name,
-			   struct snd_soc_dai_link *dai,
-			   unsigned int *dma_channel_id,
-			   unsigned int *dma_id)
-{
-	struct resource res;
-	struct device_node *dma_channel_np;
-	const u32 *iprop;
-	int ret;
-
-	dma_channel_np = get_node_by_phandle_name(ssi_np, name,
-						  "fsl,ssi-dma-channel");
-	if (!dma_channel_np)
-		return -EINVAL;
-
-	/* Determine the dev_name for the device_node.  This code mimics the
-	 * behavior of of_device_make_bus_id(). We need this because ASoC uses
-	 * the dev_name() of the device to match the platform (DMA) device with
-	 * the CPU (SSI) device.  It's all ugly and hackish, but it works (for
-	 * now).
-	 *
-	 * dai->platform name should already point to an allocated buffer.
-	 */
-	ret = of_address_to_resource(dma_channel_np, 0, &res);
-	if (ret) {
-		of_node_put(dma_channel_np);
-		return ret;
-	}
-	snprintf((char *)dai->platform_name, DAI_NAME_SIZE, "%llx.%s",
-		 (unsigned long long) res.start, dma_channel_np->name);
-
-	iprop = of_get_property(dma_channel_np, "cell-index", NULL);
-	if (!iprop) {
-		of_node_put(dma_channel_np);
-		return -EINVAL;
-	}
-
-	*dma_channel_id = be32_to_cpup(iprop);
-	*dma_id = get_parent_cell_index(dma_channel_np);
-	of_node_put(dma_channel_np);
-
-	return 0;
-}
-
 /**
  * p1022_ds_probe: platform probe function for the machine driver
  *
@@ -357,15 +224,8 @@ static int p1022_ds_probe(struct platform_device *pdev)
 	mdata->dai[0].cpu_dai_name = dev_name(&ssi_pdev->dev);
 	mdata->dai[0].ops = &p1022_ds_ops;
 
-	/* Determine the codec name, it will be used as the codec DAI name */
-	ret = codec_node_dev_name(codec_np, mdata->codec_name, DAI_NAME_SIZE);
-	if (ret) {
-		dev_err(&pdev->dev, "invalid codec node %s\n",
-			codec_np->full_name);
-		ret = -EINVAL;
-		goto error;
-	}
-	mdata->dai[0].codec_name = mdata->codec_name;
+	/* ASoC core can match codec with device node */
+	mdata->dai[0].codec_of_node = codec_np;
 
 	/* We register two DAIs per SSI, one for playback and the other for
 	 * capture.  We support codecs that have separate DAIs for both playback
@@ -462,9 +322,9 @@ static int p1022_ds_probe(struct platform_device *pdev)
 
 	/* Find the playback DMA channel to use. */
 	mdata->dai[0].platform_name = mdata->platform_name[0];
-	ret = get_dma_channel(np, "fsl,playback-dma", &mdata->dai[0],
-			      &mdata->dma_channel_id[0],
-			      &mdata->dma_id[0]);
+	ret = fsl_asoc_get_dma_channel(np, "fsl,playback-dma", &mdata->dai[0],
+				       &mdata->dma_channel_id[0],
+				       &mdata->dma_id[0]);
 	if (ret) {
 		dev_err(&pdev->dev, "missing/invalid playback DMA phandle\n");
 		goto error;
@@ -472,9 +332,9 @@ static int p1022_ds_probe(struct platform_device *pdev)
 
 	/* Find the capture DMA channel to use. */
 	mdata->dai[1].platform_name = mdata->platform_name[1];
-	ret = get_dma_channel(np, "fsl,capture-dma", &mdata->dai[1],
-			      &mdata->dma_channel_id[1],
-			      &mdata->dma_id[1]);
+	ret = fsl_asoc_get_dma_channel(np, "fsl,capture-dma", &mdata->dai[1],
+				       &mdata->dma_channel_id[1],
+				       &mdata->dma_id[1]);
 	if (ret) {
 		dev_err(&pdev->dev, "missing/invalid capture DMA phandle\n");
 		goto error;
diff --git a/sound/soc/imx/phycore-ac97.c b/sound/soc/fsl/phycore-ac97.c
similarity index 100%
rename from sound/soc/imx/phycore-ac97.c
rename to sound/soc/fsl/phycore-ac97.c
diff --git a/sound/soc/imx/wm1133-ev1.c b/sound/soc/fsl/wm1133-ev1.c
similarity index 100%
rename from sound/soc/imx/wm1133-ev1.c
rename to sound/soc/fsl/wm1133-ev1.c
diff --git a/sound/soc/generic/Kconfig b/sound/soc/generic/Kconfig
new file mode 100644
index 000000000000..610f61251640
--- /dev/null
+++ b/sound/soc/generic/Kconfig
@@ -0,0 +1,4 @@
+config SND_SIMPLE_CARD
+	tristate "ASoC Simple sound card support"
+	help
+	  This option enables generic simple sound card support
diff --git a/sound/soc/generic/Makefile b/sound/soc/generic/Makefile
new file mode 100644
index 000000000000..9c3b246792bf
--- /dev/null
+++ b/sound/soc/generic/Makefile
@@ -0,0 +1,3 @@
+snd-soc-simple-card-objs	:= simple-card.o
+
+obj-$(CONFIG_SND_SIMPLE_CARD)	+= snd-soc-simple-card.o
diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c
new file mode 100644
index 000000000000..b4b4cab30232
--- /dev/null
+++ b/sound/soc/generic/simple-card.c
@@ -0,0 +1,114 @@
+/*
+ * ASoC simple sound card support
+ *
+ * Copyright (C) 2012 Renesas Solutions Corp.
+ * Kuninori Morimoto <kuninori.morimoto.gx@renesas.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/platform_device.h>
+#include <linux/module.h>
+#include <sound/simple_card.h>
+
+#define asoc_simple_get_card_info(p) \
+	container_of(p->dai_link, struct asoc_simple_card_info, snd_link)
+
+static int asoc_simple_card_dai_init(struct snd_soc_pcm_runtime *rtd)
+{
+	struct asoc_simple_card_info *cinfo = asoc_simple_get_card_info(rtd);
+	struct asoc_simple_dai_init_info *iinfo = cinfo->init;
+	struct snd_soc_dai *codec = rtd->codec_dai;
+	struct snd_soc_dai *cpu = rtd->cpu_dai;
+	unsigned int cpu_daifmt = iinfo->fmt | iinfo->cpu_daifmt;
+	unsigned int codec_daifmt = iinfo->fmt | iinfo->codec_daifmt;
+	int ret;
+
+	if (codec_daifmt) {
+		ret = snd_soc_dai_set_fmt(codec, codec_daifmt);
+		if (ret < 0)
+			return ret;
+	}
+
+	if (iinfo->sysclk) {
+		ret = snd_soc_dai_set_sysclk(codec, 0, iinfo->sysclk, 0);
+		if (ret < 0)
+			return ret;
+	}
+
+	if (cpu_daifmt) {
+		ret = snd_soc_dai_set_fmt(cpu, cpu_daifmt);
+		if (ret < 0)
+			return ret;
+	}
+
+	return 0;
+}
+
+static int asoc_simple_card_probe(struct platform_device *pdev)
+{
+	struct asoc_simple_card_info *cinfo = pdev->dev.platform_data;
+
+	if (!cinfo) {
+		dev_err(&pdev->dev, "no info for asoc-simple-card\n");
+		return -EINVAL;
+	}
+
+	if (!cinfo->name	||
+	    !cinfo->card	||
+	    !cinfo->cpu_dai	||
+	    !cinfo->codec	||
+	    !cinfo->platform	||
+	    !cinfo->codec_dai) {
+		dev_err(&pdev->dev, "insufficient asoc_simple_card_info settings\n");
+		return -EINVAL;
+	}
+
+	/*
+	 * init snd_soc_dai_link
+	 */
+	cinfo->snd_link.name		= cinfo->name;
+	cinfo->snd_link.stream_name	= cinfo->name;
+	cinfo->snd_link.cpu_dai_name	= cinfo->cpu_dai;
+	cinfo->snd_link.platform_name	= cinfo->platform;
+	cinfo->snd_link.codec_name	= cinfo->codec;
+	cinfo->snd_link.codec_dai_name	= cinfo->codec_dai;
+
+	/* enable snd_link.init if cinfo has settings */
+	if (cinfo->init)
+		cinfo->snd_link.init	= asoc_simple_card_dai_init;
+
+	/*
+	 * init snd_soc_card
+	 */
+	cinfo->snd_card.name		= cinfo->card;
+	cinfo->snd_card.owner		= THIS_MODULE;
+	cinfo->snd_card.dai_link	= &cinfo->snd_link;
+	cinfo->snd_card.num_links	= 1;
+	cinfo->snd_card.dev		= &pdev->dev;
+
+	return snd_soc_register_card(&cinfo->snd_card);
+}
+
+static int asoc_simple_card_remove(struct platform_device *pdev)
+{
+	struct asoc_simple_card_info *cinfo = pdev->dev.platform_data;
+
+	return snd_soc_unregister_card(&cinfo->snd_card);
+}
+
+static struct platform_driver asoc_simple_card = {
+	.driver = {
+		.name	= "asoc-simple-card",
+	},
+	.probe		= asoc_simple_card_probe,
+	.remove		= asoc_simple_card_remove,
+};
+
+module_platform_driver(asoc_simple_card);
+
+MODULE_LICENSE("GPL");
+MODULE_DESCRIPTION("ASoC Simple Sound Card");
+MODULE_AUTHOR("Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>");
diff --git a/sound/soc/imx/Kconfig b/sound/soc/imx/Kconfig
deleted file mode 100644
index d83e5d0b5d52..000000000000
--- a/sound/soc/imx/Kconfig
+++ /dev/null
@@ -1,79 +0,0 @@
-menuconfig SND_IMX_SOC
-	tristate "SoC Audio for Freescale i.MX CPUs"
-	depends on ARCH_MXC
-	help
-	  Say Y or M if you want to add support for codecs attached to
-	  the i.MX SSI interface.
-
-
-if SND_IMX_SOC
-
-config SND_SOC_IMX_SSI
-	tristate
-
-config SND_SOC_IMX_PCM
-	tristate
-
-config SND_MXC_SOC_FIQ
-	tristate
-	select FIQ
-	select SND_SOC_IMX_PCM
-
-config SND_MXC_SOC_MX2
-	select SND_SOC_DMAENGINE_PCM
-	tristate
-	select SND_SOC_IMX_PCM
-
-config SND_SOC_IMX_AUDMUX
-	tristate
-
-config SND_MXC_SOC_WM1133_EV1
-	tristate "Audio on the i.MX31ADS with WM1133-EV1 fitted"
-	depends on MACH_MX31ADS_WM1133_EV1 && EXPERIMENTAL
-	select SND_SOC_WM8350
-	select SND_MXC_SOC_FIQ
-	select SND_SOC_IMX_AUDMUX
-	select SND_SOC_IMX_SSI
-	help
-	  Enable support for audio on the i.MX31ADS with the WM1133-EV1
-	  PMIC board with WM8835x fitted.
-
-config SND_SOC_MX27VIS_AIC32X4
-	tristate "SoC audio support for Visstrim M10 boards"
-	depends on MACH_IMX27_VISSTRIM_M10 && I2C
-	select SND_SOC_TLV320AIC32X4
-	select SND_MXC_SOC_MX2
-	select SND_SOC_IMX_AUDMUX
-	select SND_SOC_IMX_SSI
-	help
-	  Say Y if you want to add support for SoC audio on Visstrim SM10
-	  board with TLV320AIC32X4 codec.
-
-config SND_SOC_PHYCORE_AC97
-	tristate "SoC Audio support for Phytec phyCORE (and phyCARD) boards"
-	depends on MACH_PCM043 || MACH_PCA100
-	select SND_SOC_AC97_BUS
-	select SND_SOC_WM9712
-	select SND_MXC_SOC_FIQ
-	select SND_SOC_IMX_AUDMUX
-	select SND_SOC_IMX_SSI
-	help
-	  Say Y if you want to add support for SoC audio on Phytec phyCORE
-	  and phyCARD boards in AC97 mode
-
-config SND_SOC_EUKREA_TLV320
-	tristate "Eukrea TLV320"
-	depends on MACH_EUKREA_MBIMX27_BASEBOARD \
-		|| MACH_EUKREA_MBIMXSD25_BASEBOARD \
-		|| MACH_EUKREA_MBIMXSD35_BASEBOARD \
-		|| MACH_EUKREA_MBIMXSD51_BASEBOARD
-	depends on I2C
-	select SND_SOC_TLV320AIC23
-	select SND_MXC_SOC_FIQ
-	select SND_SOC_IMX_AUDMUX
-	select SND_SOC_IMX_SSI
-	help
-	  Enable I2S based access to the TLV320AIC23B codec attached
-	  to the SSI interface
-
-endif	# SND_IMX_SOC
diff --git a/sound/soc/imx/Makefile b/sound/soc/imx/Makefile
deleted file mode 100644
index f5db3e92d0d1..000000000000
--- a/sound/soc/imx/Makefile
+++ /dev/null
@@ -1,22 +0,0 @@
-# i.MX Platform Support
-snd-soc-imx-ssi-objs := imx-ssi.o
-snd-soc-imx-audmux-objs := imx-audmux.o
-
-obj-$(CONFIG_SND_SOC_IMX_SSI) += snd-soc-imx-ssi.o
-obj-$(CONFIG_SND_SOC_IMX_AUDMUX) += snd-soc-imx-audmux.o
-
-obj-$(CONFIG_SND_SOC_IMX_PCM) += snd-soc-imx-pcm.o
-snd-soc-imx-pcm-y := imx-pcm.o
-snd-soc-imx-pcm-$(CONFIG_SND_MXC_SOC_FIQ) += imx-pcm-fiq.o
-snd-soc-imx-pcm-$(CONFIG_SND_MXC_SOC_MX2) += imx-pcm-dma-mx2.o
-
-# i.MX Machine Support
-snd-soc-eukrea-tlv320-objs := eukrea-tlv320.o
-snd-soc-phycore-ac97-objs := phycore-ac97.o
-snd-soc-mx27vis-aic32x4-objs := mx27vis-aic32x4.o
-snd-soc-wm1133-ev1-objs := wm1133-ev1.o
-
-obj-$(CONFIG_SND_SOC_EUKREA_TLV320) += snd-soc-eukrea-tlv320.o
-obj-$(CONFIG_SND_SOC_PHYCORE_AC97) += snd-soc-phycore-ac97.o
-obj-$(CONFIG_SND_SOC_MX27VIS_AIC32X4) += snd-soc-mx27vis-aic32x4.o
-obj-$(CONFIG_SND_MXC_SOC_WM1133_EV1) += snd-soc-wm1133-ev1.o
diff --git a/sound/soc/jz4740/jz4740-i2s.c b/sound/soc/jz4740/jz4740-i2s.c
index a5af7c42e62b..41349670adab 100644
--- a/sound/soc/jz4740/jz4740-i2s.c
+++ b/sound/soc/jz4740/jz4740-i2s.c
@@ -346,7 +346,7 @@ static void jz4740_i2c_init_pcm_config(struct jz4740_i2s *i2s)
 
 	/* Playback */
 	dma_config = &i2s->pcm_config_playback.dma_config;
-	dma_config->src_width = JZ4740_DMA_WIDTH_32BIT,
+	dma_config->src_width = JZ4740_DMA_WIDTH_32BIT;
 	dma_config->transfer_size = JZ4740_DMA_TRANSFER_SIZE_16BYTE;
 	dma_config->request_type = JZ4740_DMA_TYPE_AIC_TRANSMIT;
 	dma_config->flags = JZ4740_DMA_SRC_AUTOINC;
@@ -355,7 +355,7 @@ static void jz4740_i2c_init_pcm_config(struct jz4740_i2s *i2s)
 
 	/* Capture */
 	dma_config = &i2s->pcm_config_capture.dma_config;
-	dma_config->dst_width = JZ4740_DMA_WIDTH_32BIT,
+	dma_config->dst_width = JZ4740_DMA_WIDTH_32BIT;
 	dma_config->transfer_size = JZ4740_DMA_TRANSFER_SIZE_16BYTE;
 	dma_config->request_type = JZ4740_DMA_TYPE_AIC_RECEIVE;
 	dma_config->flags = JZ4740_DMA_DST_AUTOINC;
diff --git a/sound/soc/mxs/mxs-pcm.c b/sound/soc/mxs/mxs-pcm.c
index e373fbbc97a0..373dec90579f 100644
--- a/sound/soc/mxs/mxs-pcm.c
+++ b/sound/soc/mxs/mxs-pcm.c
@@ -220,28 +220,16 @@ static struct snd_soc_platform_driver mxs_soc_platform = {
 	.pcm_free	= mxs_pcm_free,
 };
 
-static int __devinit mxs_soc_platform_probe(struct platform_device *pdev)
+int __devinit mxs_pcm_platform_register(struct device *dev)
 {
-	return snd_soc_register_platform(&pdev->dev, &mxs_soc_platform);
+	return snd_soc_register_platform(dev, &mxs_soc_platform);
 }
+EXPORT_SYMBOL_GPL(mxs_pcm_platform_register);
 
-static int __devexit mxs_soc_platform_remove(struct platform_device *pdev)
+void __devexit mxs_pcm_platform_unregister(struct device *dev)
 {
-	snd_soc_unregister_platform(&pdev->dev);
-
-	return 0;
+	snd_soc_unregister_platform(dev);
 }
-
-static struct platform_driver mxs_pcm_driver = {
-	.driver = {
-		.name = "mxs-pcm-audio",
-		.owner = THIS_MODULE,
-	},
-	.probe = mxs_soc_platform_probe,
-	.remove = __devexit_p(mxs_soc_platform_remove),
-};
-
-module_platform_driver(mxs_pcm_driver);
+EXPORT_SYMBOL_GPL(mxs_pcm_platform_unregister);
 
 MODULE_LICENSE("GPL");
-MODULE_ALIAS("platform:mxs-pcm-audio");
diff --git a/sound/soc/mxs/mxs-pcm.h b/sound/soc/mxs/mxs-pcm.h
index 5f01a9124b3d..35ba2ca42384 100644
--- a/sound/soc/mxs/mxs-pcm.h
+++ b/sound/soc/mxs/mxs-pcm.h
@@ -24,4 +24,7 @@ struct mxs_pcm_dma_params {
 	int chan_num;
 };
 
+int mxs_pcm_platform_register(struct device *dev);
+void mxs_pcm_platform_unregister(struct device *dev);
+
 #endif
diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c
index 7fd224bb7324..aba71bfa33b1 100644
--- a/sound/soc/mxs/mxs-saif.c
+++ b/sound/soc/mxs/mxs-saif.c
@@ -18,6 +18,8 @@
 
 #include <linux/module.h>
 #include <linux/init.h>
+#include <linux/of.h>
+#include <linux/of_device.h>
 #include <linux/platform_device.h>
 #include <linux/slab.h>
 #include <linux/dma-mapping.h>
@@ -621,37 +623,57 @@ static irqreturn_t mxs_saif_irq(int irq, void *dev_id)
 	return IRQ_HANDLED;
 }
 
-static int mxs_saif_probe(struct platform_device *pdev)
+static int __devinit mxs_saif_probe(struct platform_device *pdev)
 {
+	struct device_node *np = pdev->dev.of_node;
 	struct resource *iores, *dmares;
 	struct mxs_saif *saif;
 	struct mxs_saif_platform_data *pdata;
 	struct pinctrl *pinctrl;
 	int ret = 0;
 
-	if (pdev->id >= ARRAY_SIZE(mxs_saif))
+
+	if (!np && pdev->id >= ARRAY_SIZE(mxs_saif))
 		return -EINVAL;
 
 	saif = devm_kzalloc(&pdev->dev, sizeof(*saif), GFP_KERNEL);
 	if (!saif)
 		return -ENOMEM;
 
-	mxs_saif[pdev->id] = saif;
-	saif->id = pdev->id;
-
-	pdata = pdev->dev.platform_data;
-	if (pdata && !pdata->master_mode) {
-		saif->master_id = pdata->master_id;
-		if (saif->master_id < 0 ||
-			saif->master_id >= ARRAY_SIZE(mxs_saif) ||
-			saif->master_id == saif->id) {
-			dev_err(&pdev->dev, "get wrong master id\n");
-			return -EINVAL;
+	if (np) {
+		struct device_node *master;
+		saif->id = of_alias_get_id(np, "saif");
+		if (saif->id < 0)
+			return saif->id;
+		/*
+		 * If there is no "fsl,saif-master" phandle, it's a saif
+		 * master.  Otherwise, it's a slave and its phandle points
+		 * to the master.
+		 */
+		master = of_parse_phandle(np, "fsl,saif-master", 0);
+		if (!master) {
+			saif->master_id = saif->id;
+		} else {
+			saif->master_id = of_alias_get_id(master, "saif");
+			if (saif->master_id < 0)
+				return saif->master_id;
 		}
 	} else {
-		saif->master_id = saif->id;
+		saif->id = pdev->id;
+		pdata = pdev->dev.platform_data;
+		if (pdata && !pdata->master_mode)
+			saif->master_id = pdata->master_id;
+		else
+			saif->master_id = saif->id;
 	}
 
+	if (saif->master_id < 0 || saif->master_id >= ARRAY_SIZE(mxs_saif)) {
+		dev_err(&pdev->dev, "get wrong master id\n");
+		return -EINVAL;
+	}
+
+	mxs_saif[saif->id] = saif;
+
 	pinctrl = devm_pinctrl_get_select_default(&pdev->dev);
 	if (IS_ERR(pinctrl)) {
 		ret = PTR_ERR(pinctrl);
@@ -677,12 +699,19 @@ static int mxs_saif_probe(struct platform_device *pdev)
 
 	dmares = platform_get_resource(pdev, IORESOURCE_DMA, 0);
 	if (!dmares) {
-		ret = -ENODEV;
-		dev_err(&pdev->dev, "failed to get dma resource: %d\n",
-			ret);
-		goto failed_get_resource;
+		/*
+		 * TODO: This is a temporary solution and should be changed
+		 * to use generic DMA binding later when the helplers get in.
+		 */
+		ret = of_property_read_u32(np, "fsl,saif-dma-channel",
+					   &saif->dma_param.chan_num);
+		if (ret) {
+			dev_err(&pdev->dev, "failed to get dma channel\n");
+			goto failed_get_resource;
+		}
+	} else {
+		saif->dma_param.chan_num = dmares->start;
 	}
-	saif->dma_param.chan_num = dmares->start;
 
 	saif->irq = platform_get_irq(pdev, 0);
 	if (saif->irq < 0) {
@@ -716,24 +745,14 @@ static int mxs_saif_probe(struct platform_device *pdev)
 		goto failed_get_resource;
 	}
 
-	saif->soc_platform_pdev = platform_device_alloc(
-					"mxs-pcm-audio", pdev->id);
-	if (!saif->soc_platform_pdev) {
-		ret = -ENOMEM;
-		goto failed_pdev_alloc;
-	}
-
-	platform_set_drvdata(saif->soc_platform_pdev, saif);
-	ret = platform_device_add(saif->soc_platform_pdev);
+	ret = mxs_pcm_platform_register(&pdev->dev);
 	if (ret) {
-		dev_err(&pdev->dev, "failed to add soc platform device\n");
-		goto failed_pdev_add;
+		dev_err(&pdev->dev, "register PCM failed: %d\n", ret);
+		goto failed_pdev_alloc;
 	}
 
 	return 0;
 
-failed_pdev_add:
-	platform_device_put(saif->soc_platform_pdev);
 failed_pdev_alloc:
 	snd_soc_unregister_dai(&pdev->dev);
 failed_get_resource:
@@ -746,13 +765,19 @@ static int __devexit mxs_saif_remove(struct platform_device *pdev)
 {
 	struct mxs_saif *saif = platform_get_drvdata(pdev);
 
-	platform_device_unregister(saif->soc_platform_pdev);
+	mxs_pcm_platform_unregister(&pdev->dev);
 	snd_soc_unregister_dai(&pdev->dev);
 	clk_put(saif->clk);
 
 	return 0;
 }
 
+static const struct of_device_id mxs_saif_dt_ids[] = {
+	{ .compatible = "fsl,imx28-saif", },
+	{ /* sentinel */ }
+};
+MODULE_DEVICE_TABLE(of, mxs_saif_dt_ids);
+
 static struct platform_driver mxs_saif_driver = {
 	.probe = mxs_saif_probe,
 	.remove = __devexit_p(mxs_saif_remove),
@@ -760,6 +785,7 @@ static struct platform_driver mxs_saif_driver = {
 	.driver = {
 		.name = "mxs-saif",
 		.owner = THIS_MODULE,
+		.of_match_table = mxs_saif_dt_ids,
 	},
 };
 
diff --git a/sound/soc/mxs/mxs-saif.h b/sound/soc/mxs/mxs-saif.h
index 12c91e4eb941..3cb342e5bc90 100644
--- a/sound/soc/mxs/mxs-saif.h
+++ b/sound/soc/mxs/mxs-saif.h
@@ -123,7 +123,6 @@ struct mxs_saif {
 	unsigned int cur_rate;
 	unsigned int ongoing;
 
-	struct platform_device *soc_platform_pdev;
 	u32 fifo_underrun;
 	u32 fifo_overrun;
 };
diff --git a/sound/soc/mxs/mxs-sgtl5000.c b/sound/soc/mxs/mxs-sgtl5000.c
index 60f052b7cf22..3e6e8764b2e6 100644
--- a/sound/soc/mxs/mxs-sgtl5000.c
+++ b/sound/soc/mxs/mxs-sgtl5000.c
@@ -18,6 +18,8 @@
 
 #include <linux/module.h>
 #include <linux/device.h>
+#include <linux/of.h>
+#include <linux/of_device.h>
 #include <sound/core.h>
 #include <sound/pcm.h>
 #include <sound/soc.h>
@@ -90,7 +92,7 @@ static struct snd_soc_dai_link mxs_sgtl5000_dai[] = {
 		.codec_dai_name	= "sgtl5000",
 		.codec_name	= "sgtl5000.0-000a",
 		.cpu_dai_name	= "mxs-saif.0",
-		.platform_name	= "mxs-pcm-audio.0",
+		.platform_name	= "mxs-saif.0",
 		.ops		= &mxs_sgtl5000_hifi_ops,
 	}, {
 		.name		= "HiFi Rx",
@@ -98,7 +100,7 @@ static struct snd_soc_dai_link mxs_sgtl5000_dai[] = {
 		.codec_dai_name	= "sgtl5000",
 		.codec_name	= "sgtl5000.0-000a",
 		.cpu_dai_name	= "mxs-saif.1",
-		.platform_name	= "mxs-pcm-audio.1",
+		.platform_name	= "mxs-saif.1",
 		.ops		= &mxs_sgtl5000_hifi_ops,
 	},
 };
@@ -110,11 +112,48 @@ static struct snd_soc_card mxs_sgtl5000 = {
 	.num_links	= ARRAY_SIZE(mxs_sgtl5000_dai),
 };
 
+static int __devinit mxs_sgtl5000_probe_dt(struct platform_device *pdev)
+{
+	struct device_node *np = pdev->dev.of_node;
+	struct device_node *saif_np[2], *codec_np;
+	int i, ret = 0;
+
+	if (!np)
+		return 1; /* no device tree */
+
+	saif_np[0] = of_parse_phandle(np, "saif-controllers", 0);
+	saif_np[1] = of_parse_phandle(np, "saif-controllers", 1);
+	codec_np = of_parse_phandle(np, "audio-codec", 0);
+	if (!saif_np[0] || !saif_np[1] || !codec_np) {
+		dev_err(&pdev->dev, "phandle missing or invalid\n");
+		return -EINVAL;
+	}
+
+	for (i = 0; i < 2; i++) {
+		mxs_sgtl5000_dai[i].codec_name = NULL;
+		mxs_sgtl5000_dai[i].codec_of_node = codec_np;
+		mxs_sgtl5000_dai[i].cpu_dai_name = NULL;
+		mxs_sgtl5000_dai[i].cpu_dai_of_node = saif_np[i];
+		mxs_sgtl5000_dai[i].platform_name = NULL;
+		mxs_sgtl5000_dai[i].platform_of_node = saif_np[i];
+	}
+
+	of_node_put(codec_np);
+	of_node_put(saif_np[0]);
+	of_node_put(saif_np[1]);
+
+	return ret;
+}
+
 static int __devinit mxs_sgtl5000_probe(struct platform_device *pdev)
 {
 	struct snd_soc_card *card = &mxs_sgtl5000;
 	int ret;
 
+	ret = mxs_sgtl5000_probe_dt(pdev);
+	if (ret < 0)
+		return ret;
+
 	/*
 	 * Set an init clock(11.28Mhz) for sgtl5000 initialization(i2c r/w).
 	 * The Sgtl5000 sysclk is derived from saif0 mclk and it's range
@@ -148,10 +187,17 @@ static int __devexit mxs_sgtl5000_remove(struct platform_device *pdev)
 	return 0;
 }
 
+static const struct of_device_id mxs_sgtl5000_dt_ids[] = {
+	{ .compatible = "fsl,mxs-audio-sgtl5000", },
+	{ /* sentinel */ }
+};
+MODULE_DEVICE_TABLE(of, mxs_sgtl5000_dt_ids);
+
 static struct platform_driver mxs_sgtl5000_audio_driver = {
 	.driver = {
 		.name = "mxs-sgtl5000",
 		.owner = THIS_MODULE,
+		.of_match_table = mxs_sgtl5000_dt_ids,
 	},
 	.probe = mxs_sgtl5000_probe,
 	.remove = __devexit_p(mxs_sgtl5000_remove),
diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig
index deafbfaacdbf..9ccfa5e1c11b 100644
--- a/sound/soc/omap/Kconfig
+++ b/sound/soc/omap/Kconfig
@@ -113,6 +113,7 @@ config SND_OMAP_SOC_OMAP4_HDMI
 	tristate "SoC Audio support for Texas Instruments OMAP4 HDMI"
 	depends on SND_OMAP_SOC && OMAP4_DSS_HDMI && OMAP2_DSS && ARCH_OMAP4
 	select SND_OMAP_SOC_HDMI
+	select SND_SOC_OMAP_HDMI_CODEC
 	help
 	  Say Y if you want to add support for SoC HDMI audio on Texas Instruments
 	  OMAP4 chips
diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c
index fd04ce139031..1c2aa7fab3fd 100644
--- a/sound/soc/pxa/pxa-ssp.c
+++ b/sound/soc/pxa/pxa-ssp.c
@@ -85,14 +85,12 @@ struct pxa2xx_pcm_dma_data {
 	char name[20];
 };
 
-static struct pxa2xx_pcm_dma_params *
-pxa_ssp_get_dma_params(struct ssp_device *ssp, int width4, int out)
+static void pxa_ssp_set_dma_params(struct ssp_device *ssp, int width4,
+			int out, struct pxa2xx_pcm_dma_params *dma_data)
 {
 	struct pxa2xx_pcm_dma_data *dma;
 
-	dma = kzalloc(sizeof(struct pxa2xx_pcm_dma_data), GFP_KERNEL);
-	if (dma == NULL)
-		return NULL;
+	dma = container_of(dma_data, struct pxa2xx_pcm_dma_data, params);
 
 	snprintf(dma->name, 20, "SSP%d PCM %s %s", ssp->port_id,
 			width4 ? "32-bit" : "16-bit", out ? "out" : "in");
@@ -103,8 +101,6 @@ pxa_ssp_get_dma_params(struct ssp_device *ssp, int width4, int out)
 				  (DCMD_INCTRGADDR | DCMD_FLOWSRC)) |
 			(width4 ? DCMD_WIDTH4 : DCMD_WIDTH2) | DCMD_BURST16;
 	dma->params.dev_addr = ssp->phys_base + SSDR;
-
-	return &dma->params;
 }
 
 static int pxa_ssp_startup(struct snd_pcm_substream *substream,
@@ -112,6 +108,7 @@ static int pxa_ssp_startup(struct snd_pcm_substream *substream,
 {
 	struct ssp_priv *priv = snd_soc_dai_get_drvdata(cpu_dai);
 	struct ssp_device *ssp = priv->ssp;
+	struct pxa2xx_pcm_dma_data *dma;
 	int ret = 0;
 
 	if (!cpu_dai->active) {
@@ -119,8 +116,10 @@ static int pxa_ssp_startup(struct snd_pcm_substream *substream,
 		pxa_ssp_disable(ssp);
 	}
 
-	kfree(snd_soc_dai_get_dma_data(cpu_dai, substream));
-	snd_soc_dai_set_dma_data(cpu_dai, substream, NULL);
+	dma = kzalloc(sizeof(struct pxa2xx_pcm_dma_data), GFP_KERNEL);
+	if (!dma)
+		return -ENOMEM;
+	snd_soc_dai_set_dma_data(cpu_dai, substream, &dma->params);
 
 	return ret;
 }
@@ -573,18 +572,13 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream,
 
 	dma_data = snd_soc_dai_get_dma_data(cpu_dai, substream);
 
-	/* generate correct DMA params */
-	kfree(dma_data);
-
 	/* Network mode with one active slot (ttsa == 1) can be used
 	 * to force 16-bit frame width on the wire (for S16_LE), even
 	 * with two channels. Use 16-bit DMA transfers for this case.
 	 */
-	dma_data = pxa_ssp_get_dma_params(ssp,
-			((chn == 2) && (ttsa != 1)) || (width == 32),
-			substream->stream == SNDRV_PCM_STREAM_PLAYBACK);
-
-	snd_soc_dai_set_dma_data(cpu_dai, substream, dma_data);
+	pxa_ssp_set_dma_params(ssp,
+		((chn == 2) && (ttsa != 1)) || (width == 32),
+		substream->stream == SNDRV_PCM_STREAM_PLAYBACK, dma_data);
 
 	/* we can only change the settings if the port is not in use */
 	if (pxa_ssp_read_reg(ssp, SSCR0) & SSCR0_SSE)
diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c
index d08583790d23..3075a426124c 100644
--- a/sound/soc/pxa/pxa2xx-i2s.c
+++ b/sound/soc/pxa/pxa2xx-i2s.c
@@ -166,7 +166,7 @@ static int pxa2xx_i2s_hw_params(struct snd_pcm_substream *substream,
 	struct pxa2xx_pcm_dma_params *dma_data;
 
 	BUG_ON(IS_ERR(clk_i2s));
-	clk_enable(clk_i2s);
+	clk_prepare_enable(clk_i2s);
 	clk_ena = 1;
 	pxa_i2s_wait();
 
@@ -259,7 +259,7 @@ static void pxa2xx_i2s_shutdown(struct snd_pcm_substream *substream,
 		SACR0 &= ~SACR0_ENB;
 		pxa_i2s_wait();
 		if (clk_ena) {
-			clk_disable(clk_i2s);
+			clk_disable_unprepare(clk_i2s);
 			clk_ena = 0;
 		}
 	}
diff --git a/sound/soc/samsung/littlemill.c b/sound/soc/samsung/littlemill.c
index e7416851bf7d..c82c646b8a08 100644
--- a/sound/soc/samsung/littlemill.c
+++ b/sound/soc/samsung/littlemill.c
@@ -23,10 +23,10 @@ static int littlemill_set_bias_level(struct snd_soc_card *card,
 					  struct snd_soc_dapm_context *dapm,
 					  enum snd_soc_bias_level level)
 {
-	struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai;
+	struct snd_soc_dai *aif1_dai = card->rtd[0].codec_dai;
 	int ret;
 
-	if (dapm->dev != codec_dai->dev)
+	if (dapm->dev != aif1_dai->dev)
 		return 0;
 
 	switch (level) {
@@ -36,7 +36,7 @@ static int littlemill_set_bias_level(struct snd_soc_card *card,
 		 * then do so now, otherwise these are noops.
 		 */
 		if (dapm->bias_level == SND_SOC_BIAS_STANDBY) {
-			ret = snd_soc_dai_set_pll(codec_dai, WM8994_FLL1,
+			ret = snd_soc_dai_set_pll(aif1_dai, WM8994_FLL1,
 						  WM8994_FLL_SRC_MCLK2, 32768,
 						  sample_rate * 512);
 			if (ret < 0) {
@@ -44,7 +44,7 @@ static int littlemill_set_bias_level(struct snd_soc_card *card,
 				return ret;
 			}
 
-			ret = snd_soc_dai_set_sysclk(codec_dai,
+			ret = snd_soc_dai_set_sysclk(aif1_dai,
 						     WM8994_SYSCLK_FLL1,
 						     sample_rate * 512,
 						     SND_SOC_CLOCK_IN);
@@ -66,25 +66,25 @@ static int littlemill_set_bias_level_post(struct snd_soc_card *card,
 					       struct snd_soc_dapm_context *dapm,
 					       enum snd_soc_bias_level level)
 {
-	struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai;
+	struct snd_soc_dai *aif1_dai = card->rtd[0].codec_dai;
 	int ret;
 
-	if (dapm->dev != codec_dai->dev)
+	if (dapm->dev != aif1_dai->dev)
 		return 0;
 
 	switch (level) {
 	case SND_SOC_BIAS_STANDBY:
-		ret = snd_soc_dai_set_sysclk(codec_dai, WM8994_SYSCLK_MCLK2,
+		ret = snd_soc_dai_set_sysclk(aif1_dai, WM8994_SYSCLK_MCLK2,
 					     32768, SND_SOC_CLOCK_IN);
 		if (ret < 0) {
-			pr_err("Failed to switch away from FLL: %d\n", ret);
+			pr_err("Failed to switch away from FLL1: %d\n", ret);
 			return ret;
 		}
 
-		ret = snd_soc_dai_set_pll(codec_dai, WM8994_FLL1,
+		ret = snd_soc_dai_set_pll(aif1_dai, WM8994_FLL1,
 					  0, 0, 0);
 		if (ret < 0) {
-			pr_err("Failed to stop FLL: %d\n", ret);
+			pr_err("Failed to stop FLL1: %d\n", ret);
 			return ret;
 		}
 		break;
@@ -131,6 +131,14 @@ static struct snd_soc_ops littlemill_ops = {
 	.hw_params = littlemill_hw_params,
 };
 
+static const struct snd_soc_pcm_stream baseband_params = {
+	.formats = SNDRV_PCM_FMTBIT_S32_LE,
+	.rate_min = 8000,
+	.rate_max = 8000,
+	.channels_min = 2,
+	.channels_max = 2,
+};
+
 static struct snd_soc_dai_link littlemill_dai[] = {
 	{
 		.name = "CPU",
@@ -143,13 +151,75 @@ static struct snd_soc_dai_link littlemill_dai[] = {
 				| SND_SOC_DAIFMT_CBM_CFM,
 		.ops = &littlemill_ops,
 	},
+	{
+		.name = "Baseband",
+		.stream_name = "Baseband",
+		.cpu_dai_name = "wm8994-aif2",
+		.codec_dai_name = "wm1250-ev1",
+		.codec_name = "wm1250-ev1.1-0027",
+		.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
+				| SND_SOC_DAIFMT_CBM_CFM,
+		.ignore_suspend = 1,
+		.params = &baseband_params,
+	},
 };
 
+static int bbclk_ev(struct snd_soc_dapm_widget *w,
+		    struct snd_kcontrol *kcontrol, int event)
+{
+	struct snd_soc_card *card = w->dapm->card;
+	struct snd_soc_dai *aif2_dai = card->rtd[1].cpu_dai;
+	int ret;
+
+	switch (event) {
+	case SND_SOC_DAPM_PRE_PMU:
+		ret = snd_soc_dai_set_pll(aif2_dai, WM8994_FLL2,
+					  WM8994_FLL_SRC_BCLK, 64 * 8000,
+					  8000 * 256);
+		if (ret < 0) {
+			pr_err("Failed to start FLL: %d\n", ret);
+			return ret;
+		}
+
+		ret = snd_soc_dai_set_sysclk(aif2_dai, WM8994_SYSCLK_FLL2,
+					     8000 * 256,
+					     SND_SOC_CLOCK_IN);
+		if (ret < 0) {
+			pr_err("Failed to set SYSCLK: %d\n", ret);
+			return ret;
+		}
+		break;
+	case SND_SOC_DAPM_POST_PMD:
+		ret = snd_soc_dai_set_sysclk(aif2_dai, WM8994_SYSCLK_MCLK2,
+					     32768, SND_SOC_CLOCK_IN);
+		if (ret < 0) {
+			pr_err("Failed to switch away from FLL2: %d\n", ret);
+			return ret;
+		}
+
+		ret = snd_soc_dai_set_pll(aif2_dai, WM8994_FLL2,
+					  0, 0, 0);
+		if (ret < 0) {
+			pr_err("Failed to stop FLL2: %d\n", ret);
+			return ret;
+		}
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	return 0;
+}
+
 static struct snd_soc_dapm_widget widgets[] = {
 	SND_SOC_DAPM_HP("Headphone", NULL),
 
 	SND_SOC_DAPM_MIC("AMIC", NULL),
 	SND_SOC_DAPM_MIC("DMIC", NULL),
+
+	SND_SOC_DAPM_SUPPLY_S("Baseband Clock", -1, SND_SOC_NOPM, 0, 0,
+			      bbclk_ev,
+			      SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
 };
 
 static struct snd_soc_dapm_route audio_paths[] = {
@@ -162,6 +232,8 @@ static struct snd_soc_dapm_route audio_paths[] = {
 	{ "DMIC", NULL, "MICBIAS2" },   /* Default for DMICBIAS jumper */
 	{ "DMIC1DAT", NULL, "DMIC" },
 	{ "DMIC2DAT", NULL, "DMIC" },
+
+	{ "AIF2CLK", NULL, "Baseband Clock" },
 };
 
 static struct snd_soc_jack littlemill_headset;
@@ -169,10 +241,16 @@ static struct snd_soc_jack littlemill_headset;
 static int littlemill_late_probe(struct snd_soc_card *card)
 {
 	struct snd_soc_codec *codec = card->rtd[0].codec;
-	struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai;
+	struct snd_soc_dai *aif1_dai = card->rtd[0].codec_dai;
+	struct snd_soc_dai *aif2_dai = card->rtd[1].cpu_dai;
 	int ret;
 
-	ret = snd_soc_dai_set_sysclk(codec_dai, WM8994_SYSCLK_MCLK2,
+	ret = snd_soc_dai_set_sysclk(aif1_dai, WM8994_SYSCLK_MCLK2,
+				     32768, SND_SOC_CLOCK_IN);
+	if (ret < 0)
+		return ret;
+
+	ret = snd_soc_dai_set_sysclk(aif2_dai, WM8994_SYSCLK_MCLK2,
 				     32768, SND_SOC_CLOCK_IN);
 	if (ret < 0)
 		return ret;
diff --git a/sound/soc/samsung/lowland.c b/sound/soc/samsung/lowland.c
index 4adff934f771..6abf341c4a2a 100644
--- a/sound/soc/samsung/lowland.c
+++ b/sound/soc/samsung/lowland.c
@@ -21,33 +21,6 @@
 #define MCLK1_RATE (44100 * 512)
 #define CLKOUT_RATE (44100 * 256)
 
-static int lowland_hw_params(struct snd_pcm_substream *substream,
-			     struct snd_pcm_hw_params *params)
-{
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
-	int ret;
-
-	ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S
-					 | SND_SOC_DAIFMT_NB_NF
-					 | SND_SOC_DAIFMT_CBM_CFM);
-	if (ret < 0)
-		return ret;
-
-	ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S
-					 | SND_SOC_DAIFMT_NB_NF
-					 | SND_SOC_DAIFMT_CBM_CFM);
-	if (ret < 0)
-		return ret;
-
-	return 0;
-}
-
-static struct snd_soc_ops lowland_ops = {
-	.hw_params = lowland_hw_params,
-};
-
 static struct snd_soc_jack lowland_headset;
 
 /* Headset jack detection DAPM pins */
@@ -101,6 +74,25 @@ static int lowland_wm5100_init(struct snd_soc_pcm_runtime *rtd)
 	return 0;
 }
 
+static int lowland_wm9081_init(struct snd_soc_pcm_runtime *rtd)
+{
+	struct snd_soc_codec *codec = rtd->codec;
+
+	snd_soc_dapm_nc_pin(&codec->dapm, "LINEOUT");
+
+	/* At any time the WM9081 is active it will have this clock */
+	return snd_soc_codec_set_sysclk(codec, WM9081_SYSCLK_MCLK, 0,
+					CLKOUT_RATE, 0);
+}
+
+static const struct snd_soc_pcm_stream sub_params = {
+	.formats = SNDRV_PCM_FMTBIT_S32_LE,
+	.rate_min = 44100,
+	.rate_max = 44100,
+	.channels_min = 2,
+	.channels_max = 2,
+};
+
 static struct snd_soc_dai_link lowland_dai[] = {
 	{
 		.name = "CPU",
@@ -109,7 +101,8 @@ static struct snd_soc_dai_link lowland_dai[] = {
 		.codec_dai_name = "wm5100-aif1",
 		.platform_name = "samsung-audio",
 		.codec_name = "wm5100.1-001a",
-		.ops = &lowland_ops,
+		.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+				SND_SOC_DAIFMT_CBM_CFM,
 		.init = lowland_wm5100_init,
 	},
 	{
@@ -118,24 +111,20 @@ static struct snd_soc_dai_link lowland_dai[] = {
 		.cpu_dai_name = "wm5100-aif2",
 		.codec_dai_name = "wm1250-ev1",
 		.codec_name = "wm1250-ev1.1-0027",
-		.ops = &lowland_ops,
+		.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+				SND_SOC_DAIFMT_CBM_CFM,
 		.ignore_suspend = 1,
 	},
-};
-
-static int lowland_wm9081_init(struct snd_soc_dapm_context *dapm)
-{
-	snd_soc_dapm_nc_pin(dapm, "LINEOUT");
-
-	/* At any time the WM9081 is active it will have this clock */
-	return snd_soc_codec_set_sysclk(dapm->codec, WM9081_SYSCLK_MCLK, 0,
-					CLKOUT_RATE, 0);
-}
-
-static struct snd_soc_aux_dev lowland_aux_dev[] = {
 	{
-		.name = "wm9081",
+		.name = "Sub Speaker",
+		.stream_name = "Sub Speaker",
+		.cpu_dai_name = "wm5100-aif3",
+		.codec_dai_name = "wm9081-hifi",
 		.codec_name = "wm9081.1-006c",
+		.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+				SND_SOC_DAIFMT_CBM_CFM,
+		.ignore_suspend = 1,
+		.params = &sub_params,
 		.init = lowland_wm9081_init,
 	},
 };
@@ -180,8 +169,6 @@ static struct snd_soc_card lowland = {
 	.owner = THIS_MODULE,
 	.dai_link = lowland_dai,
 	.num_links = ARRAY_SIZE(lowland_dai),
-	.aux_dev = lowland_aux_dev,
-	.num_aux_devs = ARRAY_SIZE(lowland_aux_dev),
 	.codec_conf = lowland_codec_conf,
 	.num_configs = ARRAY_SIZE(lowland_codec_conf),
 
diff --git a/sound/soc/samsung/speyside.c b/sound/soc/samsung/speyside.c
index f9ab7707a3e4..a4a9fc7e8c76 100644
--- a/sound/soc/samsung/speyside.c
+++ b/sound/soc/samsung/speyside.c
@@ -92,33 +92,6 @@ static int speyside_set_bias_level_post(struct snd_soc_card *card,
 	return 0;
 }
 
-static int speyside_hw_params(struct snd_pcm_substream *substream,
-			      struct snd_pcm_hw_params *params)
-{
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
-	int ret;
-
-	ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S
-					 | SND_SOC_DAIFMT_NB_NF
-					 | SND_SOC_DAIFMT_CBM_CFM);
-	if (ret < 0)
-		return ret;
-
-	ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S
-					 | SND_SOC_DAIFMT_NB_NF
-					 | SND_SOC_DAIFMT_CBM_CFM);
-	if (ret < 0)
-		return ret;
-
-	return 0;
-}
-
-static struct snd_soc_ops speyside_ops = {
-	.hw_params = speyside_hw_params,
-};
-
 static struct snd_soc_jack speyside_headset;
 
 /* Headset jack detection DAPM pins */
@@ -208,7 +181,8 @@ static struct snd_soc_dai_link speyside_dai[] = {
 		.platform_name = "samsung-audio",
 		.codec_name = "wm8996.1-001a",
 		.init = speyside_wm8996_init,
-		.ops = &speyside_ops,
+		.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
+				| SND_SOC_DAIFMT_CBM_CFM,
 	},
 	{
 		.name = "Baseband",
@@ -216,7 +190,8 @@ static struct snd_soc_dai_link speyside_dai[] = {
 		.cpu_dai_name = "wm8996-aif2",
 		.codec_dai_name = "wm1250-ev1",
 		.codec_name = "wm1250-ev1.1-0027",
-		.ops = &speyside_ops,
+		.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
+				| SND_SOC_DAIFMT_CBM_CFM,
 		.ignore_suspend = 1,
 	},
 };
diff --git a/sound/soc/sh/Kconfig b/sound/soc/sh/Kconfig
index d8e06a607a22..6bcb1164d599 100644
--- a/sound/soc/sh/Kconfig
+++ b/sound/soc/sh/Kconfig
@@ -22,6 +22,7 @@ config SND_SOC_SH4_SSI
 
 config SND_SOC_SH4_FSI
 	tristate "SH4 FSI support"
+	select SND_SIMPLE_CARD
 	help
 	  This option enables FSI sound support
 
@@ -46,29 +47,6 @@ config SND_SH7760_AC97
 	  This option enables generic sound support for the first
 	  AC97 unit of the SH7760.
 
-config SND_FSI_AK4642
-	tristate "FSI-AK4642 sound support"
-	depends on SND_SOC_SH4_FSI && I2C
-	select SND_SOC_AK4642
-	help
-	  This option enables generic sound support for the
-	  FSI - AK4642 unit
-
-config SND_FSI_DA7210
-	tristate "FSI-DA7210 sound support"
-	depends on SND_SOC_SH4_FSI && I2C
-	select SND_SOC_DA7210
-	help
-	  This option enables generic sound support for the
-	  FSI - DA7210 unit
-
-config SND_FSI_HDMI
-	tristate "FSI-HDMI sound support"
-	depends on SND_SOC_SH4_FSI && FB_SH_MOBILE_HDMI
-	help
-	  This option enables generic sound support for the
-	  FSI - HDMI unit
-
 config SND_SIU_MIGOR
 	tristate "SIU sound support on Migo-R"
 	depends on SH_MIGOR
diff --git a/sound/soc/sh/Makefile b/sound/soc/sh/Makefile
index 94476d4c0fd5..849b387d17d9 100644
--- a/sound/soc/sh/Makefile
+++ b/sound/soc/sh/Makefile
@@ -14,13 +14,7 @@ obj-$(CONFIG_SND_SOC_SH4_SIU)	+= snd-soc-siu.o
 
 ## boards
 snd-soc-sh7760-ac97-objs	:= sh7760-ac97.o
-snd-soc-fsi-ak4642-objs		:= fsi-ak4642.o
-snd-soc-fsi-da7210-objs		:= fsi-da7210.o
-snd-soc-fsi-hdmi-objs		:= fsi-hdmi.o
 snd-soc-migor-objs		:= migor.o
 
 obj-$(CONFIG_SND_SH7760_AC97)	+= snd-soc-sh7760-ac97.o
-obj-$(CONFIG_SND_FSI_AK4642)	+= snd-soc-fsi-ak4642.o
-obj-$(CONFIG_SND_FSI_DA7210)	+= snd-soc-fsi-da7210.o
-obj-$(CONFIG_SND_FSI_HDMI)	+= snd-soc-fsi-hdmi.o
 obj-$(CONFIG_SND_SIU_MIGOR)	+= snd-soc-migor.o
diff --git a/sound/soc/sh/fsi-ak4642.c b/sound/soc/sh/fsi-ak4642.c
deleted file mode 100644
index 97f540aabbdd..000000000000
--- a/sound/soc/sh/fsi-ak4642.c
+++ /dev/null
@@ -1,108 +0,0 @@
-/*
- * FSI-AK464x sound support for ms7724se
- *
- * Copyright (C) 2009 Renesas Solutions Corp.
- * Kuninori Morimoto <morimoto.kuninori@renesas.com>
- *
- * This file is subject to the terms and conditions of the GNU General Public
- * License.  See the file "COPYING" in the main directory of this archive
- * for more details.
- */
-
-#include <linux/platform_device.h>
-#include <linux/module.h>
-#include <sound/sh_fsi.h>
-
-struct fsi_ak4642_data {
-	const char *name;
-	const char *card;
-	const char *cpu_dai;
-	const char *codec;
-	const char *platform;
-	int id;
-};
-
-static int fsi_ak4642_dai_init(struct snd_soc_pcm_runtime *rtd)
-{
-	struct snd_soc_dai *codec = rtd->codec_dai;
-	struct snd_soc_dai *cpu = rtd->cpu_dai;
-	int ret;
-
-	ret = snd_soc_dai_set_fmt(codec, SND_SOC_DAIFMT_LEFT_J |
-					 SND_SOC_DAIFMT_CBM_CFM);
-	if (ret < 0)
-		return ret;
-
-	ret = snd_soc_dai_set_sysclk(codec, 0, 11289600, 0);
-	if (ret < 0)
-		return ret;
-
-	ret = snd_soc_dai_set_fmt(cpu, SND_SOC_DAIFMT_LEFT_J |
-				       SND_SOC_DAIFMT_CBS_CFS);
-
-	return ret;
-}
-
-static struct snd_soc_dai_link fsi_dai_link = {
-	.codec_dai_name	= "ak4642-hifi",
-	.init		= fsi_ak4642_dai_init,
-};
-
-static struct snd_soc_card fsi_soc_card  = {
-	.owner		= THIS_MODULE,
-	.dai_link	= &fsi_dai_link,
-	.num_links	= 1,
-};
-
-static struct platform_device *fsi_snd_device;
-
-static int fsi_ak4642_probe(struct platform_device *pdev)
-{
-	int ret = -ENOMEM;
-	struct fsi_ak4642_info *pinfo = pdev->dev.platform_data;
-
-	if (!pinfo) {
-		dev_err(&pdev->dev, "no info for fsi ak4642\n");
-		goto out;
-	}
-
-	fsi_snd_device = platform_device_alloc("soc-audio", pinfo->id);
-	if (!fsi_snd_device)
-		goto out;
-
-	fsi_dai_link.name		= pinfo->name;
-	fsi_dai_link.stream_name	= pinfo->name;
-	fsi_dai_link.cpu_dai_name	= pinfo->cpu_dai;
-	fsi_dai_link.platform_name	= pinfo->platform;
-	fsi_dai_link.codec_name		= pinfo->codec;
-	fsi_soc_card.name		= pinfo->card;
-
-	platform_set_drvdata(fsi_snd_device, &fsi_soc_card);
-	ret = platform_device_add(fsi_snd_device);
-
-	if (ret)
-		platform_device_put(fsi_snd_device);
-
-out:
-	return ret;
-}
-
-static int fsi_ak4642_remove(struct platform_device *pdev)
-{
-	platform_device_unregister(fsi_snd_device);
-	return 0;
-}
-
-static struct platform_driver fsi_ak4642 = {
-	.driver = {
-		.name	= "fsi-ak4642-audio",
-	},
-	.probe		= fsi_ak4642_probe,
-	.remove		= fsi_ak4642_remove,
-};
-
-module_platform_driver(fsi_ak4642);
-
-MODULE_LICENSE("GPL");
-MODULE_DESCRIPTION("Generic SH4 FSI-AK4642 sound card");
-MODULE_AUTHOR("Kuninori Morimoto <morimoto.kuninori@renesas.com>");
diff --git a/sound/soc/sh/fsi-da7210.c b/sound/soc/sh/fsi-da7210.c
deleted file mode 100644
index 1dd3354c7411..000000000000
--- a/sound/soc/sh/fsi-da7210.c
+++ /dev/null
@@ -1,81 +0,0 @@
-/*
- * fsi-da7210.c
- *
- * Copyright (C) 2009 Renesas Solutions Corp.
- * Kuninori Morimoto <morimoto.kuninori@renesas.com>
- *
- *  This program is free software; you can redistribute  it and/or modify it
- *  under  the terms of  the GNU General  Public License as published by the
- *  Free Software Foundation;  either version 2 of the  License, or (at your
- *  option) any later version.
- */
-
-#include <linux/platform_device.h>
-#include <linux/module.h>
-#include <sound/sh_fsi.h>
-
-static int fsi_da7210_init(struct snd_soc_pcm_runtime *rtd)
-{
-	struct snd_soc_dai *codec = rtd->codec_dai;
-	struct snd_soc_dai *cpu = rtd->cpu_dai;
-	int ret;
-
-	ret = snd_soc_dai_set_fmt(codec,
-				   SND_SOC_DAIFMT_I2S |
-				   SND_SOC_DAIFMT_CBM_CFM);
-	if (ret < 0)
-		return ret;
-
-	ret = snd_soc_dai_set_fmt(cpu, SND_SOC_DAIFMT_I2S |
-				       SND_SOC_DAIFMT_CBS_CFS);
-
-	return ret;
-}
-
-static struct snd_soc_dai_link fsi_da7210_dai = {
-	.name		= "DA7210",
-	.stream_name	= "DA7210",
-	.cpu_dai_name	= "fsib-dai", /* FSI B */
-	.codec_dai_name	= "da7210-hifi",
-	.platform_name	= "sh_fsi.0",
-	.codec_name	= "da7210-codec.0-001a",
-	.init		= fsi_da7210_init,
-};
-
-static struct snd_soc_card fsi_soc_card = {
-	.name		= "FSI-DA7210",
-	.owner		= THIS_MODULE,
-	.dai_link	= &fsi_da7210_dai,
-	.num_links	= 1,
-};
-
-static struct platform_device *fsi_da7210_snd_device;
-
-static int __init fsi_da7210_sound_init(void)
-{
-	int ret;
-
-	fsi_da7210_snd_device = platform_device_alloc("soc-audio", FSI_PORT_B);
-	if (!fsi_da7210_snd_device)
-		return -ENOMEM;
-
-	platform_set_drvdata(fsi_da7210_snd_device, &fsi_soc_card);
-	ret = platform_device_add(fsi_da7210_snd_device);
-	if (ret)
-		platform_device_put(fsi_da7210_snd_device);
-
-	return ret;
-}
-
-static void __exit fsi_da7210_sound_exit(void)
-{
-	platform_device_unregister(fsi_da7210_snd_device);
-}
-
-module_init(fsi_da7210_sound_init);
-module_exit(fsi_da7210_sound_exit);
-
-/* Module information */
-MODULE_DESCRIPTION("ALSA SoC FSI DA2710");
-MODULE_AUTHOR("Kuninori Morimoto <morimoto.kuninori@renesas.com>");
-MODULE_LICENSE("GPL");
diff --git a/sound/soc/sh/fsi-hdmi.c b/sound/soc/sh/fsi-hdmi.c
deleted file mode 100644
index 6e41908323e8..000000000000
--- a/sound/soc/sh/fsi-hdmi.c
+++ /dev/null
@@ -1,118 +0,0 @@
-/*
- * FSI - HDMI sound support
- *
- * Copyright (C) 2010 Renesas Solutions Corp.
- * Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
- *
- * This file is subject to the terms and conditions of the GNU General Public
- * License.  See the file "COPYING" in the main directory of this archive
- * for more details.
- */
-
-#include <linux/platform_device.h>
-#include <linux/module.h>
-#include <sound/sh_fsi.h>
-
-struct fsi_hdmi_data {
-	const char *cpu_dai;
-	const char *card;
-	int id;
-};
-
-static int fsi_hdmi_dai_init(struct snd_soc_pcm_runtime *rtd)
-{
-	struct snd_soc_dai *cpu = rtd->cpu_dai;
-	int ret;
-
-	ret = snd_soc_dai_set_fmt(cpu, SND_SOC_DAIFMT_CBM_CFM);
-
-	return ret;
-}
-
-static struct snd_soc_dai_link fsi_dai_link = {
-	.name		= "HDMI",
-	.stream_name	= "HDMI",
-	.codec_dai_name	= "sh_mobile_hdmi-hifi",
-	.platform_name	= "sh_fsi2",
-	.codec_name	= "sh-mobile-hdmi",
-	.init		= fsi_hdmi_dai_init,
-};
-
-static struct snd_soc_card fsi_soc_card  = {
-	.owner		= THIS_MODULE,
-	.dai_link	= &fsi_dai_link,
-	.num_links	= 1,
-};
-
-static struct platform_device *fsi_snd_device;
-
-static int fsi_hdmi_probe(struct platform_device *pdev)
-{
-	int ret = -ENOMEM;
-	const struct platform_device_id	*id_entry;
-	struct fsi_hdmi_data *pdata;
-
-	id_entry = pdev->id_entry;
-	if (!id_entry) {
-		dev_err(&pdev->dev, "unknown fsi hdmi\n");
-		return -ENODEV;
-	}
-
-	pdata = (struct fsi_hdmi_data *)id_entry->driver_data;
-
-	fsi_snd_device = platform_device_alloc("soc-audio", pdata->id);
-	if (!fsi_snd_device)
-		goto out;
-
-	fsi_dai_link.cpu_dai_name	= pdata->cpu_dai;
-	fsi_soc_card.name		= pdata->card;
-
-	platform_set_drvdata(fsi_snd_device, &fsi_soc_card);
-	ret = platform_device_add(fsi_snd_device);
-
-	if (ret)
-		platform_device_put(fsi_snd_device);
-
-out:
-	return ret;
-}
-
-static int fsi_hdmi_remove(struct platform_device *pdev)
-{
-	platform_device_unregister(fsi_snd_device);
-	return 0;
-}
-
-static struct fsi_hdmi_data fsi2_a_hdmi = {
-	.cpu_dai	= "fsia-dai",
-	.card		= "FSI2A-HDMI",
-	.id		= FSI_PORT_A,
-};
-
-static struct fsi_hdmi_data fsi2_b_hdmi = {
-	.cpu_dai	= "fsib-dai",
-	.card		= "FSI2B-HDMI",
-	.id		= FSI_PORT_B,
-};
-
-static struct platform_device_id fsi_id_table[] = {
-	/* FSI 2 */
-	{ "sh_fsi2_a_hdmi",	(kernel_ulong_t)&fsi2_a_hdmi },
-	{ "sh_fsi2_b_hdmi",	(kernel_ulong_t)&fsi2_b_hdmi },
-	{},
-};
-
-static struct platform_driver fsi_hdmi = {
-	.driver = {
-		.name	= "fsi-hdmi-audio",
-	},
-	.probe		= fsi_hdmi_probe,
-	.remove		= fsi_hdmi_remove,
-	.id_table	= fsi_id_table,
-};
-
-module_platform_driver(fsi_hdmi);
-
-MODULE_LICENSE("GPL");
-MODULE_DESCRIPTION("Generic SH4 FSI-HDMI sound card");
-MODULE_AUTHOR("Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>");
diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c
index 74ed2dffbffd..7cee22515d9d 100644
--- a/sound/soc/sh/fsi.c
+++ b/sound/soc/sh/fsi.c
@@ -131,6 +131,25 @@
 
 typedef int (*set_rate_func)(struct device *dev, int rate, int enable);
 
+/*
+ * bus options
+ *
+ * 0x000000BA
+ *
+ * A : sample widtht 16bit setting
+ * B : sample widtht 24bit setting
+ */
+
+#define SHIFT_16DATA		0
+#define SHIFT_24DATA		4
+
+#define PACKAGE_24BITBUS_BACK		0
+#define PACKAGE_24BITBUS_FRONT		1
+#define PACKAGE_16BITBUS_STREAM		2
+
+#define BUSOP_SET(s, a)	((a) << SHIFT_ ## s ## DATA)
+#define BUSOP_GET(s, a)	(((a) >> SHIFT_ ## s ## DATA) & 0xF)
+
 /*
  * FSI driver use below type name for variable
  *
@@ -188,6 +207,11 @@ struct fsi_stream {
 	int uerr_num;
 	int oerr_num;
 
+	/*
+	 * bus options
+	 */
+	u32 bus_option;
+
 	/*
 	 * thse are initialized by fsi_handler_init()
 	 */
@@ -211,8 +235,7 @@ struct fsi_priv {
 	struct fsi_stream playback;
 	struct fsi_stream capture;
 
-	u32 do_fmt;
-	u32 di_fmt;
+	u32 fmt;
 
 	int chan_num:16;
 	int clk_master:1;
@@ -321,6 +344,10 @@ static void _fsi_master_mask_set(struct fsi_master *master,
 /*
  *		basic function
  */
+static int fsi_version(struct fsi_master *master)
+{
+	return master->core->ver;
+}
 
 static struct fsi_master *fsi_get_master(struct fsi_priv *fsi)
 {
@@ -495,6 +522,7 @@ static void fsi_stream_init(struct fsi_priv *fsi,
 	io->period_samples	= fsi_frame2sample(fsi, runtime->period_size);
 	io->period_pos		= 0;
 	io->sample_width	= samples_to_bytes(runtime, 1);
+	io->bus_option		= 0;
 	io->oerr_num	= -1; /* ignore 1st err */
 	io->uerr_num	= -1; /* ignore 1st err */
 	fsi_stream_handler_call(io, init, fsi, io);
@@ -522,6 +550,7 @@ static void fsi_stream_quit(struct fsi_priv *fsi, struct fsi_stream *io)
 	io->period_samples	= 0;
 	io->period_pos		= 0;
 	io->sample_width	= 0;
+	io->bus_option		= 0;
 	io->oerr_num	= 0;
 	io->uerr_num	= 0;
 	spin_unlock_irqrestore(&master->lock, flags);
@@ -580,6 +609,53 @@ static int fsi_stream_remove(struct fsi_priv *fsi)
 	return 0;
 }
 
+/*
+ *	format/bus/dma setting
+ */
+static void fsi_format_bus_setup(struct fsi_priv *fsi, struct fsi_stream *io,
+				 u32 bus, struct device *dev)
+{
+	struct fsi_master *master = fsi_get_master(fsi);
+	int is_play = fsi_stream_is_play(fsi, io);
+	u32 fmt = fsi->fmt;
+
+	if (fsi_version(master) >= 2) {
+		u32 dma = 0;
+
+		/*
+		 * FSI2 needs DMA/Bus setting
+		 */
+		switch (bus) {
+		case PACKAGE_24BITBUS_FRONT:
+			fmt |= CR_BWS_24;
+			dma |= VDMD_FRONT;
+			dev_dbg(dev, "24bit bus / package in front\n");
+			break;
+		case PACKAGE_16BITBUS_STREAM:
+			fmt |= CR_BWS_16;
+			dma |= VDMD_STREAM;
+			dev_dbg(dev, "16bit bus / stream mode\n");
+			break;
+		case PACKAGE_24BITBUS_BACK:
+		default:
+			fmt |= CR_BWS_24;
+			dma |= VDMD_BACK;
+			dev_dbg(dev, "24bit bus / package in back\n");
+			break;
+		}
+
+		if (is_play)
+			fsi_reg_write(fsi, OUT_DMAC,	dma);
+		else
+			fsi_reg_write(fsi, IN_DMAC,	dma);
+	}
+
+	if (is_play)
+		fsi_reg_write(fsi, DO_FMT, fmt);
+	else
+		fsi_reg_write(fsi, DI_FMT, fmt);
+}
+
 /*
  *		irq function
  */
@@ -629,11 +705,6 @@ static void fsi_spdif_clk_ctrl(struct fsi_priv *fsi, int enable)
 	struct fsi_master *master = fsi_get_master(fsi);
 	u32 mask, val;
 
-	if (master->core->ver < 2) {
-		pr_err("fsi: register access err (%s)\n", __func__);
-		return;
-	}
-
 	mask = BP | SE;
 	val = enable ? mask : 0;
 
@@ -648,9 +719,7 @@ static void fsi_spdif_clk_ctrl(struct fsi_priv *fsi, int enable)
 static int fsi_set_master_clk(struct device *dev, struct fsi_priv *fsi,
 			      long rate, int enable)
 {
-	struct fsi_master *master = fsi_get_master(fsi);
 	set_rate_func set_rate = fsi_get_info_set_rate(fsi);
-	int fsi_ver = master->core->ver;
 	int ret;
 
 	if (!set_rate)
@@ -682,10 +751,7 @@ static int fsi_set_master_clk(struct device *dev, struct fsi_priv *fsi,
 			data |= (0x3 << 12);
 			break;
 		case SH_FSI_ACKMD_32:
-			if (fsi_ver < 2)
-				dev_err(dev, "unsupported ACKMD\n");
-			else
-				data |= (0x4 << 12);
+			data |= (0x4 << 12);
 			break;
 		}
 
@@ -708,10 +774,7 @@ static int fsi_set_master_clk(struct device *dev, struct fsi_priv *fsi,
 			data |= (0x4 << 8);
 			break;
 		case SH_FSI_BPFMD_16:
-			if (fsi_ver < 2)
-				dev_err(dev, "unsupported ACKMD\n");
-			else
-				data |= (0x7 << 8);
+			data |= (0x7 << 8);
 			break;
 		}
 
@@ -728,11 +791,26 @@ static int fsi_set_master_clk(struct device *dev, struct fsi_priv *fsi,
  */
 static void fsi_pio_push16(struct fsi_priv *fsi, u8 *_buf, int samples)
 {
-	u16 *buf = (u16 *)_buf;
+	u32 enable_stream = fsi_get_info_flags(fsi) & SH_FSI_ENABLE_STREAM_MODE;
 	int i;
 
-	for (i = 0; i < samples; i++)
-		fsi_reg_write(fsi, DODT, ((u32)*(buf + i) << 8));
+	if (enable_stream) {
+		/*
+		 * stream mode
+		 * see
+		 *	fsi_pio_push_init()
+		 */
+		u32 *buf = (u32 *)_buf;
+
+		for (i = 0; i < samples / 2; i++)
+			fsi_reg_write(fsi, DODT, buf[i]);
+	} else {
+		/* normal mode */
+		u16 *buf = (u16 *)_buf;
+
+		for (i = 0; i < samples; i++)
+			fsi_reg_write(fsi, DODT, ((u32)*(buf + i) << 8));
+	}
 }
 
 static void fsi_pio_pop16(struct fsi_priv *fsi, u8 *_buf, int samples)
@@ -872,12 +950,44 @@ static void fsi_pio_start_stop(struct fsi_priv *fsi, struct fsi_stream *io,
 		fsi_master_mask_set(master, CLK_RST, clk, (enable) ? clk : 0);
 }
 
+static int fsi_pio_push_init(struct fsi_priv *fsi, struct fsi_stream *io)
+{
+	u32 enable_stream = fsi_get_info_flags(fsi) & SH_FSI_ENABLE_STREAM_MODE;
+
+	/*
+	 * we can use 16bit stream mode
+	 * when "playback" and "16bit data"
+	 * and platform allows "stream mode"
+	 * see
+	 *	fsi_pio_push16()
+	 */
+	if (enable_stream)
+		io->bus_option = BUSOP_SET(24, PACKAGE_24BITBUS_BACK) |
+				 BUSOP_SET(16, PACKAGE_16BITBUS_STREAM);
+	else
+		io->bus_option = BUSOP_SET(24, PACKAGE_24BITBUS_BACK) |
+				 BUSOP_SET(16, PACKAGE_24BITBUS_BACK);
+	return 0;
+}
+
+static int fsi_pio_pop_init(struct fsi_priv *fsi, struct fsi_stream *io)
+{
+	/*
+	 * always 24bit bus, package back when "capture"
+	 */
+	io->bus_option = BUSOP_SET(24, PACKAGE_24BITBUS_BACK) |
+			 BUSOP_SET(16, PACKAGE_24BITBUS_BACK);
+	return 0;
+}
+
 static struct fsi_stream_handler fsi_pio_push_handler = {
+	.init		= fsi_pio_push_init,
 	.transfer	= fsi_pio_push,
 	.start_stop	= fsi_pio_start_stop,
 };
 
 static struct fsi_stream_handler fsi_pio_pop_handler = {
+	.init		= fsi_pio_pop_init,
 	.transfer	= fsi_pio_pop,
 	.start_stop	= fsi_pio_start_stop,
 };
@@ -919,6 +1029,13 @@ static int fsi_dma_init(struct fsi_priv *fsi, struct fsi_stream *io)
 	enum dma_data_direction dir = fsi_stream_is_play(fsi, io) ?
 				DMA_TO_DEVICE : DMA_FROM_DEVICE;
 
+	/*
+	 * 24bit data : 24bit bus / package in back
+	 * 16bit data : 16bit bus / stream mode
+	 */
+	io->bus_option = BUSOP_SET(24, PACKAGE_24BITBUS_BACK) |
+			 BUSOP_SET(16, PACKAGE_16BITBUS_STREAM);
+
 	io->dma = dma_map_single(dai->dev, runtime->dma_area,
 				 snd_pcm_lib_buffer_bytes(io->substream), dir);
 	return 0;
@@ -1055,25 +1172,9 @@ static int fsi_dma_transfer(struct fsi_priv *fsi, struct fsi_stream *io)
 static void fsi_dma_push_start_stop(struct fsi_priv *fsi, struct fsi_stream *io,
 				 int start)
 {
-	u32 bws;
-	u32 dma;
+	u32 enable = start ? DMA_ON : 0;
 
-	switch (io->sample_width * start) {
-	case 2:
-		bws = CR_BWS_16;
-		dma = VDMD_STREAM | DMA_ON;
-		break;
-	case 4:
-		bws = CR_BWS_24;
-		dma = VDMD_BACK | DMA_ON;
-		break;
-	default:
-		bws = 0;
-		dma = 0;
-	}
-
-	fsi_reg_mask_set(fsi, DO_FMT, CR_BWS_MASK, bws);
-	fsi_reg_write(fsi, OUT_DMAC, dma);
+	fsi_reg_mask_set(fsi, OUT_DMAC, DMA_ON, enable);
 }
 
 static int fsi_dma_probe(struct fsi_priv *fsi, struct fsi_stream *io)
@@ -1176,8 +1277,6 @@ static int fsi_hw_startup(struct fsi_priv *fsi,
 			  struct fsi_stream *io,
 			  struct device *dev)
 {
-	struct fsi_master *master = fsi_get_master(fsi);
-	int fsi_ver = master->core->ver;
 	u32 flags = fsi_get_info_flags(fsi);
 	u32 data = 0;
 
@@ -1200,10 +1299,6 @@ static int fsi_hw_startup(struct fsi_priv *fsi,
 
 	fsi_reg_write(fsi, CKG2, data);
 
-	/* set format */
-	fsi_reg_write(fsi, DO_FMT, fsi->do_fmt);
-	fsi_reg_write(fsi, DI_FMT, fsi->di_fmt);
-
 	/* spdif ? */
 	if (fsi_is_spdif(fsi)) {
 		fsi_spdif_clk_ctrl(fsi, 1);
@@ -1211,15 +1306,18 @@ static int fsi_hw_startup(struct fsi_priv *fsi,
 	}
 
 	/*
-	 * FIXME
-	 *
-	 * FSI driver assumed that data package is in-back.
-	 * FSI2 chip can select it.
+	 * get bus settings
 	 */
-	if (fsi_ver >= 2) {
-		fsi_reg_write(fsi, OUT_DMAC,	(1 << 4));
-		fsi_reg_write(fsi, IN_DMAC,	(1 << 4));
+	data = 0;
+	switch (io->sample_width) {
+	case 2:
+		data = BUSOP_GET(16, io->bus_option);
+		break;
+	case 4:
+		data = BUSOP_GET(24, io->bus_option);
+		break;
 	}
+	fsi_format_bus_setup(fsi, io, data, dev);
 
 	/* irq clear */
 	fsi_irq_disable(fsi, io);
@@ -1243,7 +1341,9 @@ static int fsi_dai_startup(struct snd_pcm_substream *substream,
 {
 	struct fsi_priv *fsi = fsi_get_priv(substream);
 
-	return fsi_hw_startup(fsi, fsi_stream_get(fsi, substream), dai->dev);
+	fsi->rate = 0;
+
+	return 0;
 }
 
 static void fsi_dai_shutdown(struct snd_pcm_substream *substream,
@@ -1251,7 +1351,6 @@ static void fsi_dai_shutdown(struct snd_pcm_substream *substream,
 {
 	struct fsi_priv *fsi = fsi_get_priv(substream);
 
-	fsi_hw_shutdown(fsi, dai->dev);
 	fsi->rate = 0;
 }
 
@@ -1265,11 +1364,13 @@ static int fsi_dai_trigger(struct snd_pcm_substream *substream, int cmd,
 	switch (cmd) {
 	case SNDRV_PCM_TRIGGER_START:
 		fsi_stream_init(fsi, io, substream);
+		fsi_hw_startup(fsi, io, dai->dev);
 		ret = fsi_stream_transfer(io);
 		if (0 == ret)
 			fsi_stream_start(fsi, io);
 		break;
 	case SNDRV_PCM_TRIGGER_STOP:
+		fsi_hw_shutdown(fsi, dai->dev);
 		fsi_stream_stop(fsi, io);
 		fsi_stream_quit(fsi, io);
 		break;
@@ -1280,42 +1381,33 @@ static int fsi_dai_trigger(struct snd_pcm_substream *substream, int cmd,
 
 static int fsi_set_fmt_dai(struct fsi_priv *fsi, unsigned int fmt)
 {
-	u32 data = 0;
-
 	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
 	case SND_SOC_DAIFMT_I2S:
-		data = CR_I2S;
+		fsi->fmt = CR_I2S;
 		fsi->chan_num = 2;
 		break;
 	case SND_SOC_DAIFMT_LEFT_J:
-		data = CR_PCM;
+		fsi->fmt = CR_PCM;
 		fsi->chan_num = 2;
 		break;
 	default:
 		return -EINVAL;
 	}
 
-	fsi->do_fmt = data;
-	fsi->di_fmt = data;
-
 	return 0;
 }
 
 static int fsi_set_fmt_spdif(struct fsi_priv *fsi)
 {
 	struct fsi_master *master = fsi_get_master(fsi);
-	u32 data = 0;
 
-	if (master->core->ver < 2)
+	if (fsi_version(master) < 2)
 		return -EINVAL;
 
-	data = CR_BWS_16 | CR_DTMD_SPDIF_PCM | CR_PCM;
+	fsi->fmt = CR_DTMD_SPDIF_PCM | CR_PCM;
 	fsi->chan_num = 2;
 	fsi->spdif = 1;
 
-	fsi->do_fmt = data;
-	fsi->di_fmt = data;
-
 	return 0;
 }
 
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index c88d9741b9e7..b37ee8077ed1 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -39,6 +39,7 @@
 #include <sound/pcm.h>
 #include <sound/pcm_params.h>
 #include <sound/soc.h>
+#include <sound/soc-dpcm.h>
 #include <sound/initval.h>
 
 #define CREATE_TRACE_POINTS
@@ -54,7 +55,6 @@ EXPORT_SYMBOL_GPL(snd_soc_debugfs_root);
 #endif
 
 static DEFINE_MUTEX(client_mutex);
-static LIST_HEAD(card_list);
 static LIST_HEAD(dai_list);
 static LIST_HEAD(platform_list);
 static LIST_HEAD(codec_list);
@@ -465,6 +465,35 @@ static inline void soc_cleanup_card_debugfs(struct snd_soc_card *card)
 }
 #endif
 
+struct snd_pcm_substream *snd_soc_get_dai_substream(struct snd_soc_card *card,
+		const char *dai_link, int stream)
+{
+	int i;
+
+	for (i = 0; i < card->num_links; i++) {
+		if (card->rtd[i].dai_link->no_pcm &&
+			!strcmp(card->rtd[i].dai_link->name, dai_link))
+			return card->rtd[i].pcm->streams[stream].substream;
+	}
+	dev_dbg(card->dev, "failed to find dai link %s\n", dai_link);
+	return NULL;
+}
+EXPORT_SYMBOL_GPL(snd_soc_get_dai_substream);
+
+struct snd_soc_pcm_runtime *snd_soc_get_pcm_runtime(struct snd_soc_card *card,
+		const char *dai_link)
+{
+	int i;
+
+	for (i = 0; i < card->num_links; i++) {
+		if (!strcmp(card->rtd[i].dai_link->name, dai_link))
+			return &card->rtd[i];
+	}
+	dev_dbg(card->dev, "failed to find rtd %s\n", dai_link);
+	return NULL;
+}
+EXPORT_SYMBOL_GPL(snd_soc_get_pcm_runtime);
+
 #ifdef CONFIG_SND_SOC_AC97_BUS
 /* unregister ac97 codec */
 static int soc_ac97_dev_unregister(struct snd_soc_codec *codec)
@@ -567,19 +596,16 @@ int snd_soc_suspend(struct device *dev)
 	}
 
 	for (i = 0; i < card->num_rtd; i++) {
-		struct snd_soc_dai *codec_dai = card->rtd[i].codec_dai;
 
 		if (card->rtd[i].dai_link->ignore_suspend)
 			continue;
 
 		snd_soc_dapm_stream_event(&card->rtd[i],
 					  SNDRV_PCM_STREAM_PLAYBACK,
-					  codec_dai,
 					  SND_SOC_DAPM_STREAM_SUSPEND);
 
 		snd_soc_dapm_stream_event(&card->rtd[i],
 					  SNDRV_PCM_STREAM_CAPTURE,
-					  codec_dai,
 					  SND_SOC_DAPM_STREAM_SUSPEND);
 	}
 
@@ -683,17 +709,16 @@ static void soc_resume_deferred(struct work_struct *work)
 	}
 
 	for (i = 0; i < card->num_rtd; i++) {
-		struct snd_soc_dai *codec_dai = card->rtd[i].codec_dai;
 
 		if (card->rtd[i].dai_link->ignore_suspend)
 			continue;
 
 		snd_soc_dapm_stream_event(&card->rtd[i],
-					  SNDRV_PCM_STREAM_PLAYBACK, codec_dai,
+					  SNDRV_PCM_STREAM_PLAYBACK,
 					  SND_SOC_DAPM_STREAM_RESUME);
 
 		snd_soc_dapm_stream_event(&card->rtd[i],
-					  SNDRV_PCM_STREAM_CAPTURE, codec_dai,
+					  SNDRV_PCM_STREAM_CAPTURE,
 					  SND_SOC_DAPM_STREAM_RESUME);
 	}
 
@@ -783,15 +808,9 @@ static int soc_bind_dai_link(struct snd_soc_card *card, int num)
 	struct snd_soc_dai *codec_dai, *cpu_dai;
 	const char *platform_name;
 
-	if (rtd->complete)
-		return 1;
 	dev_dbg(card->dev, "binding %s at idx %d\n", dai_link->name, num);
 
-	/* do we already have the CPU DAI for this link ? */
-	if (rtd->cpu_dai) {
-		goto find_codec;
-	}
-	/* no, then find CPU DAI from registered DAIs*/
+	/* Find CPU DAI from registered DAIs*/
 	list_for_each_entry(cpu_dai, &dai_list, list) {
 		if (dai_link->cpu_dai_of_node) {
 			if (cpu_dai->dev->of_node != dai_link->cpu_dai_of_node)
@@ -802,18 +821,15 @@ static int soc_bind_dai_link(struct snd_soc_card *card, int num)
 		}
 
 		rtd->cpu_dai = cpu_dai;
-		goto find_codec;
 	}
-	dev_dbg(card->dev, "CPU DAI %s not registered\n",
+
+	if (!rtd->cpu_dai) {
+		dev_dbg(card->dev, "CPU DAI %s not registered\n",
 			dai_link->cpu_dai_name);
-
-find_codec:
-	/* do we already have the CODEC for this link ? */
-	if (rtd->codec) {
-		goto find_platform;
+		return -EPROBE_DEFER;
 	}
 
-	/* no, then find CODEC from registered CODECs*/
+	/* Find CODEC from registered CODECs */
 	list_for_each_entry(codec, &codec_list, list) {
 		if (dai_link->codec_of_node) {
 			if (codec->dev->of_node != dai_link->codec_of_node)
@@ -835,28 +851,28 @@ find_codec:
 					dai_link->codec_dai_name)) {
 
 				rtd->codec_dai = codec_dai;
-				goto find_platform;
 			}
 		}
-		dev_dbg(card->dev, "CODEC DAI %s not registered\n",
+
+		if (!rtd->codec_dai) {
+			dev_dbg(card->dev, "CODEC DAI %s not registered\n",
 				dai_link->codec_dai_name);
-
-		goto find_platform;
+			return -EPROBE_DEFER;
+		}
 	}
-	dev_dbg(card->dev, "CODEC %s not registered\n",
-			dai_link->codec_name);
 
-find_platform:
-	/* do we need a platform? */
-	if (rtd->platform)
-		goto out;
+	if (!rtd->codec) {
+		dev_dbg(card->dev, "CODEC %s not registered\n",
+			dai_link->codec_name);
+		return -EPROBE_DEFER;
+	}
 
 	/* if there's no platform we match on the empty platform */
 	platform_name = dai_link->platform_name;
 	if (!platform_name && !dai_link->platform_of_node)
 		platform_name = "snd-soc-dummy";
 
-	/* no, then find one from the set of registered platforms */
+	/* find one from the set of registered platforms */
 	list_for_each_entry(platform, &platform_list, list) {
 		if (dai_link->platform_of_node) {
 			if (platform->dev->of_node !=
@@ -868,20 +884,16 @@ find_platform:
 		}
 
 		rtd->platform = platform;
-		goto out;
 	}
-
-	dev_dbg(card->dev, "platform %s not registered\n",
+	if (!rtd->platform) {
+		dev_dbg(card->dev, "platform %s not registered\n",
 			dai_link->platform_name);
-	return 0;
-
-out:
-	/* mark rtd as complete if we found all 4 of our client devices */
-	if (rtd->codec && rtd->codec_dai && rtd->platform && rtd->cpu_dai) {
-		rtd->complete = 1;
-		card->num_rtd++;
+		return -EPROBE_DEFER;
 	}
-	return 1;
+
+	card->num_rtd++;
+
+	return 0;
 }
 
 static void soc_remove_codec(struct snd_soc_codec *codec)
@@ -1068,6 +1080,7 @@ static int soc_probe_platform(struct snd_soc_card *card,
 {
 	int ret = 0;
 	const struct snd_soc_platform_driver *driver = platform->driver;
+	struct snd_soc_dai *dai;
 
 	platform->card = card;
 	platform->dapm.card = card;
@@ -1081,6 +1094,14 @@ static int soc_probe_platform(struct snd_soc_card *card,
 		snd_soc_dapm_new_controls(&platform->dapm,
 			driver->dapm_widgets, driver->num_dapm_widgets);
 
+	/* Create DAPM widgets for each DAI stream */
+	list_for_each_entry(dai, &dai_list, list) {
+		if (dai->dev != platform->dev)
+			continue;
+
+		snd_soc_dapm_new_dai_widgets(&platform->dapm, dai);
+	}
+
 	platform->dapm.idle_bias_off = 1;
 
 	if (driver->probe) {
@@ -1170,6 +1191,10 @@ static int soc_post_component_init(struct snd_soc_card *card,
 	rtd->dev->init_name = name;
 	dev_set_drvdata(rtd->dev, rtd);
 	mutex_init(&rtd->pcm_mutex);
+	INIT_LIST_HEAD(&rtd->dpcm[SNDRV_PCM_STREAM_PLAYBACK].be_clients);
+	INIT_LIST_HEAD(&rtd->dpcm[SNDRV_PCM_STREAM_CAPTURE].be_clients);
+	INIT_LIST_HEAD(&rtd->dpcm[SNDRV_PCM_STREAM_PLAYBACK].fe_clients);
+	INIT_LIST_HEAD(&rtd->dpcm[SNDRV_PCM_STREAM_CAPTURE].fe_clients);
 	ret = device_add(rtd->dev);
 	if (ret < 0) {
 		dev_err(card->dev,
@@ -1191,6 +1216,17 @@ static int soc_post_component_init(struct snd_soc_card *card,
 		dev_err(codec->dev,
 			"asoc: failed to add codec sysfs files: %d\n", ret);
 
+#ifdef CONFIG_DEBUG_FS
+	/* add DPCM sysfs entries */
+	if (!dailess && !dai_link->dynamic)
+		goto out;
+
+	ret = soc_dpcm_debugfs_add(rtd);
+	if (ret < 0)
+		dev_err(rtd->dev, "asoc: failed to add dpcm sysfs entries: %d\n", ret);
+
+out:
+#endif
 	return 0;
 }
 
@@ -1200,14 +1236,15 @@ static int soc_probe_dai_link(struct snd_soc_card *card, int num, int order)
 	struct snd_soc_pcm_runtime *rtd = &card->rtd[num];
 	struct snd_soc_codec *codec = rtd->codec;
 	struct snd_soc_platform *platform = rtd->platform;
-	struct snd_soc_dai *codec_dai = rtd->codec_dai, *cpu_dai = rtd->cpu_dai;
+	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	struct snd_soc_dapm_widget *play_w, *capture_w;
 	int ret;
 
 	dev_dbg(card->dev, "probe %s dai link %d late %d\n",
 			card->name, num, order);
 
 	/* config components */
-	codec_dai->codec = codec;
 	cpu_dai->platform = platform;
 	codec_dai->card = card;
 	cpu_dai->card = card;
@@ -1218,9 +1255,12 @@ static int soc_probe_dai_link(struct snd_soc_card *card, int num, int order)
 	/* probe the cpu_dai */
 	if (!cpu_dai->probed &&
 			cpu_dai->driver->probe_order == order) {
+		cpu_dai->dapm.card = card;
 		if (!try_module_get(cpu_dai->dev->driver->owner))
 			return -ENODEV;
 
+		snd_soc_dapm_new_dai_widgets(&cpu_dai->dapm, cpu_dai);
+
 		if (cpu_dai->driver->probe) {
 			ret = cpu_dai->driver->probe(cpu_dai);
 			if (ret < 0) {
@@ -1279,12 +1319,39 @@ static int soc_probe_dai_link(struct snd_soc_card *card, int num, int order)
 	if (ret < 0)
 		pr_warn("asoc: failed to add pmdown_time sysfs:%d\n", ret);
 
-	/* create the pcm */
-	ret = soc_new_pcm(rtd, num);
-	if (ret < 0) {
-		pr_err("asoc: can't create pcm %s :%d\n",
-				dai_link->stream_name, ret);
-		return ret;
+	if (!dai_link->params) {
+		/* create the pcm */
+		ret = soc_new_pcm(rtd, num);
+		if (ret < 0) {
+			pr_err("asoc: can't create pcm %s :%d\n",
+			       dai_link->stream_name, ret);
+			return ret;
+		}
+	} else {
+		/* link the DAI widgets */
+		play_w = codec_dai->playback_widget;
+		capture_w = cpu_dai->capture_widget;
+		if (play_w && capture_w) {
+			ret = snd_soc_dapm_new_pcm(card, dai_link->params,
+						   capture_w, play_w);
+			if (ret != 0) {
+				dev_err(card->dev, "Can't link %s to %s: %d\n",
+					play_w->name, capture_w->name, ret);
+				return ret;
+			}
+		}
+
+		play_w = cpu_dai->playback_widget;
+		capture_w = codec_dai->capture_widget;
+		if (play_w && capture_w) {
+			ret = snd_soc_dapm_new_pcm(card, dai_link->params,
+						   capture_w, play_w);
+			if (ret != 0) {
+				dev_err(card->dev, "Can't link %s to %s: %d\n",
+					play_w->name, capture_w->name, ret);
+				return ret;
+			}
+		}
 	}
 
 	/* add platform data for AC97 devices */
@@ -1334,6 +1401,20 @@ static void soc_unregister_ac97_dai_link(struct snd_soc_codec *codec)
 }
 #endif
 
+static int soc_check_aux_dev(struct snd_soc_card *card, int num)
+{
+	struct snd_soc_aux_dev *aux_dev = &card->aux_dev[num];
+	struct snd_soc_codec *codec;
+
+	/* find CODEC from registered CODECs*/
+	list_for_each_entry(codec, &codec_list, list) {
+		if (!strcmp(codec->name, aux_dev->codec_name))
+			return 0;
+	}
+
+	return -EPROBE_DEFER;
+}
+
 static int soc_probe_aux_dev(struct snd_soc_card *card, int num)
 {
 	struct snd_soc_aux_dev *aux_dev = &card->aux_dev[num];
@@ -1354,7 +1435,7 @@ static int soc_probe_aux_dev(struct snd_soc_card *card, int num)
 	}
 	/* codec not found */
 	dev_err(card->dev, "asoc: codec %s not found", aux_dev->codec_name);
-	goto out;
+	return -EPROBE_DEFER;
 
 found:
 	ret = soc_probe_codec(card, codec);
@@ -1404,29 +1485,28 @@ static int snd_soc_init_codec_cache(struct snd_soc_codec *codec,
 	return 0;
 }
 
-static void snd_soc_instantiate_card(struct snd_soc_card *card)
+static int snd_soc_instantiate_card(struct snd_soc_card *card)
 {
 	struct snd_soc_codec *codec;
 	struct snd_soc_codec_conf *codec_conf;
 	enum snd_soc_compress_type compress_type;
 	struct snd_soc_dai_link *dai_link;
-	int ret, i, order;
+	int ret, i, order, dai_fmt;
 
-	mutex_lock(&card->mutex);
-
-	if (card->instantiated) {
-		mutex_unlock(&card->mutex);
-		return;
-	}
+	mutex_lock_nested(&card->mutex, SND_SOC_CARD_CLASS_INIT);
 
 	/* bind DAIs */
-	for (i = 0; i < card->num_links; i++)
-		soc_bind_dai_link(card, i);
+	for (i = 0; i < card->num_links; i++) {
+		ret = soc_bind_dai_link(card, i);
+		if (ret != 0)
+			goto base_error;
+	}
 
-	/* bind completed ? */
-	if (card->num_rtd != card->num_links) {
-		mutex_unlock(&card->mutex);
-		return;
+	/* check aux_devs too */
+	for (i = 0; i < card->num_aux_devs; i++) {
+		ret = soc_check_aux_dev(card, i);
+		if (ret != 0)
+			goto base_error;
 	}
 
 	/* initialize the register cache for each available codec */
@@ -1446,10 +1526,8 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card)
 			}
 		}
 		ret = snd_soc_init_codec_cache(codec, compress_type);
-		if (ret < 0) {
-			mutex_unlock(&card->mutex);
-			return;
-		}
+		if (ret < 0)
+			goto base_error;
 	}
 
 	/* card bind complete so register a sound card */
@@ -1458,8 +1536,7 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card)
 	if (ret < 0) {
 		pr_err("asoc: can't create sound card for card %s: %d\n",
 			card->name, ret);
-		mutex_unlock(&card->mutex);
-		return;
+		goto base_error;
 	}
 	card->snd_card->dev = card->dev;
 
@@ -1523,17 +1600,47 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card)
 
 	for (i = 0; i < card->num_links; i++) {
 		dai_link = &card->dai_link[i];
+		dai_fmt = dai_link->dai_fmt;
 
-		if (dai_link->dai_fmt) {
+		if (dai_fmt) {
 			ret = snd_soc_dai_set_fmt(card->rtd[i].codec_dai,
-						  dai_link->dai_fmt);
+						  dai_fmt);
 			if (ret != 0 && ret != -ENOTSUPP)
 				dev_warn(card->rtd[i].codec_dai->dev,
 					 "Failed to set DAI format: %d\n",
 					 ret);
+		}
+
+		/* If this is a regular CPU link there will be a platform */
+		if (dai_fmt &&
+		    (dai_link->platform_name || dai_link->platform_of_node)) {
+			ret = snd_soc_dai_set_fmt(card->rtd[i].cpu_dai,
+						  dai_fmt);
+			if (ret != 0 && ret != -ENOTSUPP)
+				dev_warn(card->rtd[i].cpu_dai->dev,
+					 "Failed to set DAI format: %d\n",
+					 ret);
+		} else if (dai_fmt) {
+			/* Flip the polarity for the "CPU" end */
+			dai_fmt &= ~SND_SOC_DAIFMT_MASTER_MASK;
+			switch (dai_link->dai_fmt &
+				SND_SOC_DAIFMT_MASTER_MASK) {
+			case SND_SOC_DAIFMT_CBM_CFM:
+				dai_fmt |= SND_SOC_DAIFMT_CBS_CFS;
+				break;
+			case SND_SOC_DAIFMT_CBM_CFS:
+				dai_fmt |= SND_SOC_DAIFMT_CBS_CFM;
+				break;
+			case SND_SOC_DAIFMT_CBS_CFM:
+				dai_fmt |= SND_SOC_DAIFMT_CBM_CFS;
+				break;
+			case SND_SOC_DAIFMT_CBS_CFS:
+				dai_fmt |= SND_SOC_DAIFMT_CBM_CFM;
+				break;
+			}
 
 			ret = snd_soc_dai_set_fmt(card->rtd[i].cpu_dai,
-						  dai_link->dai_fmt);
+						  dai_fmt);
 			if (ret != 0 && ret != -ENOTSUPP)
 				dev_warn(card->rtd[i].cpu_dai->dev,
 					 "Failed to set DAI format: %d\n",
@@ -1599,7 +1706,8 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card)
 	card->instantiated = 1;
 	snd_soc_dapm_sync(&card->dapm);
 	mutex_unlock(&card->mutex);
-	return;
+
+	return 0;
 
 probe_aux_dev_err:
 	for (i = 0; i < card->num_aux_devs; i++)
@@ -1614,18 +1722,10 @@ card_probe_error:
 
 	snd_card_free(card->snd_card);
 
+base_error:
 	mutex_unlock(&card->mutex);
-}
 
-/*
- * Attempt to initialise any uninitialised cards.  Must be called with
- * client_mutex.
- */
-static void snd_soc_instantiate_cards(void)
-{
-	struct snd_soc_card *card;
-	list_for_each_entry(card, &card_list, list)
-		snd_soc_instantiate_card(card);
+	return ret;
 }
 
 /* probes a new socdev */
@@ -2526,6 +2626,87 @@ int snd_soc_put_volsw(struct snd_kcontrol *kcontrol,
 }
 EXPORT_SYMBOL_GPL(snd_soc_put_volsw);
 
+/**
+ * snd_soc_get_volsw_sx - single mixer get callback
+ * @kcontrol: mixer control
+ * @ucontrol: control element information
+ *
+ * Callback to get the value of a single mixer control, or a double mixer
+ * control that spans 2 registers.
+ *
+ * Returns 0 for success.
+ */
+int snd_soc_get_volsw_sx(struct snd_kcontrol *kcontrol,
+		      struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct soc_mixer_control *mc =
+	    (struct soc_mixer_control *)kcontrol->private_value;
+
+	unsigned int reg = mc->reg;
+	unsigned int reg2 = mc->rreg;
+	unsigned int shift = mc->shift;
+	unsigned int rshift = mc->rshift;
+	int max = mc->max;
+	int min = mc->min;
+	int mask = (1 << (fls(min + max) - 1)) - 1;
+
+	ucontrol->value.integer.value[0] =
+	    ((snd_soc_read(codec, reg) >> shift) - min) & mask;
+
+	if (snd_soc_volsw_is_stereo(mc))
+		ucontrol->value.integer.value[1] =
+			((snd_soc_read(codec, reg2) >> rshift) - min) & mask;
+
+	return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_get_volsw_sx);
+
+/**
+ * snd_soc_put_volsw_sx - double mixer set callback
+ * @kcontrol: mixer control
+ * @uinfo: control element information
+ *
+ * Callback to set the value of a double mixer control that spans 2 registers.
+ *
+ * Returns 0 for success.
+ */
+int snd_soc_put_volsw_sx(struct snd_kcontrol *kcontrol,
+			 struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct soc_mixer_control *mc =
+	    (struct soc_mixer_control *)kcontrol->private_value;
+
+	unsigned int reg = mc->reg;
+	unsigned int reg2 = mc->rreg;
+	unsigned int shift = mc->shift;
+	unsigned int rshift = mc->rshift;
+	int max = mc->max;
+	int min = mc->min;
+	int mask = (1 << (fls(min + max) - 1)) - 1;
+	int err = 0;
+	unsigned short val, val_mask, val2 = 0;
+
+	val_mask = mask << shift;
+	val = (ucontrol->value.integer.value[0] + min) & mask;
+	val = val << shift;
+
+	if (snd_soc_update_bits_locked(codec, reg, val_mask, val))
+			return err;
+
+	if (snd_soc_volsw_is_stereo(mc)) {
+		val_mask = mask << rshift;
+		val2 = (ucontrol->value.integer.value[1] + min) & mask;
+		val2 = val2 << rshift;
+
+		if (snd_soc_update_bits_locked(codec, reg2, val_mask, val2))
+			return err;
+	}
+	return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_put_volsw_sx);
+
 /**
  * snd_soc_info_volsw_s8 - signed mixer info callback
  * @kcontrol: mixer control
@@ -2647,99 +2828,6 @@ int snd_soc_limit_volume(struct snd_soc_codec *codec,
 }
 EXPORT_SYMBOL_GPL(snd_soc_limit_volume);
 
-/**
- * snd_soc_info_volsw_2r_sx - double with tlv and variable data size
- *  mixer info callback
- * @kcontrol: mixer control
- * @uinfo: control element information
- *
- * Returns 0 for success.
- */
-int snd_soc_info_volsw_2r_sx(struct snd_kcontrol *kcontrol,
-			struct snd_ctl_elem_info *uinfo)
-{
-	struct soc_mixer_control *mc =
-		(struct soc_mixer_control *)kcontrol->private_value;
-	int max = mc->max;
-	int min = mc->min;
-
-	uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
-	uinfo->count = 2;
-	uinfo->value.integer.min = 0;
-	uinfo->value.integer.max = max-min;
-
-	return 0;
-}
-EXPORT_SYMBOL_GPL(snd_soc_info_volsw_2r_sx);
-
-/**
- * snd_soc_get_volsw_2r_sx - double with tlv and variable data size
- *  mixer get callback
- * @kcontrol: mixer control
- * @uinfo: control element information
- *
- * Returns 0 for success.
- */
-int snd_soc_get_volsw_2r_sx(struct snd_kcontrol *kcontrol,
-			struct snd_ctl_elem_value *ucontrol)
-{
-	struct soc_mixer_control *mc =
-		(struct soc_mixer_control *)kcontrol->private_value;
-	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
-	unsigned int mask = (1<<mc->shift)-1;
-	int min = mc->min;
-	int val = snd_soc_read(codec, mc->reg) & mask;
-	int valr = snd_soc_read(codec, mc->rreg) & mask;
-
-	ucontrol->value.integer.value[0] = ((val & 0xff)-min) & mask;
-	ucontrol->value.integer.value[1] = ((valr & 0xff)-min) & mask;
-	return 0;
-}
-EXPORT_SYMBOL_GPL(snd_soc_get_volsw_2r_sx);
-
-/**
- * snd_soc_put_volsw_2r_sx - double with tlv and variable data size
- *  mixer put callback
- * @kcontrol: mixer control
- * @uinfo: control element information
- *
- * Returns 0 for success.
- */
-int snd_soc_put_volsw_2r_sx(struct snd_kcontrol *kcontrol,
-			struct snd_ctl_elem_value *ucontrol)
-{
-	struct soc_mixer_control *mc =
-		(struct soc_mixer_control *)kcontrol->private_value;
-	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
-	unsigned int mask = (1<<mc->shift)-1;
-	int min = mc->min;
-	int ret;
-	unsigned int val, valr, oval, ovalr;
-
-	val = ((ucontrol->value.integer.value[0]+min) & 0xff);
-	val &= mask;
-	valr = ((ucontrol->value.integer.value[1]+min) & 0xff);
-	valr &= mask;
-
-	oval = snd_soc_read(codec, mc->reg) & mask;
-	ovalr = snd_soc_read(codec, mc->rreg) & mask;
-
-	ret = 0;
-	if (oval != val) {
-		ret = snd_soc_write(codec, mc->reg, val);
-		if (ret < 0)
-			return ret;
-	}
-	if (ovalr != valr) {
-		ret = snd_soc_write(codec, mc->rreg, valr);
-		if (ret < 0)
-			return ret;
-	}
-
-	return 0;
-}
-EXPORT_SYMBOL_GPL(snd_soc_put_volsw_2r_sx);
-
 int snd_soc_bytes_info(struct snd_kcontrol *kcontrol,
 		       struct snd_ctl_elem_info *uinfo)
 {
@@ -2849,6 +2937,186 @@ int snd_soc_bytes_put(struct snd_kcontrol *kcontrol,
 }
 EXPORT_SYMBOL_GPL(snd_soc_bytes_put);
 
+/**
+ * snd_soc_info_xr_sx - signed multi register info callback
+ * @kcontrol: mreg control
+ * @uinfo: control element information
+ *
+ * Callback to provide information of a control that can
+ * span multiple codec registers which together
+ * forms a single signed value in a MSB/LSB manner.
+ *
+ * Returns 0 for success.
+ */
+int snd_soc_info_xr_sx(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_info *uinfo)
+{
+	struct soc_mreg_control *mc =
+		(struct soc_mreg_control *)kcontrol->private_value;
+	uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+	uinfo->count = 1;
+	uinfo->value.integer.min = mc->min;
+	uinfo->value.integer.max = mc->max;
+
+	return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_info_xr_sx);
+
+/**
+ * snd_soc_get_xr_sx - signed multi register get callback
+ * @kcontrol: mreg control
+ * @ucontrol: control element information
+ *
+ * Callback to get the value of a control that can span
+ * multiple codec registers which together forms a single
+ * signed value in a MSB/LSB manner. The control supports
+ * specifying total no of bits used to allow for bitfields
+ * across the multiple codec registers.
+ *
+ * Returns 0 for success.
+ */
+int snd_soc_get_xr_sx(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct soc_mreg_control *mc =
+		(struct soc_mreg_control *)kcontrol->private_value;
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	unsigned int regbase = mc->regbase;
+	unsigned int regcount = mc->regcount;
+	unsigned int regwshift = codec->driver->reg_word_size * BITS_PER_BYTE;
+	unsigned int regwmask = (1<<regwshift)-1;
+	unsigned int invert = mc->invert;
+	unsigned long mask = (1UL<<mc->nbits)-1;
+	long min = mc->min;
+	long max = mc->max;
+	long val = 0;
+	unsigned long regval;
+	unsigned int i;
+
+	for (i = 0; i < regcount; i++) {
+		regval = snd_soc_read(codec, regbase+i) & regwmask;
+		val |= regval << (regwshift*(regcount-i-1));
+	}
+	val &= mask;
+	if (min < 0 && val > max)
+		val |= ~mask;
+	if (invert)
+		val = max - val;
+	ucontrol->value.integer.value[0] = val;
+
+	return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_get_xr_sx);
+
+/**
+ * snd_soc_put_xr_sx - signed multi register get callback
+ * @kcontrol: mreg control
+ * @ucontrol: control element information
+ *
+ * Callback to set the value of a control that can span
+ * multiple codec registers which together forms a single
+ * signed value in a MSB/LSB manner. The control supports
+ * specifying total no of bits used to allow for bitfields
+ * across the multiple codec registers.
+ *
+ * Returns 0 for success.
+ */
+int snd_soc_put_xr_sx(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct soc_mreg_control *mc =
+		(struct soc_mreg_control *)kcontrol->private_value;
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	unsigned int regbase = mc->regbase;
+	unsigned int regcount = mc->regcount;
+	unsigned int regwshift = codec->driver->reg_word_size * BITS_PER_BYTE;
+	unsigned int regwmask = (1<<regwshift)-1;
+	unsigned int invert = mc->invert;
+	unsigned long mask = (1UL<<mc->nbits)-1;
+	long max = mc->max;
+	long val = ucontrol->value.integer.value[0];
+	unsigned int i, regval, regmask;
+	int err;
+
+	if (invert)
+		val = max - val;
+	val &= mask;
+	for (i = 0; i < regcount; i++) {
+		regval = (val >> (regwshift*(regcount-i-1))) & regwmask;
+		regmask = (mask >> (regwshift*(regcount-i-1))) & regwmask;
+		err = snd_soc_update_bits_locked(codec, regbase+i,
+				regmask, regval);
+		if (err < 0)
+			return err;
+	}
+
+	return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_put_xr_sx);
+
+/**
+ * snd_soc_get_strobe - strobe get callback
+ * @kcontrol: mixer control
+ * @ucontrol: control element information
+ *
+ * Callback get the value of a strobe mixer control.
+ *
+ * Returns 0 for success.
+ */
+int snd_soc_get_strobe(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct soc_mixer_control *mc =
+		(struct soc_mixer_control *)kcontrol->private_value;
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	unsigned int reg = mc->reg;
+	unsigned int shift = mc->shift;
+	unsigned int mask = 1 << shift;
+	unsigned int invert = mc->invert != 0;
+	unsigned int val = snd_soc_read(codec, reg) & mask;
+
+	if (shift != 0 && val != 0)
+		val = val >> shift;
+	ucontrol->value.enumerated.item[0] = val ^ invert;
+
+	return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_get_strobe);
+
+/**
+ * snd_soc_put_strobe - strobe put callback
+ * @kcontrol: mixer control
+ * @ucontrol: control element information
+ *
+ * Callback strobe a register bit to high then low (or the inverse)
+ * in one pass of a single mixer enum control.
+ *
+ * Returns 1 for success.
+ */
+int snd_soc_put_strobe(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct soc_mixer_control *mc =
+		(struct soc_mixer_control *)kcontrol->private_value;
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	unsigned int reg = mc->reg;
+	unsigned int shift = mc->shift;
+	unsigned int mask = 1 << shift;
+	unsigned int invert = mc->invert != 0;
+	unsigned int strobe = ucontrol->value.enumerated.item[0] != 0;
+	unsigned int val1 = (strobe ^ invert) ? mask : 0;
+	unsigned int val2 = (strobe ^ invert) ? 0 : mask;
+	int err;
+
+	err = snd_soc_update_bits_locked(codec, reg, mask, val1);
+	if (err < 0)
+		return err;
+
+	err = snd_soc_update_bits_locked(codec, reg, mask, val2);
+	return err;
+}
+EXPORT_SYMBOL_GPL(snd_soc_put_strobe);
+
 /**
  * snd_soc_dai_set_sysclk - configure DAI system or master clock.
  * @dai: DAI
@@ -3048,7 +3316,7 @@ int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute)
 	if (dai->driver && dai->driver->ops->digital_mute)
 		return dai->driver->ops->digital_mute(dai, mute);
 	else
-		return -EINVAL;
+		return -ENOTSUPP;
 }
 EXPORT_SYMBOL_GPL(snd_soc_dai_digital_mute);
 
@@ -3060,7 +3328,7 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_digital_mute);
  */
 int snd_soc_register_card(struct snd_soc_card *card)
 {
-	int i;
+	int i, ret;
 
 	if (!card->name || !card->dev)
 		return -EINVAL;
@@ -3123,15 +3391,13 @@ int snd_soc_register_card(struct snd_soc_card *card)
 	INIT_LIST_HEAD(&card->dapm_dirty);
 	card->instantiated = 0;
 	mutex_init(&card->mutex);
+	mutex_init(&card->dapm_mutex);
 
-	mutex_lock(&client_mutex);
-	list_add(&card->list, &card_list);
-	snd_soc_instantiate_cards();
-	mutex_unlock(&client_mutex);
+	ret = snd_soc_instantiate_card(card);
+	if (ret != 0)
+		soc_cleanup_card_debugfs(card);
 
-	dev_dbg(card->dev, "Registered card '%s'\n", card->name);
-
-	return 0;
+	return ret;
 }
 EXPORT_SYMBOL_GPL(snd_soc_register_card);
 
@@ -3145,9 +3411,6 @@ int snd_soc_unregister_card(struct snd_soc_card *card)
 {
 	if (card->instantiated)
 		soc_cleanup_card_resources(card);
-	mutex_lock(&client_mutex);
-	list_del(&card->list);
-	mutex_unlock(&client_mutex);
 	dev_dbg(card->dev, "Unregistered card '%s'\n", card->name);
 
 	return 0;
@@ -3221,6 +3484,7 @@ static inline char *fmt_multiple_name(struct device *dev,
 int snd_soc_register_dai(struct device *dev,
 		struct snd_soc_dai_driver *dai_drv)
 {
+	struct snd_soc_codec *codec;
 	struct snd_soc_dai *dai;
 
 	dev_dbg(dev, "dai register %s\n", dev_name(dev));
@@ -3238,12 +3502,23 @@ int snd_soc_register_dai(struct device *dev,
 
 	dai->dev = dev;
 	dai->driver = dai_drv;
+	dai->dapm.dev = dev;
 	if (!dai->driver->ops)
 		dai->driver->ops = &null_dai_ops;
 
 	mutex_lock(&client_mutex);
+
+	list_for_each_entry(codec, &codec_list, list) {
+		if (codec->dev == dev) {
+			dev_dbg(dev, "Mapped DAI %s to CODEC %s\n",
+				dai->name, codec->name);
+			dai->codec = codec;
+			break;
+		}
+	}
+
 	list_add(&dai->list, &dai_list);
-	snd_soc_instantiate_cards();
+
 	mutex_unlock(&client_mutex);
 
 	pr_debug("Registered DAI '%s'\n", dai->name);
@@ -3287,6 +3562,7 @@ EXPORT_SYMBOL_GPL(snd_soc_unregister_dai);
 int snd_soc_register_dais(struct device *dev,
 		struct snd_soc_dai_driver *dai_drv, size_t count)
 {
+	struct snd_soc_codec *codec;
 	struct snd_soc_dai *dai;
 	int i, ret = 0;
 
@@ -3314,19 +3590,28 @@ int snd_soc_register_dais(struct device *dev,
 			dai->id = dai->driver->id;
 		else
 			dai->id = i;
+		dai->dapm.dev = dev;
 		if (!dai->driver->ops)
 			dai->driver->ops = &null_dai_ops;
 
 		mutex_lock(&client_mutex);
+
+		list_for_each_entry(codec, &codec_list, list) {
+			if (codec->dev == dev) {
+				dev_dbg(dev, "Mapped DAI %s to CODEC %s\n",
+					dai->name, codec->name);
+				dai->codec = codec;
+				break;
+			}
+		}
+
 		list_add(&dai->list, &dai_list);
+
 		mutex_unlock(&client_mutex);
 
 		pr_debug("Registered DAI '%s'\n", dai->name);
 	}
 
-	mutex_lock(&client_mutex);
-	snd_soc_instantiate_cards();
-	mutex_unlock(&client_mutex);
 	return 0;
 
 err:
@@ -3384,7 +3669,6 @@ int snd_soc_register_platform(struct device *dev,
 
 	mutex_lock(&client_mutex);
 	list_add(&platform->list, &platform_list);
-	snd_soc_instantiate_cards();
 	mutex_unlock(&client_mutex);
 
 	pr_debug("Registered platform '%s'\n", platform->name);
@@ -3534,18 +3818,18 @@ int snd_soc_register_codec(struct device *dev,
 		fixup_codec_formats(&dai_drv[i].capture);
 	}
 
+	mutex_lock(&client_mutex);
+	list_add(&codec->list, &codec_list);
+	mutex_unlock(&client_mutex);
+
 	/* register any DAIs */
 	if (num_dai) {
 		ret = snd_soc_register_dais(dev, dai_drv, num_dai);
 		if (ret < 0)
-			goto fail;
+			dev_err(codec->dev, "Failed to regster DAIs: %d\n",
+				ret);
 	}
 
-	mutex_lock(&client_mutex);
-	list_add(&codec->list, &codec_list);
-	snd_soc_instantiate_cards();
-	mutex_unlock(&client_mutex);
-
 	pr_debug("Registered codec '%s'\n", codec->name);
 	return 0;
 
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 1bb6d4a63cd8..90ee77d2409d 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -52,6 +52,7 @@ static int dapm_up_seq[] = {
 	[snd_soc_dapm_supply] = 1,
 	[snd_soc_dapm_regulator_supply] = 1,
 	[snd_soc_dapm_micbias] = 2,
+	[snd_soc_dapm_dai_link] = 2,
 	[snd_soc_dapm_dai] = 3,
 	[snd_soc_dapm_aif_in] = 3,
 	[snd_soc_dapm_aif_out] = 3,
@@ -90,9 +91,10 @@ static int dapm_down_seq[] = {
 	[snd_soc_dapm_aif_in] = 10,
 	[snd_soc_dapm_aif_out] = 10,
 	[snd_soc_dapm_dai] = 10,
-	[snd_soc_dapm_regulator_supply] = 11,
-	[snd_soc_dapm_supply] = 11,
-	[snd_soc_dapm_post] = 12,
+	[snd_soc_dapm_dai_link] = 11,
+	[snd_soc_dapm_regulator_supply] = 12,
+	[snd_soc_dapm_supply] = 12,
+	[snd_soc_dapm_post] = 13,
 };
 
 static void pop_wait(u32 pop_time)
@@ -208,7 +210,23 @@ static int soc_widget_write(struct snd_soc_dapm_widget *w, int reg, int val)
 	return -1;
 }
 
-static int soc_widget_update_bits(struct snd_soc_dapm_widget *w,
+static inline void soc_widget_lock(struct snd_soc_dapm_widget *w)
+{
+	if (w->codec && !w->codec->using_regmap)
+		mutex_lock(&w->codec->mutex);
+	else if (w->platform)
+		mutex_lock(&w->platform->mutex);
+}
+
+static inline void soc_widget_unlock(struct snd_soc_dapm_widget *w)
+{
+	if (w->codec && !w->codec->using_regmap)
+		mutex_unlock(&w->codec->mutex);
+	else if (w->platform)
+		mutex_unlock(&w->platform->mutex);
+}
+
+static int soc_widget_update_bits_locked(struct snd_soc_dapm_widget *w,
 	unsigned short reg, unsigned int mask, unsigned int value)
 {
 	bool change;
@@ -221,18 +239,24 @@ static int soc_widget_update_bits(struct snd_soc_dapm_widget *w,
 		if (ret != 0)
 			return ret;
 	} else {
+		soc_widget_lock(w);
 		ret = soc_widget_read(w, reg);
-		if (ret < 0)
+		if (ret < 0) {
+			soc_widget_unlock(w);
 			return ret;
+		}
 
 		old = ret;
 		new = (old & ~mask) | (value & mask);
 		change = old != new;
 		if (change) {
 			ret = soc_widget_write(w, reg, new);
-			if (ret < 0)
+			if (ret < 0) {
+				soc_widget_unlock(w);
 				return ret;
+			}
 		}
+		soc_widget_unlock(w);
 	}
 
 	return change;
@@ -374,6 +398,7 @@ static void dapm_set_path_status(struct snd_soc_dapm_widget *w,
 	case snd_soc_dapm_mic:
 	case snd_soc_dapm_spk:
 	case snd_soc_dapm_line:
+	case snd_soc_dapm_dai_link:
 		p->connect = 1;
 	break;
 	/* does affect routing - dynamically connected */
@@ -682,11 +707,51 @@ static int snd_soc_dapm_suspend_check(struct snd_soc_dapm_widget *widget)
 	}
 }
 
+/* add widget to list if it's not already in the list */
+static int dapm_list_add_widget(struct snd_soc_dapm_widget_list **list,
+	struct snd_soc_dapm_widget *w)
+{
+	struct snd_soc_dapm_widget_list *wlist;
+	int wlistsize, wlistentries, i;
+
+	if (*list == NULL)
+		return -EINVAL;
+
+	wlist = *list;
+
+	/* is this widget already in the list */
+	for (i = 0; i < wlist->num_widgets; i++) {
+		if (wlist->widgets[i] == w)
+			return 0;
+	}
+
+	/* allocate some new space */
+	wlistentries = wlist->num_widgets + 1;
+	wlistsize = sizeof(struct snd_soc_dapm_widget_list) +
+			wlistentries * sizeof(struct snd_soc_dapm_widget *);
+	*list = krealloc(wlist, wlistsize, GFP_KERNEL);
+	if (*list == NULL) {
+		dev_err(w->dapm->dev, "can't allocate widget list for %s\n",
+			w->name);
+		return -ENOMEM;
+	}
+	wlist = *list;
+
+	/* insert the widget */
+	dev_dbg(w->dapm->dev, "added %s in widget list pos %d\n",
+			w->name, wlist->num_widgets);
+
+	wlist->widgets[wlist->num_widgets] = w;
+	wlist->num_widgets++;
+	return 1;
+}
+
 /*
  * Recursively check for a completed path to an active or physically connected
  * output widget. Returns number of complete paths.
  */
-static int is_connected_output_ep(struct snd_soc_dapm_widget *widget)
+static int is_connected_output_ep(struct snd_soc_dapm_widget *widget,
+	struct snd_soc_dapm_widget_list **list)
 {
 	struct snd_soc_dapm_path *path;
 	int con = 0;
@@ -742,9 +807,23 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget)
 		if (path->walked)
 			continue;
 
+		trace_snd_soc_dapm_output_path(widget, path);
+
 		if (path->sink && path->connect) {
 			path->walked = 1;
-			con += is_connected_output_ep(path->sink);
+
+			/* do we need to add this widget to the list ? */
+			if (list) {
+				int err;
+				err = dapm_list_add_widget(list, path->sink);
+				if (err < 0) {
+					dev_err(widget->dapm->dev, "could not add widget %s\n",
+						widget->name);
+					return con;
+				}
+			}
+
+			con += is_connected_output_ep(path->sink, list);
 		}
 	}
 
@@ -757,7 +836,8 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget)
  * Recursively check for a completed path to an active or physically connected
  * input widget. Returns number of complete paths.
  */
-static int is_connected_input_ep(struct snd_soc_dapm_widget *widget)
+static int is_connected_input_ep(struct snd_soc_dapm_widget *widget,
+	struct snd_soc_dapm_widget_list **list)
 {
 	struct snd_soc_dapm_path *path;
 	int con = 0;
@@ -825,9 +905,23 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget)
 		if (path->walked)
 			continue;
 
+		trace_snd_soc_dapm_input_path(widget, path);
+
 		if (path->source && path->connect) {
 			path->walked = 1;
-			con += is_connected_input_ep(path->source);
+
+			/* do we need to add this widget to the list ? */
+			if (list) {
+				int err;
+				err = dapm_list_add_widget(list, path->sink);
+				if (err < 0) {
+					dev_err(widget->dapm->dev, "could not add widget %s\n",
+						widget->name);
+					return con;
+				}
+			}
+
+			con += is_connected_input_ep(path->source, list);
 		}
 	}
 
@@ -836,6 +930,39 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget)
 	return con;
 }
 
+/**
+ * snd_soc_dapm_get_connected_widgets - query audio path and it's widgets.
+ * @dai: the soc DAI.
+ * @stream: stream direction.
+ * @list: list of active widgets for this stream.
+ *
+ * Queries DAPM graph as to whether an valid audio stream path exists for
+ * the initial stream specified by name. This takes into account
+ * current mixer and mux kcontrol settings. Creates list of valid widgets.
+ *
+ * Returns the number of valid paths or negative error.
+ */
+int snd_soc_dapm_dai_get_connected_widgets(struct snd_soc_dai *dai, int stream,
+	struct snd_soc_dapm_widget_list **list)
+{
+	struct snd_soc_card *card = dai->card;
+	int paths;
+
+	mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME);
+	dapm_reset(card);
+
+	if (stream == SNDRV_PCM_STREAM_PLAYBACK)
+		paths = is_connected_output_ep(dai->playback_widget, list);
+	else
+		paths = is_connected_input_ep(dai->playback_widget, list);
+
+	trace_snd_soc_dapm_connected(paths, stream);
+	dapm_clear_walk(&card->dapm);
+	mutex_unlock(&card->dapm_mutex);
+
+	return paths;
+}
+
 /*
  * Handler for generic register modifier widget.
  */
@@ -849,7 +976,7 @@ int dapm_reg_event(struct snd_soc_dapm_widget *w,
 	else
 		val = w->off_val;
 
-	soc_widget_update_bits(w, -(w->reg + 1),
+	soc_widget_update_bits_locked(w, -(w->reg + 1),
 			    w->mask << w->shift, val << w->shift);
 
 	return 0;
@@ -863,9 +990,9 @@ int dapm_regulator_event(struct snd_soc_dapm_widget *w,
 		   struct snd_kcontrol *kcontrol, int event)
 {
 	if (SND_SOC_DAPM_EVENT_ON(event))
-		return regulator_enable(w->priv);
+		return regulator_enable(w->regulator);
 	else
-		return regulator_disable_deferred(w->priv, w->shift);
+		return regulator_disable_deferred(w->regulator, w->shift);
 }
 EXPORT_SYMBOL_GPL(dapm_regulator_event);
 
@@ -892,9 +1019,9 @@ static int dapm_generic_check_power(struct snd_soc_dapm_widget *w)
 
 	DAPM_UPDATE_STAT(w, power_checks);
 
-	in = is_connected_input_ep(w);
+	in = is_connected_input_ep(w, NULL);
 	dapm_clear_walk(w->dapm);
-	out = is_connected_output_ep(w);
+	out = is_connected_output_ep(w, NULL);
 	dapm_clear_walk(w->dapm);
 	return out != 0 && in != 0;
 }
@@ -903,7 +1030,10 @@ static int dapm_dai_check_power(struct snd_soc_dapm_widget *w)
 {
 	DAPM_UPDATE_STAT(w, power_checks);
 
-	return w->active;
+	if (w->active)
+		return w->active;
+
+	return dapm_generic_check_power(w);
 }
 
 /* Check to see if an ADC has power */
@@ -914,7 +1044,7 @@ static int dapm_adc_check_power(struct snd_soc_dapm_widget *w)
 	DAPM_UPDATE_STAT(w, power_checks);
 
 	if (w->active) {
-		in = is_connected_input_ep(w);
+		in = is_connected_input_ep(w, NULL);
 		dapm_clear_walk(w->dapm);
 		return in != 0;
 	} else {
@@ -930,7 +1060,7 @@ static int dapm_dac_check_power(struct snd_soc_dapm_widget *w)
 	DAPM_UPDATE_STAT(w, power_checks);
 
 	if (w->active) {
-		out = is_connected_output_ep(w);
+		out = is_connected_output_ep(w, NULL);
 		dapm_clear_walk(w->dapm);
 		return out != 0;
 	} else {
@@ -1107,7 +1237,7 @@ static void dapm_seq_run_coalesced(struct snd_soc_dapm_context *dapm,
 			"pop test : Applying 0x%x/0x%x to %x in %dms\n",
 			value, mask, reg, card->pop_time);
 		pop_wait(card->pop_time);
-		soc_widget_update_bits(w, reg, mask, value);
+		soc_widget_update_bits_locked(w, reg, mask, value);
 	}
 
 	list_for_each_entry(w, pending, power_list) {
@@ -1237,7 +1367,7 @@ static void dapm_widget_update(struct snd_soc_dapm_context *dapm)
 			       w->name, ret);
 	}
 
-	ret = snd_soc_update_bits(w->codec, update->reg, update->mask,
+	ret = soc_widget_update_bits_locked(w, update->reg, update->mask,
 				  update->val);
 	if (ret < 0)
 		pr_err("%s DAPM update failed: %d\n", w->name, ret);
@@ -1421,12 +1551,10 @@ static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event)
 	trace_snd_soc_dapm_start(card);
 
 	list_for_each_entry(d, &card->dapm_list, list) {
-		if (d->n_widgets || d->codec == NULL) {
-			if (d->idle_bias_off)
-				d->target_bias_level = SND_SOC_BIAS_OFF;
-			else
-				d->target_bias_level = SND_SOC_BIAS_STANDBY;
-		}
+		if (d->idle_bias_off)
+			d->target_bias_level = SND_SOC_BIAS_OFF;
+		else
+			d->target_bias_level = SND_SOC_BIAS_STANDBY;
 	}
 
 	dapm_reset(card);
@@ -1471,32 +1599,6 @@ static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event)
 
 	}
 
-	/* If there are no DAPM widgets then try to figure out power from the
-	 * event type.
-	 */
-	if (!dapm->n_widgets) {
-		switch (event) {
-		case SND_SOC_DAPM_STREAM_START:
-		case SND_SOC_DAPM_STREAM_RESUME:
-			dapm->target_bias_level = SND_SOC_BIAS_ON;
-			break;
-		case SND_SOC_DAPM_STREAM_STOP:
-			if (dapm->codec && dapm->codec->active)
-				dapm->target_bias_level = SND_SOC_BIAS_ON;
-			else
-				dapm->target_bias_level = SND_SOC_BIAS_STANDBY;
-			break;
-		case SND_SOC_DAPM_STREAM_SUSPEND:
-			dapm->target_bias_level = SND_SOC_BIAS_STANDBY;
-			break;
-		case SND_SOC_DAPM_STREAM_NOP:
-			dapm->target_bias_level = dapm->bias_level;
-			break;
-		default:
-			break;
-		}
-	}
-
 	/* Force all contexts in the card to the same bias state if
 	 * they're not ground referenced.
 	 */
@@ -1560,9 +1662,9 @@ static ssize_t dapm_widget_power_read_file(struct file *file,
 	if (!buf)
 		return -ENOMEM;
 
-	in = is_connected_input_ep(w);
+	in = is_connected_input_ep(w, NULL);
 	dapm_clear_walk(w->dapm);
-	out = is_connected_output_ep(w);
+	out = is_connected_output_ep(w, NULL);
 	dapm_clear_walk(w->dapm);
 
 	ret = snprintf(buf, PAGE_SIZE, "%s: %s%s  in %d out %d",
@@ -1709,7 +1811,7 @@ static inline void dapm_debugfs_cleanup(struct snd_soc_dapm_context *dapm)
 #endif
 
 /* test and update the power status of a mux widget */
-int snd_soc_dapm_mux_update_power(struct snd_soc_dapm_widget *widget,
+static int soc_dapm_mux_update_power(struct snd_soc_dapm_widget *widget,
 				 struct snd_kcontrol *kcontrol, int mux, struct soc_enum *e)
 {
 	struct snd_soc_dapm_path *path;
@@ -1746,12 +1848,26 @@ int snd_soc_dapm_mux_update_power(struct snd_soc_dapm_widget *widget,
 		dapm_power_widgets(widget->dapm, SND_SOC_DAPM_STREAM_NOP);
 	}
 
-	return 0;
+	return found;
+}
+
+int snd_soc_dapm_mux_update_power(struct snd_soc_dapm_widget *widget,
+		struct snd_kcontrol *kcontrol, int mux, struct soc_enum *e)
+{
+	struct snd_soc_card *card = widget->dapm->card;
+	int ret;
+
+	mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME);
+	ret = soc_dapm_mux_update_power(widget, kcontrol, mux, e);
+	mutex_unlock(&card->dapm_mutex);
+	if (ret > 0)
+		soc_dpcm_runtime_update(widget);
+	return ret;
 }
 EXPORT_SYMBOL_GPL(snd_soc_dapm_mux_update_power);
 
 /* test and update the power status of a mixer or switch widget */
-int snd_soc_dapm_mixer_update_power(struct snd_soc_dapm_widget *widget,
+static int soc_dapm_mixer_update_power(struct snd_soc_dapm_widget *widget,
 				   struct snd_kcontrol *kcontrol, int connect)
 {
 	struct snd_soc_dapm_path *path;
@@ -1778,7 +1894,21 @@ int snd_soc_dapm_mixer_update_power(struct snd_soc_dapm_widget *widget,
 		dapm_power_widgets(widget->dapm, SND_SOC_DAPM_STREAM_NOP);
 	}
 
-	return 0;
+	return found;
+}
+
+int snd_soc_dapm_mixer_update_power(struct snd_soc_dapm_widget *widget,
+				struct snd_kcontrol *kcontrol, int connect)
+{
+	struct snd_soc_card *card = widget->dapm->card;
+	int ret;
+
+	mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME);
+	ret = soc_dapm_mixer_update_power(widget, kcontrol, connect);
+	mutex_unlock(&card->dapm_mutex);
+	if (ret > 0)
+		soc_dpcm_runtime_update(widget);
+	return ret;
 }
 EXPORT_SYMBOL_GPL(snd_soc_dapm_mixer_update_power);
 
@@ -1939,6 +2069,8 @@ static int snd_soc_dapm_set_pin(struct snd_soc_dapm_context *dapm,
  */
 int snd_soc_dapm_sync(struct snd_soc_dapm_context *dapm)
 {
+	int ret;
+
 	/*
 	 * Suppress early reports (eg, jacks syncing their state) to avoid
 	 * silly DAPM runs during card startup.
@@ -1946,7 +2078,10 @@ int snd_soc_dapm_sync(struct snd_soc_dapm_context *dapm)
 	if (!dapm->card || !dapm->card->instantiated)
 		return 0;
 
-	return dapm_power_widgets(dapm, SND_SOC_DAPM_STREAM_NOP);
+	mutex_lock_nested(&dapm->card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME);
+	ret = dapm_power_widgets(dapm, SND_SOC_DAPM_STREAM_NOP);
+	mutex_unlock(&dapm->card->dapm_mutex);
+	return ret;
 }
 EXPORT_SYMBOL_GPL(snd_soc_dapm_sync);
 
@@ -2055,6 +2190,7 @@ static int snd_soc_dapm_add_route(struct snd_soc_dapm_context *dapm,
 	case snd_soc_dapm_aif_in:
 	case snd_soc_dapm_aif_out:
 	case snd_soc_dapm_dai:
+	case snd_soc_dapm_dai_link:
 		list_add(&path->list, &dapm->card->paths);
 		list_add(&path->list_sink, &wsink->sources);
 		list_add(&path->list_source, &wsource->sinks);
@@ -2110,19 +2246,21 @@ err:
 int snd_soc_dapm_add_routes(struct snd_soc_dapm_context *dapm,
 			    const struct snd_soc_dapm_route *route, int num)
 {
-	int i, ret;
+	int i, ret = 0;
 
+	mutex_lock_nested(&dapm->card->dapm_mutex, SND_SOC_DAPM_CLASS_INIT);
 	for (i = 0; i < num; i++) {
 		ret = snd_soc_dapm_add_route(dapm, route);
 		if (ret < 0) {
 			dev_err(dapm->dev, "Failed to add route %s->%s\n",
 				route->source, route->sink);
-			return ret;
+			break;
 		}
 		route++;
 	}
+	mutex_unlock(&dapm->card->dapm_mutex);
 
-	return 0;
+	return ret;
 }
 EXPORT_SYMBOL_GPL(snd_soc_dapm_add_routes);
 
@@ -2193,12 +2331,14 @@ int snd_soc_dapm_weak_routes(struct snd_soc_dapm_context *dapm,
 	int i, err;
 	int ret = 0;
 
+	mutex_lock_nested(&dapm->card->dapm_mutex, SND_SOC_DAPM_CLASS_INIT);
 	for (i = 0; i < num; i++) {
 		err = snd_soc_dapm_weak_route(dapm, route);
 		if (err)
 			ret = err;
 		route++;
 	}
+	mutex_unlock(&dapm->card->dapm_mutex);
 
 	return ret;
 }
@@ -2217,6 +2357,8 @@ int snd_soc_dapm_new_widgets(struct snd_soc_dapm_context *dapm)
 	struct snd_soc_dapm_widget *w;
 	unsigned int val;
 
+	mutex_lock_nested(&dapm->card->dapm_mutex, SND_SOC_DAPM_CLASS_INIT);
+
 	list_for_each_entry(w, &dapm->card->widgets, list)
 	{
 		if (w->new)
@@ -2226,8 +2368,10 @@ int snd_soc_dapm_new_widgets(struct snd_soc_dapm_context *dapm)
 			w->kcontrols = kzalloc(w->num_kcontrols *
 						sizeof(struct snd_kcontrol *),
 						GFP_KERNEL);
-			if (!w->kcontrols)
+			if (!w->kcontrols) {
+				mutex_unlock(&dapm->card->dapm_mutex);
 				return -ENOMEM;
+			}
 		}
 
 		switch(w->id) {
@@ -2267,6 +2411,7 @@ int snd_soc_dapm_new_widgets(struct snd_soc_dapm_context *dapm)
 	}
 
 	dapm_power_widgets(dapm, SND_SOC_DAPM_STREAM_NOP);
+	mutex_unlock(&dapm->card->dapm_mutex);
 	return 0;
 }
 EXPORT_SYMBOL_GPL(snd_soc_dapm_new_widgets);
@@ -2326,6 +2471,7 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol,
 	struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol);
 	struct snd_soc_dapm_widget *widget = wlist->widgets[0];
 	struct snd_soc_codec *codec = widget->codec;
+	struct snd_soc_card *card = codec->card;
 	struct soc_mixer_control *mc =
 		(struct soc_mixer_control *)kcontrol->private_value;
 	unsigned int reg = mc->reg;
@@ -2352,7 +2498,7 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol,
 		/* old connection must be powered down */
 		connect = invert ? 1 : 0;
 
-	mutex_lock(&codec->mutex);
+	mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME);
 
 	change = snd_soc_test_bits(widget->codec, reg, mask, val);
 	if (change) {
@@ -2368,13 +2514,13 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol,
 			update.val = val;
 			widget->dapm->update = &update;
 
-			snd_soc_dapm_mixer_update_power(widget, kcontrol, connect);
+			soc_dapm_mixer_update_power(widget, kcontrol, connect);
 
 			widget->dapm->update = NULL;
 		}
 	}
 
-	mutex_unlock(&codec->mutex);
+	mutex_unlock(&card->dapm_mutex);
 	return 0;
 }
 EXPORT_SYMBOL_GPL(snd_soc_dapm_put_volsw);
@@ -2423,6 +2569,7 @@ int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol,
 	struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol);
 	struct snd_soc_dapm_widget *widget = wlist->widgets[0];
 	struct snd_soc_codec *codec = widget->codec;
+	struct snd_soc_card *card = codec->card;
 	struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
 	unsigned int val, mux, change;
 	unsigned int mask, bitmask;
@@ -2443,7 +2590,7 @@ int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol,
 		mask |= (bitmask - 1) << e->shift_r;
 	}
 
-	mutex_lock(&codec->mutex);
+	mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME);
 
 	change = snd_soc_test_bits(widget->codec, e->reg, mask, val);
 	if (change) {
@@ -2459,13 +2606,13 @@ int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol,
 			update.val = val;
 			widget->dapm->update = &update;
 
-			snd_soc_dapm_mux_update_power(widget, kcontrol, mux, e);
+			soc_dapm_mux_update_power(widget, kcontrol, mux, e);
 
 			widget->dapm->update = NULL;
 		}
 	}
 
-	mutex_unlock(&codec->mutex);
+	mutex_unlock(&card->dapm_mutex);
 	return change;
 }
 EXPORT_SYMBOL_GPL(snd_soc_dapm_put_enum_double);
@@ -2502,6 +2649,7 @@ int snd_soc_dapm_put_enum_virt(struct snd_kcontrol *kcontrol,
 	struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol);
 	struct snd_soc_dapm_widget *widget = wlist->widgets[0];
 	struct snd_soc_codec *codec = widget->codec;
+	struct snd_soc_card *card = codec->card;
 	struct soc_enum *e =
 		(struct soc_enum *)kcontrol->private_value;
 	int change;
@@ -2511,7 +2659,7 @@ int snd_soc_dapm_put_enum_virt(struct snd_kcontrol *kcontrol,
 	if (ucontrol->value.enumerated.item[0] >= e->max)
 		return -EINVAL;
 
-	mutex_lock(&codec->mutex);
+	mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME);
 
 	change = widget->value != ucontrol->value.enumerated.item[0];
 	if (change) {
@@ -2520,11 +2668,11 @@ int snd_soc_dapm_put_enum_virt(struct snd_kcontrol *kcontrol,
 
 			widget->value = ucontrol->value.enumerated.item[0];
 
-			snd_soc_dapm_mux_update_power(widget, kcontrol, widget->value, e);
+			soc_dapm_mux_update_power(widget, kcontrol, widget->value, e);
 		}
 	}
 
-	mutex_unlock(&codec->mutex);
+	mutex_unlock(&card->dapm_mutex);
 	return ret;
 }
 EXPORT_SYMBOL_GPL(snd_soc_dapm_put_enum_virt);
@@ -2589,6 +2737,7 @@ int snd_soc_dapm_put_value_enum_double(struct snd_kcontrol *kcontrol,
 	struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol);
 	struct snd_soc_dapm_widget *widget = wlist->widgets[0];
 	struct snd_soc_codec *codec = widget->codec;
+	struct snd_soc_card *card = codec->card;
 	struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
 	unsigned int val, mux, change;
 	unsigned int mask;
@@ -2607,7 +2756,7 @@ int snd_soc_dapm_put_value_enum_double(struct snd_kcontrol *kcontrol,
 		mask |= e->mask << e->shift_r;
 	}
 
-	mutex_lock(&codec->mutex);
+	mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME);
 
 	change = snd_soc_test_bits(widget->codec, e->reg, mask, val);
 	if (change) {
@@ -2623,13 +2772,13 @@ int snd_soc_dapm_put_value_enum_double(struct snd_kcontrol *kcontrol,
 			update.val = val;
 			widget->dapm->update = &update;
 
-			snd_soc_dapm_mux_update_power(widget, kcontrol, mux, e);
+			soc_dapm_mux_update_power(widget, kcontrol, mux, e);
 
 			widget->dapm->update = NULL;
 		}
 	}
 
-	mutex_unlock(&codec->mutex);
+	mutex_unlock(&card->dapm_mutex);
 	return change;
 }
 EXPORT_SYMBOL_GPL(snd_soc_dapm_put_value_enum_double);
@@ -2666,12 +2815,12 @@ int snd_soc_dapm_get_pin_switch(struct snd_kcontrol *kcontrol,
 	struct snd_soc_card *card = snd_kcontrol_chip(kcontrol);
 	const char *pin = (const char *)kcontrol->private_value;
 
-	mutex_lock(&card->mutex);
+	mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME);
 
 	ucontrol->value.integer.value[0] =
 		snd_soc_dapm_get_pin_status(&card->dapm, pin);
 
-	mutex_unlock(&card->mutex);
+	mutex_unlock(&card->dapm_mutex);
 
 	return 0;
 }
@@ -2689,17 +2838,16 @@ int snd_soc_dapm_put_pin_switch(struct snd_kcontrol *kcontrol,
 	struct snd_soc_card *card = snd_kcontrol_chip(kcontrol);
 	const char *pin = (const char *)kcontrol->private_value;
 
-	mutex_lock(&card->mutex);
+	mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME);
 
 	if (ucontrol->value.integer.value[0])
 		snd_soc_dapm_enable_pin(&card->dapm, pin);
 	else
 		snd_soc_dapm_disable_pin(&card->dapm, pin);
 
+	mutex_unlock(&card->dapm_mutex);
+
 	snd_soc_dapm_sync(&card->dapm);
-
-	mutex_unlock(&card->mutex);
-
 	return 0;
 }
 EXPORT_SYMBOL_GPL(snd_soc_dapm_put_pin_switch);
@@ -2717,9 +2865,9 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm,
 
 	switch (w->id) {
 	case snd_soc_dapm_regulator_supply:
-		w->priv = devm_regulator_get(dapm->dev, w->name);
-		if (IS_ERR(w->priv)) {
-			ret = PTR_ERR(w->priv);
+		w->regulator = devm_regulator_get(dapm->dev, w->name);
+		if (IS_ERR(w->regulator)) {
+			ret = PTR_ERR(w->regulator);
 			dev_err(dapm->dev, "Failed to request %s: %d\n",
 				w->name, ret);
 			return NULL;
@@ -2771,6 +2919,7 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm,
 	case snd_soc_dapm_hp:
 	case snd_soc_dapm_mic:
 	case snd_soc_dapm_line:
+	case snd_soc_dapm_dai_link:
 		w->power_check = dapm_generic_check_power;
 		break;
 	case snd_soc_dapm_supply:
@@ -2816,21 +2965,177 @@ int snd_soc_dapm_new_controls(struct snd_soc_dapm_context *dapm,
 {
 	struct snd_soc_dapm_widget *w;
 	int i;
+	int ret = 0;
 
+	mutex_lock_nested(&dapm->card->dapm_mutex, SND_SOC_DAPM_CLASS_INIT);
 	for (i = 0; i < num; i++) {
 		w = snd_soc_dapm_new_control(dapm, widget);
 		if (!w) {
 			dev_err(dapm->dev,
 				"ASoC: Failed to create DAPM control %s\n",
 				widget->name);
-			return -ENOMEM;
+			ret = -ENOMEM;
+			break;
 		}
 		widget++;
 	}
-	return 0;
+	mutex_unlock(&dapm->card->dapm_mutex);
+	return ret;
 }
 EXPORT_SYMBOL_GPL(snd_soc_dapm_new_controls);
 
+static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w,
+				  struct snd_kcontrol *kcontrol, int event)
+{
+	struct snd_soc_dapm_path *source_p, *sink_p;
+	struct snd_soc_dai *source, *sink;
+	const struct snd_soc_pcm_stream *config = w->params;
+	struct snd_pcm_substream substream;
+	struct snd_pcm_hw_params *params = NULL;
+	u64 fmt;
+	int ret;
+
+	BUG_ON(!config);
+	BUG_ON(list_empty(&w->sources) || list_empty(&w->sinks));
+
+	/* We only support a single source and sink, pick the first */
+	source_p = list_first_entry(&w->sources, struct snd_soc_dapm_path,
+				    list_sink);
+	sink_p = list_first_entry(&w->sinks, struct snd_soc_dapm_path,
+				  list_source);
+
+	BUG_ON(!source_p || !sink_p);
+	BUG_ON(!sink_p->source || !source_p->sink);
+	BUG_ON(!source_p->source || !sink_p->sink);
+
+	source = source_p->source->priv;
+	sink = sink_p->sink->priv;
+
+	/* Be a little careful as we don't want to overflow the mask array */
+	if (config->formats) {
+		fmt = ffs(config->formats) - 1;
+	} else {
+		dev_warn(w->dapm->dev, "Invalid format %llx specified\n",
+			 config->formats);
+		fmt = 0;
+	}
+
+	/* Currently very limited parameter selection */
+	params = kzalloc(sizeof(*params), GFP_KERNEL);
+	if (!params) {
+		ret = -ENOMEM;
+		goto out;
+	}
+	snd_mask_set(hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT), fmt);
+
+	hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE)->min =
+		config->rate_min;
+	hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE)->max =
+		config->rate_max;
+
+	hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS)->min
+		= config->channels_min;
+	hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS)->max
+		= config->channels_max;
+
+	memset(&substream, 0, sizeof(substream));
+
+	switch (event) {
+	case SND_SOC_DAPM_PRE_PMU:
+		if (source->driver->ops && source->driver->ops->hw_params) {
+			substream.stream = SNDRV_PCM_STREAM_CAPTURE;
+			ret = source->driver->ops->hw_params(&substream,
+							     params, source);
+			if (ret != 0) {
+				dev_err(source->dev,
+					"hw_params() failed: %d\n", ret);
+				goto out;
+			}
+		}
+
+		if (sink->driver->ops && sink->driver->ops->hw_params) {
+			substream.stream = SNDRV_PCM_STREAM_PLAYBACK;
+			ret = sink->driver->ops->hw_params(&substream, params,
+							   sink);
+			if (ret != 0) {
+				dev_err(sink->dev,
+					"hw_params() failed: %d\n", ret);
+				goto out;
+			}
+		}
+		break;
+
+	case SND_SOC_DAPM_POST_PMU:
+		ret = snd_soc_dai_digital_mute(sink, 0);
+		if (ret != 0 && ret != -ENOTSUPP)
+			dev_warn(sink->dev, "Failed to unmute: %d\n", ret);
+		ret = 0;
+		break;
+
+	case SND_SOC_DAPM_PRE_PMD:
+		ret = snd_soc_dai_digital_mute(sink, 1);
+		if (ret != 0 && ret != -ENOTSUPP)
+			dev_warn(sink->dev, "Failed to mute: %d\n", ret);
+		ret = 0;
+		break;
+
+	default:
+		BUG();
+		return -EINVAL;
+	}
+
+out:
+	kfree(params);
+	return ret;
+}
+
+int snd_soc_dapm_new_pcm(struct snd_soc_card *card,
+			 const struct snd_soc_pcm_stream *params,
+			 struct snd_soc_dapm_widget *source,
+			 struct snd_soc_dapm_widget *sink)
+{
+	struct snd_soc_dapm_route routes[2];
+	struct snd_soc_dapm_widget template;
+	struct snd_soc_dapm_widget *w;
+	size_t len;
+	char *link_name;
+
+	len = strlen(source->name) + strlen(sink->name) + 2;
+	link_name = devm_kzalloc(card->dev, len, GFP_KERNEL);
+	if (!link_name)
+		return -ENOMEM;
+	snprintf(link_name, len, "%s-%s", source->name, sink->name);
+
+	memset(&template, 0, sizeof(template));
+	template.reg = SND_SOC_NOPM;
+	template.id = snd_soc_dapm_dai_link;
+	template.name = link_name;
+	template.event = snd_soc_dai_link_event;
+	template.event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU |
+		SND_SOC_DAPM_PRE_PMD;
+
+	dev_dbg(card->dev, "adding %s widget\n", link_name);
+
+	w = snd_soc_dapm_new_control(&card->dapm, &template);
+	if (!w) {
+		dev_err(card->dev, "Failed to create %s widget\n",
+			link_name);
+		return -ENOMEM;
+	}
+
+	w->params = params;
+
+	memset(&routes, 0, sizeof(routes));
+
+	routes[0].source = source->name;
+	routes[0].sink = link_name;
+	routes[1].source = link_name;
+	routes[1].sink = sink->name;
+
+	return snd_soc_dapm_add_routes(&card->dapm, routes,
+				       ARRAY_SIZE(routes));
+}
+
 int snd_soc_dapm_new_dai_widgets(struct snd_soc_dapm_context *dapm,
 				 struct snd_soc_dai *dai)
 {
@@ -2934,37 +3239,61 @@ int snd_soc_dapm_link_dai_widgets(struct snd_soc_card *card)
 	return 0;
 }
 
-static void soc_dapm_stream_event(struct snd_soc_dapm_context *dapm,
-				  int stream, struct snd_soc_dai *dai,
-				  int event)
+static void soc_dapm_stream_event(struct snd_soc_pcm_runtime *rtd, int stream,
+	int event)
 {
-	struct snd_soc_dapm_widget *w;
 
-	if (stream == SNDRV_PCM_STREAM_PLAYBACK)
-		w = dai->playback_widget;
-	else
-		w = dai->capture_widget;
+	struct snd_soc_dapm_widget *w_cpu, *w_codec;
+	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	struct snd_soc_dai *codec_dai = rtd->codec_dai;
 
-	if (!w)
-		return;
-
-	dapm_mark_dirty(w, "stream event");
-
-	switch (event) {
-	case SND_SOC_DAPM_STREAM_START:
-		w->active = 1;
-		break;
-	case SND_SOC_DAPM_STREAM_STOP:
-		w->active = 0;
-		break;
-	case SND_SOC_DAPM_STREAM_SUSPEND:
-	case SND_SOC_DAPM_STREAM_RESUME:
-	case SND_SOC_DAPM_STREAM_PAUSE_PUSH:
-	case SND_SOC_DAPM_STREAM_PAUSE_RELEASE:
-		break;
+	if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		w_cpu = cpu_dai->playback_widget;
+		w_codec = codec_dai->playback_widget;
+	} else {
+		w_cpu = cpu_dai->capture_widget;
+		w_codec = codec_dai->capture_widget;
 	}
 
-	dapm_power_widgets(dapm, event);
+	if (w_cpu) {
+
+		dapm_mark_dirty(w_cpu, "stream event");
+
+		switch (event) {
+		case SND_SOC_DAPM_STREAM_START:
+			w_cpu->active = 1;
+			break;
+		case SND_SOC_DAPM_STREAM_STOP:
+			w_cpu->active = 0;
+			break;
+		case SND_SOC_DAPM_STREAM_SUSPEND:
+		case SND_SOC_DAPM_STREAM_RESUME:
+		case SND_SOC_DAPM_STREAM_PAUSE_PUSH:
+		case SND_SOC_DAPM_STREAM_PAUSE_RELEASE:
+			break;
+		}
+	}
+
+	if (w_codec) {
+
+		dapm_mark_dirty(w_codec, "stream event");
+
+		switch (event) {
+		case SND_SOC_DAPM_STREAM_START:
+			w_codec->active = 1;
+			break;
+		case SND_SOC_DAPM_STREAM_STOP:
+			w_codec->active = 0;
+			break;
+		case SND_SOC_DAPM_STREAM_SUSPEND:
+		case SND_SOC_DAPM_STREAM_RESUME:
+		case SND_SOC_DAPM_STREAM_PAUSE_PUSH:
+		case SND_SOC_DAPM_STREAM_PAUSE_RELEASE:
+			break;
+		}
+	}
+
+	dapm_power_widgets(&rtd->card->dapm, event);
 }
 
 /**
@@ -2978,15 +3307,14 @@ static void soc_dapm_stream_event(struct snd_soc_dapm_context *dapm,
  *
  * Returns 0 for success else error.
  */
-int snd_soc_dapm_stream_event(struct snd_soc_pcm_runtime *rtd, int stream,
-			      struct snd_soc_dai *dai, int event)
+void snd_soc_dapm_stream_event(struct snd_soc_pcm_runtime *rtd, int stream,
+			      int event)
 {
-	struct snd_soc_codec *codec = rtd->codec;
+	struct snd_soc_card *card = rtd->card;
 
-	mutex_lock(&codec->mutex);
-	soc_dapm_stream_event(&codec->dapm, stream, dai, event);
-	mutex_unlock(&codec->mutex);
-	return 0;
+	mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME);
+	soc_dapm_stream_event(rtd, stream, event);
+	mutex_unlock(&card->dapm_mutex);
 }
 
 /**
diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c
index ee4353f843ea..7f8b3b7428bb 100644
--- a/sound/soc/soc-jack.c
+++ b/sound/soc/soc-jack.c
@@ -36,6 +36,7 @@
 int snd_soc_jack_new(struct snd_soc_codec *codec, const char *id, int type,
 		     struct snd_soc_jack *jack)
 {
+	mutex_init(&jack->mutex);
 	jack->codec = codec;
 	INIT_LIST_HEAD(&jack->pins);
 	INIT_LIST_HEAD(&jack->jack_zones);
@@ -75,7 +76,7 @@ void snd_soc_jack_report(struct snd_soc_jack *jack, int status, int mask)
 	codec = jack->codec;
 	dapm =  &codec->dapm;
 
-	mutex_lock(&codec->mutex);
+	mutex_lock(&jack->mutex);
 
 	oldstatus = jack->status;
 
@@ -109,7 +110,7 @@ void snd_soc_jack_report(struct snd_soc_jack *jack, int status, int mask)
 	snd_jack_report(jack->jack, jack->status);
 
 out:
-	mutex_unlock(&codec->mutex);
+	mutex_unlock(&jack->mutex);
 }
 EXPORT_SYMBOL_GPL(snd_soc_jack_report);
 
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index 0ad8dcacd2f3..bedd1717a373 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -22,12 +22,38 @@
 #include <linux/pm_runtime.h>
 #include <linux/slab.h>
 #include <linux/workqueue.h>
+#include <linux/export.h>
+#include <linux/debugfs.h>
 #include <sound/core.h>
 #include <sound/pcm.h>
 #include <sound/pcm_params.h>
 #include <sound/soc.h>
+#include <sound/soc-dpcm.h>
 #include <sound/initval.h>
 
+#define DPCM_MAX_BE_USERS	8
+
+/* DPCM stream event, send event to FE and all active BEs. */
+static int dpcm_dapm_stream_event(struct snd_soc_pcm_runtime *fe, int dir,
+	int event)
+{
+	struct snd_soc_dpcm *dpcm;
+
+	list_for_each_entry(dpcm, &fe->dpcm[dir].be_clients, list_be) {
+
+		struct snd_soc_pcm_runtime *be = dpcm->be;
+
+		dev_dbg(be->dev, "pm: BE %s event %d dir %d\n",
+				be->dai_link->name, event, dir);
+
+		snd_soc_dapm_stream_event(be, dir, event);
+	}
+
+	snd_soc_dapm_stream_event(fe, dir, event);
+
+	return 0;
+}
+
 static int soc_pcm_apply_symmetry(struct snd_pcm_substream *substream,
 					struct snd_soc_dai *soc_dai)
 {
@@ -156,6 +182,10 @@ static int soc_pcm_open(struct snd_pcm_substream *substream)
 		}
 	}
 
+	/* Dynamic PCM DAI links compat checks use dynamic capabilities */
+	if (rtd->dai_link->dynamic || rtd->dai_link->no_pcm)
+		goto dynamic;
+
 	/* Check that the codec and cpu DAIs are compatible */
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
 		runtime->hw.rate_min =
@@ -248,6 +278,7 @@ static int soc_pcm_open(struct snd_pcm_substream *substream)
 	pr_debug("asoc: min rate %d max rate %d\n", runtime->hw.rate_min,
 		 runtime->hw.rate_max);
 
+dynamic:
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
 		cpu_dai->playback_active++;
 		codec_dai->playback_active++;
@@ -308,7 +339,7 @@ static void close_delayed_work(struct work_struct *work)
 	if (codec_dai->pop_wait == 1) {
 		codec_dai->pop_wait = 0;
 		snd_soc_dapm_stream_event(rtd, SNDRV_PCM_STREAM_PLAYBACK,
-					  codec_dai, SND_SOC_DAPM_STREAM_STOP);
+					  SND_SOC_DAPM_STREAM_STOP);
 	}
 
 	mutex_unlock(&rtd->pcm_mutex);
@@ -373,7 +404,6 @@ static int soc_pcm_close(struct snd_pcm_substream *substream)
 			/* powered down playback stream now */
 			snd_soc_dapm_stream_event(rtd,
 						  SNDRV_PCM_STREAM_PLAYBACK,
-						  codec_dai,
 						  SND_SOC_DAPM_STREAM_STOP);
 		} else {
 			/* start delayed pop wq here for playback streams */
@@ -384,7 +414,7 @@ static int soc_pcm_close(struct snd_pcm_substream *substream)
 	} else {
 		/* capture streams can be powered down now */
 		snd_soc_dapm_stream_event(rtd, SNDRV_PCM_STREAM_CAPTURE,
-					  codec_dai, SND_SOC_DAPM_STREAM_STOP);
+					  SND_SOC_DAPM_STREAM_STOP);
 	}
 
 	mutex_unlock(&rtd->pcm_mutex);
@@ -453,8 +483,8 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream)
 		cancel_delayed_work(&rtd->delayed_work);
 	}
 
-	snd_soc_dapm_stream_event(rtd, substream->stream, codec_dai,
-				  SND_SOC_DAPM_STREAM_START);
+	snd_soc_dapm_stream_event(rtd, substream->stream,
+			SND_SOC_DAPM_STREAM_START);
 
 	snd_soc_dai_digital_mute(codec_dai, 0);
 
@@ -602,6 +632,34 @@ static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
 	return 0;
 }
 
+static int soc_pcm_bespoke_trigger(struct snd_pcm_substream *substream,
+				   int cmd)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_platform *platform = rtd->platform;
+	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	int ret;
+
+	if (codec_dai->driver->ops->bespoke_trigger) {
+		ret = codec_dai->driver->ops->bespoke_trigger(substream, cmd, codec_dai);
+		if (ret < 0)
+			return ret;
+	}
+
+	if (platform->driver->bespoke_trigger) {
+		ret = platform->driver->bespoke_trigger(substream, cmd);
+		if (ret < 0)
+			return ret;
+	}
+
+	if (cpu_dai->driver->ops->bespoke_trigger) {
+		ret = cpu_dai->driver->ops->bespoke_trigger(substream, cmd, cpu_dai);
+		if (ret < 0)
+			return ret;
+	}
+	return 0;
+}
 /*
  * soc level wrapper for pointer callback
  * If cpu_dai, codec_dai, platform driver has the delay callback, than
@@ -634,6 +692,1308 @@ static snd_pcm_uframes_t soc_pcm_pointer(struct snd_pcm_substream *substream)
 	return offset;
 }
 
+/* connect a FE and BE */
+static int dpcm_be_connect(struct snd_soc_pcm_runtime *fe,
+		struct snd_soc_pcm_runtime *be, int stream)
+{
+	struct snd_soc_dpcm *dpcm;
+
+	/* only add new dpcms */
+	list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) {
+		if (dpcm->be == be && dpcm->fe == fe)
+			return 0;
+	}
+
+	dpcm = kzalloc(sizeof(struct snd_soc_dpcm), GFP_KERNEL);
+	if (!dpcm)
+		return -ENOMEM;
+
+	dpcm->be = be;
+	dpcm->fe = fe;
+	be->dpcm[stream].runtime = fe->dpcm[stream].runtime;
+	dpcm->state = SND_SOC_DPCM_LINK_STATE_NEW;
+	list_add(&dpcm->list_be, &fe->dpcm[stream].be_clients);
+	list_add(&dpcm->list_fe, &be->dpcm[stream].fe_clients);
+
+	dev_dbg(fe->dev, "  connected new DPCM %s path %s %s %s\n",
+			stream ? "capture" : "playback",  fe->dai_link->name,
+			stream ? "<-" : "->", be->dai_link->name);
+
+#ifdef CONFIG_DEBUG_FS
+	dpcm->debugfs_state = debugfs_create_u32(be->dai_link->name, 0644,
+			fe->debugfs_dpcm_root, &dpcm->state);
+#endif
+	return 1;
+}
+
+/* reparent a BE onto another FE */
+static void dpcm_be_reparent(struct snd_soc_pcm_runtime *fe,
+			struct snd_soc_pcm_runtime *be, int stream)
+{
+	struct snd_soc_dpcm *dpcm;
+	struct snd_pcm_substream *fe_substream, *be_substream;
+
+	/* reparent if BE is connected to other FEs */
+	if (!be->dpcm[stream].users)
+		return;
+
+	be_substream = snd_soc_dpcm_get_substream(be, stream);
+
+	list_for_each_entry(dpcm, &be->dpcm[stream].fe_clients, list_fe) {
+		if (dpcm->fe == fe)
+			continue;
+
+		dev_dbg(fe->dev, "  reparent %s path %s %s %s\n",
+			stream ? "capture" : "playback",
+			dpcm->fe->dai_link->name,
+			stream ? "<-" : "->", dpcm->be->dai_link->name);
+
+		fe_substream = snd_soc_dpcm_get_substream(dpcm->fe, stream);
+		be_substream->runtime = fe_substream->runtime;
+		break;
+	}
+}
+
+/* disconnect a BE and FE */
+static void dpcm_be_disconnect(struct snd_soc_pcm_runtime *fe, int stream)
+{
+	struct snd_soc_dpcm *dpcm, *d;
+
+	list_for_each_entry_safe(dpcm, d, &fe->dpcm[stream].be_clients, list_be) {
+		dev_dbg(fe->dev, "BE %s disconnect check for %s\n",
+				stream ? "capture" : "playback",
+				dpcm->be->dai_link->name);
+
+		if (dpcm->state != SND_SOC_DPCM_LINK_STATE_FREE)
+			continue;
+
+		dev_dbg(fe->dev, "  freed DSP %s path %s %s %s\n",
+			stream ? "capture" : "playback", fe->dai_link->name,
+			stream ? "<-" : "->", dpcm->be->dai_link->name);
+
+		/* BEs still alive need new FE */
+		dpcm_be_reparent(fe, dpcm->be, stream);
+
+#ifdef CONFIG_DEBUG_FS
+		debugfs_remove(dpcm->debugfs_state);
+#endif
+		list_del(&dpcm->list_be);
+		list_del(&dpcm->list_fe);
+		kfree(dpcm);
+	}
+}
+
+/* get BE for DAI widget and stream */
+static struct snd_soc_pcm_runtime *dpcm_get_be(struct snd_soc_card *card,
+		struct snd_soc_dapm_widget *widget, int stream)
+{
+	struct snd_soc_pcm_runtime *be;
+	int i;
+
+	if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		for (i = 0; i < card->num_links; i++) {
+			be = &card->rtd[i];
+
+			if (be->cpu_dai->playback_widget == widget ||
+				be->codec_dai->playback_widget == widget)
+				return be;
+		}
+	} else {
+
+		for (i = 0; i < card->num_links; i++) {
+			be = &card->rtd[i];
+
+			if (be->cpu_dai->capture_widget == widget ||
+				be->codec_dai->capture_widget == widget)
+				return be;
+		}
+	}
+
+	dev_err(card->dev, "can't get %s BE for %s\n",
+		stream ? "capture" : "playback", widget->name);
+	return NULL;
+}
+
+static inline struct snd_soc_dapm_widget *
+	rtd_get_cpu_widget(struct snd_soc_pcm_runtime *rtd, int stream)
+{
+	if (stream == SNDRV_PCM_STREAM_PLAYBACK)
+		return rtd->cpu_dai->playback_widget;
+	else
+		return rtd->cpu_dai->capture_widget;
+}
+
+static inline struct snd_soc_dapm_widget *
+	rtd_get_codec_widget(struct snd_soc_pcm_runtime *rtd, int stream)
+{
+	if (stream == SNDRV_PCM_STREAM_PLAYBACK)
+		return rtd->codec_dai->playback_widget;
+	else
+		return rtd->codec_dai->capture_widget;
+}
+
+static int widget_in_list(struct snd_soc_dapm_widget_list *list,
+		struct snd_soc_dapm_widget *widget)
+{
+	int i;
+
+	for (i = 0; i < list->num_widgets; i++) {
+		if (widget == list->widgets[i])
+			return 1;
+	}
+
+	return 0;
+}
+
+static int dpcm_path_get(struct snd_soc_pcm_runtime *fe,
+	int stream, struct snd_soc_dapm_widget_list **list_)
+{
+	struct snd_soc_dai *cpu_dai = fe->cpu_dai;
+	struct snd_soc_dapm_widget_list *list;
+	int paths;
+
+	list = kzalloc(sizeof(struct snd_soc_dapm_widget_list) +
+			sizeof(struct snd_soc_dapm_widget *), GFP_KERNEL);
+	if (list == NULL)
+		return -ENOMEM;
+
+	/* get number of valid DAI paths and their widgets */
+	paths = snd_soc_dapm_dai_get_connected_widgets(cpu_dai, stream, &list);
+
+	dev_dbg(fe->dev, "found %d audio %s paths\n", paths,
+			stream ? "capture" : "playback");
+
+	*list_ = list;
+	return paths;
+}
+
+static inline void dpcm_path_put(struct snd_soc_dapm_widget_list **list)
+{
+	kfree(*list);
+}
+
+static int dpcm_prune_paths(struct snd_soc_pcm_runtime *fe, int stream,
+	struct snd_soc_dapm_widget_list **list_)
+{
+	struct snd_soc_dpcm *dpcm;
+	struct snd_soc_dapm_widget_list *list = *list_;
+	struct snd_soc_dapm_widget *widget;
+	int prune = 0;
+
+	/* Destroy any old FE <--> BE connections */
+	list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) {
+
+		/* is there a valid CPU DAI widget for this BE */
+		widget = rtd_get_cpu_widget(dpcm->be, stream);
+
+		/* prune the BE if it's no longer in our active list */
+		if (widget && widget_in_list(list, widget))
+			continue;
+
+		/* is there a valid CODEC DAI widget for this BE */
+		widget = rtd_get_codec_widget(dpcm->be, stream);
+
+		/* prune the BE if it's no longer in our active list */
+		if (widget && widget_in_list(list, widget))
+			continue;
+
+		dev_dbg(fe->dev, "pruning %s BE %s for %s\n",
+			stream ? "capture" : "playback",
+			dpcm->be->dai_link->name, fe->dai_link->name);
+		dpcm->state = SND_SOC_DPCM_LINK_STATE_FREE;
+		dpcm->be->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_BE;
+		prune++;
+	}
+
+	dev_dbg(fe->dev, "found %d old BE paths for pruning\n", prune);
+	return prune;
+}
+
+static int dpcm_add_paths(struct snd_soc_pcm_runtime *fe, int stream,
+	struct snd_soc_dapm_widget_list **list_)
+{
+	struct snd_soc_card *card = fe->card;
+	struct snd_soc_dapm_widget_list *list = *list_;
+	struct snd_soc_pcm_runtime *be;
+	int i, new = 0, err;
+
+	/* Create any new FE <--> BE connections */
+	for (i = 0; i < list->num_widgets; i++) {
+
+		if (list->widgets[i]->id != snd_soc_dapm_dai)
+			continue;
+
+		/* is there a valid BE rtd for this widget */
+		be = dpcm_get_be(card, list->widgets[i], stream);
+		if (!be) {
+			dev_err(fe->dev, "no BE found for %s\n",
+					list->widgets[i]->name);
+			continue;
+		}
+
+		/* make sure BE is a real BE */
+		if (!be->dai_link->no_pcm)
+			continue;
+
+		/* don't connect if FE is not running */
+		if (!fe->dpcm[stream].runtime)
+			continue;
+
+		/* newly connected FE and BE */
+		err = dpcm_be_connect(fe, be, stream);
+		if (err < 0) {
+			dev_err(fe->dev, "can't connect %s\n",
+				list->widgets[i]->name);
+			break;
+		} else if (err == 0) /* already connected */
+			continue;
+
+		/* new */
+		be->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_BE;
+		new++;
+	}
+
+	dev_dbg(fe->dev, "found %d new BE paths\n", new);
+	return new;
+}
+
+/*
+ * Find the corresponding BE DAIs that source or sink audio to this
+ * FE substream.
+ */
+static int dpcm_process_paths(struct snd_soc_pcm_runtime *fe,
+	int stream, struct snd_soc_dapm_widget_list **list, int new)
+{
+	if (new)
+		return dpcm_add_paths(fe, stream, list);
+	else
+		return dpcm_prune_paths(fe, stream, list);
+}
+
+static void dpcm_clear_pending_state(struct snd_soc_pcm_runtime *fe, int stream)
+{
+	struct snd_soc_dpcm *dpcm;
+
+	list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be)
+		dpcm->be->dpcm[stream].runtime_update =
+						SND_SOC_DPCM_UPDATE_NO;
+}
+
+static void dpcm_be_dai_startup_unwind(struct snd_soc_pcm_runtime *fe,
+	int stream)
+{
+	struct snd_soc_dpcm *dpcm;
+
+	/* disable any enabled and non active backends */
+	list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) {
+
+		struct snd_soc_pcm_runtime *be = dpcm->be;
+		struct snd_pcm_substream *be_substream =
+			snd_soc_dpcm_get_substream(be, stream);
+
+		if (be->dpcm[stream].users == 0)
+			dev_err(be->dev, "no users %s at close - state %d\n",
+				stream ? "capture" : "playback",
+				be->dpcm[stream].state);
+
+		if (--be->dpcm[stream].users != 0)
+			continue;
+
+		if (be->dpcm[stream].state != SND_SOC_DPCM_STATE_OPEN)
+			continue;
+
+		soc_pcm_close(be_substream);
+		be_substream->runtime = NULL;
+		be->dpcm[stream].state = SND_SOC_DPCM_STATE_CLOSE;
+	}
+}
+
+static int dpcm_be_dai_startup(struct snd_soc_pcm_runtime *fe, int stream)
+{
+	struct snd_soc_dpcm *dpcm;
+	int err, count = 0;
+
+	/* only startup BE DAIs that are either sinks or sources to this FE DAI */
+	list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) {
+
+		struct snd_soc_pcm_runtime *be = dpcm->be;
+		struct snd_pcm_substream *be_substream =
+			snd_soc_dpcm_get_substream(be, stream);
+
+		/* is this op for this BE ? */
+		if (!snd_soc_dpcm_be_can_update(fe, be, stream))
+			continue;
+
+		/* first time the dpcm is open ? */
+		if (be->dpcm[stream].users == DPCM_MAX_BE_USERS)
+			dev_err(be->dev, "too many users %s at open %d\n",
+				stream ? "capture" : "playback",
+				be->dpcm[stream].state);
+
+		if (be->dpcm[stream].users++ != 0)
+			continue;
+
+		if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_NEW) &&
+		    (be->dpcm[stream].state != SND_SOC_DPCM_STATE_CLOSE))
+			continue;
+
+		dev_dbg(be->dev, "dpcm: open BE %s\n", be->dai_link->name);
+
+		be_substream->runtime = be->dpcm[stream].runtime;
+		err = soc_pcm_open(be_substream);
+		if (err < 0) {
+			dev_err(be->dev, "BE open failed %d\n", err);
+			be->dpcm[stream].users--;
+			if (be->dpcm[stream].users < 0)
+				dev_err(be->dev, "no users %s at unwind %d\n",
+					stream ? "capture" : "playback",
+					be->dpcm[stream].state);
+
+			be->dpcm[stream].state = SND_SOC_DPCM_STATE_CLOSE;
+			goto unwind;
+		}
+
+		be->dpcm[stream].state = SND_SOC_DPCM_STATE_OPEN;
+		count++;
+	}
+
+	return count;
+
+unwind:
+	/* disable any enabled and non active backends */
+	list_for_each_entry_continue_reverse(dpcm, &fe->dpcm[stream].be_clients, list_be) {
+		struct snd_soc_pcm_runtime *be = dpcm->be;
+		struct snd_pcm_substream *be_substream =
+			snd_soc_dpcm_get_substream(be, stream);
+
+		if (!snd_soc_dpcm_be_can_update(fe, be, stream))
+			continue;
+
+		if (be->dpcm[stream].users == 0)
+			dev_err(be->dev, "no users %s at close %d\n",
+				stream ? "capture" : "playback",
+				be->dpcm[stream].state);
+
+		if (--be->dpcm[stream].users != 0)
+			continue;
+
+		if (be->dpcm[stream].state != SND_SOC_DPCM_STATE_OPEN)
+			continue;
+
+		soc_pcm_close(be_substream);
+		be_substream->runtime = NULL;
+		be->dpcm[stream].state = SND_SOC_DPCM_STATE_CLOSE;
+	}
+
+	return err;
+}
+
+static void dpcm_set_fe_runtime(struct snd_pcm_substream *substream)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	struct snd_soc_dai_driver *cpu_dai_drv = cpu_dai->driver;
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		runtime->hw.rate_min = cpu_dai_drv->playback.rate_min;
+		runtime->hw.rate_max = cpu_dai_drv->playback.rate_max;
+		runtime->hw.channels_min = cpu_dai_drv->playback.channels_min;
+		runtime->hw.channels_max = cpu_dai_drv->playback.channels_max;
+		runtime->hw.formats &= cpu_dai_drv->playback.formats;
+		runtime->hw.rates = cpu_dai_drv->playback.rates;
+	} else {
+		runtime->hw.rate_min = cpu_dai_drv->capture.rate_min;
+		runtime->hw.rate_max = cpu_dai_drv->capture.rate_max;
+		runtime->hw.channels_min = cpu_dai_drv->capture.channels_min;
+		runtime->hw.channels_max = cpu_dai_drv->capture.channels_max;
+		runtime->hw.formats &= cpu_dai_drv->capture.formats;
+		runtime->hw.rates = cpu_dai_drv->capture.rates;
+	}
+}
+
+static int dpcm_fe_dai_startup(struct snd_pcm_substream *fe_substream)
+{
+	struct snd_soc_pcm_runtime *fe = fe_substream->private_data;
+	struct snd_pcm_runtime *runtime = fe_substream->runtime;
+	int stream = fe_substream->stream, ret = 0;
+
+	fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_FE;
+
+	ret = dpcm_be_dai_startup(fe, fe_substream->stream);
+	if (ret < 0) {
+		dev_err(fe->dev,"dpcm: failed to start some BEs %d\n", ret);
+		goto be_err;
+	}
+
+	dev_dbg(fe->dev, "dpcm: open FE %s\n", fe->dai_link->name);
+
+	/* start the DAI frontend */
+	ret = soc_pcm_open(fe_substream);
+	if (ret < 0) {
+		dev_err(fe->dev,"dpcm: failed to start FE %d\n", ret);
+		goto unwind;
+	}
+
+	fe->dpcm[stream].state = SND_SOC_DPCM_STATE_OPEN;
+
+	dpcm_set_fe_runtime(fe_substream);
+	snd_pcm_limit_hw_rates(runtime);
+
+	fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO;
+	return 0;
+
+unwind:
+	dpcm_be_dai_startup_unwind(fe, fe_substream->stream);
+be_err:
+	fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO;
+	return ret;
+}
+
+static int dpcm_be_dai_shutdown(struct snd_soc_pcm_runtime *fe, int stream)
+{
+	struct snd_soc_dpcm *dpcm;
+
+	/* only shutdown BEs that are either sinks or sources to this FE DAI */
+	list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) {
+
+		struct snd_soc_pcm_runtime *be = dpcm->be;
+		struct snd_pcm_substream *be_substream =
+			snd_soc_dpcm_get_substream(be, stream);
+
+		/* is this op for this BE ? */
+		if (!snd_soc_dpcm_be_can_update(fe, be, stream))
+			continue;
+
+		if (be->dpcm[stream].users == 0)
+			dev_err(be->dev, "no users %s at close - state %d\n",
+				stream ? "capture" : "playback",
+				be->dpcm[stream].state);
+
+		if (--be->dpcm[stream].users != 0)
+			continue;
+
+		if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_HW_FREE) &&
+		    (be->dpcm[stream].state != SND_SOC_DPCM_STATE_OPEN))
+			continue;
+
+		dev_dbg(be->dev, "dpcm: close BE %s\n",
+			dpcm->fe->dai_link->name);
+
+		soc_pcm_close(be_substream);
+		be_substream->runtime = NULL;
+
+		be->dpcm[stream].state = SND_SOC_DPCM_STATE_CLOSE;
+	}
+	return 0;
+}
+
+static int dpcm_fe_dai_shutdown(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *fe = substream->private_data;
+	int stream = substream->stream;
+
+	fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_FE;
+
+	/* shutdown the BEs */
+	dpcm_be_dai_shutdown(fe, substream->stream);
+
+	dev_dbg(fe->dev, "dpcm: close FE %s\n", fe->dai_link->name);
+
+	/* now shutdown the frontend */
+	soc_pcm_close(substream);
+
+	/* run the stream event for each BE */
+	dpcm_dapm_stream_event(fe, stream, SND_SOC_DAPM_STREAM_STOP);
+
+	fe->dpcm[stream].state = SND_SOC_DPCM_STATE_CLOSE;
+	fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO;
+	return 0;
+}
+
+static int dpcm_be_dai_hw_free(struct snd_soc_pcm_runtime *fe, int stream)
+{
+	struct snd_soc_dpcm *dpcm;
+
+	/* only hw_params backends that are either sinks or sources
+	 * to this frontend DAI */
+	list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) {
+
+		struct snd_soc_pcm_runtime *be = dpcm->be;
+		struct snd_pcm_substream *be_substream =
+			snd_soc_dpcm_get_substream(be, stream);
+
+		/* is this op for this BE ? */
+		if (!snd_soc_dpcm_be_can_update(fe, be, stream))
+			continue;
+
+		/* only free hw when no longer used - check all FEs */
+		if (!snd_soc_dpcm_can_be_free_stop(fe, be, stream))
+				continue;
+
+		if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_HW_PARAMS) &&
+		    (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PREPARE) &&
+		    (be->dpcm[stream].state != SND_SOC_DPCM_STATE_HW_FREE) &&
+		    (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP))
+			continue;
+
+		dev_dbg(be->dev, "dpcm: hw_free BE %s\n",
+			dpcm->fe->dai_link->name);
+
+		soc_pcm_hw_free(be_substream);
+
+		be->dpcm[stream].state = SND_SOC_DPCM_STATE_HW_FREE;
+	}
+
+	return 0;
+}
+
+static int dpcm_fe_dai_hw_free(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *fe = substream->private_data;
+	int err, stream = substream->stream;
+
+	mutex_lock_nested(&fe->card->mutex, SND_SOC_CARD_CLASS_RUNTIME);
+	fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_FE;
+
+	dev_dbg(fe->dev, "dpcm: hw_free FE %s\n", fe->dai_link->name);
+
+	/* call hw_free on the frontend */
+	err = soc_pcm_hw_free(substream);
+	if (err < 0)
+		dev_err(fe->dev,"dpcm: hw_free FE %s failed\n",
+			fe->dai_link->name);
+
+	/* only hw_params backends that are either sinks or sources
+	 * to this frontend DAI */
+	err = dpcm_be_dai_hw_free(fe, stream);
+
+	fe->dpcm[stream].state = SND_SOC_DPCM_STATE_HW_FREE;
+	fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO;
+
+	mutex_unlock(&fe->card->mutex);
+	return 0;
+}
+
+static int dpcm_be_dai_hw_params(struct snd_soc_pcm_runtime *fe, int stream)
+{
+	struct snd_soc_dpcm *dpcm;
+	int ret;
+
+	list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) {
+
+		struct snd_soc_pcm_runtime *be = dpcm->be;
+		struct snd_pcm_substream *be_substream =
+			snd_soc_dpcm_get_substream(be, stream);
+
+		/* is this op for this BE ? */
+		if (!snd_soc_dpcm_be_can_update(fe, be, stream))
+			continue;
+
+		/* only allow hw_params() if no connected FEs are running */
+		if (!snd_soc_dpcm_can_be_params(fe, be, stream))
+			continue;
+
+		if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_OPEN) &&
+		    (be->dpcm[stream].state != SND_SOC_DPCM_STATE_HW_PARAMS) &&
+		    (be->dpcm[stream].state != SND_SOC_DPCM_STATE_HW_FREE))
+			continue;
+
+		dev_dbg(be->dev, "dpcm: hw_params BE %s\n",
+			dpcm->fe->dai_link->name);
+
+		/* copy params for each dpcm */
+		memcpy(&dpcm->hw_params, &fe->dpcm[stream].hw_params,
+				sizeof(struct snd_pcm_hw_params));
+
+		/* perform any hw_params fixups */
+		if (be->dai_link->be_hw_params_fixup) {
+			ret = be->dai_link->be_hw_params_fixup(be,
+					&dpcm->hw_params);
+			if (ret < 0) {
+				dev_err(be->dev,
+					"dpcm: hw_params BE fixup failed %d\n",
+					ret);
+				goto unwind;
+			}
+		}
+
+		ret = soc_pcm_hw_params(be_substream, &dpcm->hw_params);
+		if (ret < 0) {
+			dev_err(dpcm->be->dev,
+				"dpcm: hw_params BE failed %d\n", ret);
+			goto unwind;
+		}
+
+		be->dpcm[stream].state = SND_SOC_DPCM_STATE_HW_PARAMS;
+	}
+	return 0;
+
+unwind:
+	/* disable any enabled and non active backends */
+	list_for_each_entry_continue_reverse(dpcm, &fe->dpcm[stream].be_clients, list_be) {
+		struct snd_soc_pcm_runtime *be = dpcm->be;
+		struct snd_pcm_substream *be_substream =
+			snd_soc_dpcm_get_substream(be, stream);
+
+		if (!snd_soc_dpcm_be_can_update(fe, be, stream))
+			continue;
+
+		/* only allow hw_free() if no connected FEs are running */
+		if (!snd_soc_dpcm_can_be_free_stop(fe, be, stream))
+			continue;
+
+		if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_OPEN) &&
+		   (be->dpcm[stream].state != SND_SOC_DPCM_STATE_HW_PARAMS) &&
+		   (be->dpcm[stream].state != SND_SOC_DPCM_STATE_HW_FREE) &&
+		   (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP))
+			continue;
+
+		soc_pcm_hw_free(be_substream);
+	}
+
+	return ret;
+}
+
+static int dpcm_fe_dai_hw_params(struct snd_pcm_substream *substream,
+				 struct snd_pcm_hw_params *params)
+{
+	struct snd_soc_pcm_runtime *fe = substream->private_data;
+	int ret, stream = substream->stream;
+
+	mutex_lock_nested(&fe->card->mutex, SND_SOC_CARD_CLASS_RUNTIME);
+	fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_FE;
+
+	memcpy(&fe->dpcm[substream->stream].hw_params, params,
+			sizeof(struct snd_pcm_hw_params));
+	ret = dpcm_be_dai_hw_params(fe, substream->stream);
+	if (ret < 0) {
+		dev_err(fe->dev,"dpcm: hw_params BE failed %d\n", ret);
+		goto out;
+	}
+
+	dev_dbg(fe->dev, "dpcm: hw_params FE %s rate %d chan %x fmt %d\n",
+			fe->dai_link->name, params_rate(params),
+			params_channels(params), params_format(params));
+
+	/* call hw_params on the frontend */
+	ret = soc_pcm_hw_params(substream, params);
+	if (ret < 0) {
+		dev_err(fe->dev,"dpcm: hw_params FE failed %d\n", ret);
+		dpcm_be_dai_hw_free(fe, stream);
+	 } else
+		fe->dpcm[stream].state = SND_SOC_DPCM_STATE_HW_PARAMS;
+
+out:
+	fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO;
+	mutex_unlock(&fe->card->mutex);
+	return ret;
+}
+
+static int dpcm_do_trigger(struct snd_soc_dpcm *dpcm,
+		struct snd_pcm_substream *substream, int cmd)
+{
+	int ret;
+
+	dev_dbg(dpcm->be->dev, "dpcm: trigger BE %s cmd %d\n",
+			dpcm->fe->dai_link->name, cmd);
+
+	ret = soc_pcm_trigger(substream, cmd);
+	if (ret < 0)
+		dev_err(dpcm->be->dev,"dpcm: trigger BE failed %d\n", ret);
+
+	return ret;
+}
+
+static int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream,
+			       int cmd)
+{
+	struct snd_soc_dpcm *dpcm;
+	int ret = 0;
+
+	list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) {
+
+		struct snd_soc_pcm_runtime *be = dpcm->be;
+		struct snd_pcm_substream *be_substream =
+			snd_soc_dpcm_get_substream(be, stream);
+
+		/* is this op for this BE ? */
+		if (!snd_soc_dpcm_be_can_update(fe, be, stream))
+			continue;
+
+		switch (cmd) {
+		case SNDRV_PCM_TRIGGER_START:
+			if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_PREPARE) &&
+			    (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP))
+				continue;
+
+			ret = dpcm_do_trigger(dpcm, be_substream, cmd);
+			if (ret)
+				return ret;
+
+			be->dpcm[stream].state = SND_SOC_DPCM_STATE_START;
+			break;
+		case SNDRV_PCM_TRIGGER_RESUME:
+			if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_SUSPEND))
+				continue;
+
+			ret = dpcm_do_trigger(dpcm, be_substream, cmd);
+			if (ret)
+				return ret;
+
+			be->dpcm[stream].state = SND_SOC_DPCM_STATE_START;
+			break;
+		case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+			if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_PAUSED))
+				continue;
+
+			ret = dpcm_do_trigger(dpcm, be_substream, cmd);
+			if (ret)
+				return ret;
+
+			be->dpcm[stream].state = SND_SOC_DPCM_STATE_START;
+			break;
+		case SNDRV_PCM_TRIGGER_STOP:
+			if (be->dpcm[stream].state != SND_SOC_DPCM_STATE_START)
+				continue;
+
+			if (!snd_soc_dpcm_can_be_free_stop(fe, be, stream))
+				continue;
+
+			ret = dpcm_do_trigger(dpcm, be_substream, cmd);
+			if (ret)
+				return ret;
+
+			be->dpcm[stream].state = SND_SOC_DPCM_STATE_STOP;
+			break;
+		case SNDRV_PCM_TRIGGER_SUSPEND:
+			if (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP)
+				continue;
+
+			if (!snd_soc_dpcm_can_be_free_stop(fe, be, stream))
+				continue;
+
+			ret = dpcm_do_trigger(dpcm, be_substream, cmd);
+			if (ret)
+				return ret;
+
+			be->dpcm[stream].state = SND_SOC_DPCM_STATE_SUSPEND;
+			break;
+		case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+			if (be->dpcm[stream].state != SND_SOC_DPCM_STATE_START)
+				continue;
+
+			if (!snd_soc_dpcm_can_be_free_stop(fe, be, stream))
+				continue;
+
+			ret = dpcm_do_trigger(dpcm, be_substream, cmd);
+			if (ret)
+				return ret;
+
+			be->dpcm[stream].state = SND_SOC_DPCM_STATE_PAUSED;
+			break;
+		}
+	}
+
+	return ret;
+}
+EXPORT_SYMBOL_GPL(dpcm_be_dai_trigger);
+
+static int dpcm_fe_dai_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+	struct snd_soc_pcm_runtime *fe = substream->private_data;
+	int stream = substream->stream, ret;
+	enum snd_soc_dpcm_trigger trigger = fe->dai_link->trigger[stream];
+
+	fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_FE;
+
+	switch (trigger) {
+	case SND_SOC_DPCM_TRIGGER_PRE:
+		/* call trigger on the frontend before the backend. */
+
+		dev_dbg(fe->dev, "dpcm: pre trigger FE %s cmd %d\n",
+				fe->dai_link->name, cmd);
+
+		ret = soc_pcm_trigger(substream, cmd);
+		if (ret < 0) {
+			dev_err(fe->dev,"dpcm: trigger FE failed %d\n", ret);
+			goto out;
+		}
+
+		ret = dpcm_be_dai_trigger(fe, substream->stream, cmd);
+		break;
+	case SND_SOC_DPCM_TRIGGER_POST:
+		/* call trigger on the frontend after the backend. */
+
+		ret = dpcm_be_dai_trigger(fe, substream->stream, cmd);
+		if (ret < 0) {
+			dev_err(fe->dev,"dpcm: trigger FE failed %d\n", ret);
+			goto out;
+		}
+
+		dev_dbg(fe->dev, "dpcm: post trigger FE %s cmd %d\n",
+				fe->dai_link->name, cmd);
+
+		ret = soc_pcm_trigger(substream, cmd);
+		break;
+	case SND_SOC_DPCM_TRIGGER_BESPOKE:
+		/* bespoke trigger() - handles both FE and BEs */
+
+		dev_dbg(fe->dev, "dpcm: bespoke trigger FE %s cmd %d\n",
+				fe->dai_link->name, cmd);
+
+		ret = soc_pcm_bespoke_trigger(substream, cmd);
+		if (ret < 0) {
+			dev_err(fe->dev,"dpcm: trigger FE failed %d\n", ret);
+			goto out;
+		}
+		break;
+	default:
+		dev_err(fe->dev, "dpcm: invalid trigger cmd %d for %s\n", cmd,
+				fe->dai_link->name);
+		ret = -EINVAL;
+		goto out;
+	}
+
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_START:
+	case SNDRV_PCM_TRIGGER_RESUME:
+	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+		fe->dpcm[stream].state = SND_SOC_DPCM_STATE_START;
+		break;
+	case SNDRV_PCM_TRIGGER_STOP:
+	case SNDRV_PCM_TRIGGER_SUSPEND:
+	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+		fe->dpcm[stream].state = SND_SOC_DPCM_STATE_STOP;
+		break;
+	}
+
+out:
+	fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO;
+	return ret;
+}
+
+static int dpcm_be_dai_prepare(struct snd_soc_pcm_runtime *fe, int stream)
+{
+	struct snd_soc_dpcm *dpcm;
+	int ret = 0;
+
+	list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) {
+
+		struct snd_soc_pcm_runtime *be = dpcm->be;
+		struct snd_pcm_substream *be_substream =
+			snd_soc_dpcm_get_substream(be, stream);
+
+		/* is this op for this BE ? */
+		if (!snd_soc_dpcm_be_can_update(fe, be, stream))
+			continue;
+
+		if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_HW_PARAMS) &&
+		    (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP))
+			continue;
+
+		dev_dbg(be->dev, "dpcm: prepare BE %s\n",
+			dpcm->fe->dai_link->name);
+
+		ret = soc_pcm_prepare(be_substream);
+		if (ret < 0) {
+			dev_err(be->dev, "dpcm: backend prepare failed %d\n",
+				ret);
+			break;
+		}
+
+		be->dpcm[stream].state = SND_SOC_DPCM_STATE_PREPARE;
+	}
+	return ret;
+}
+
+static int dpcm_fe_dai_prepare(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *fe = substream->private_data;
+	int stream = substream->stream, ret = 0;
+
+	mutex_lock_nested(&fe->card->mutex, SND_SOC_CARD_CLASS_RUNTIME);
+
+	dev_dbg(fe->dev, "dpcm: prepare FE %s\n", fe->dai_link->name);
+
+	fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_FE;
+
+	/* there is no point preparing this FE if there are no BEs */
+	if (list_empty(&fe->dpcm[stream].be_clients)) {
+		dev_err(fe->dev, "dpcm: no backend DAIs enabled for %s\n",
+				fe->dai_link->name);
+		ret = -EINVAL;
+		goto out;
+	}
+
+	ret = dpcm_be_dai_prepare(fe, substream->stream);
+	if (ret < 0)
+		goto out;
+
+	/* call prepare on the frontend */
+	ret = soc_pcm_prepare(substream);
+	if (ret < 0) {
+		dev_err(fe->dev,"dpcm: prepare FE %s failed\n",
+			fe->dai_link->name);
+		goto out;
+	}
+
+	/* run the stream event for each BE */
+	dpcm_dapm_stream_event(fe, stream, SND_SOC_DAPM_STREAM_START);
+	fe->dpcm[stream].state = SND_SOC_DPCM_STATE_PREPARE;
+
+out:
+	fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO;
+	mutex_unlock(&fe->card->mutex);
+
+	return ret;
+}
+
+static int soc_pcm_ioctl(struct snd_pcm_substream *substream,
+		     unsigned int cmd, void *arg)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_platform *platform = rtd->platform;
+
+	if (platform->driver->ops->ioctl)
+		return platform->driver->ops->ioctl(substream, cmd, arg);
+	return snd_pcm_lib_ioctl(substream, cmd, arg);
+}
+
+static int dpcm_run_update_shutdown(struct snd_soc_pcm_runtime *fe, int stream)
+{
+	struct snd_pcm_substream *substream =
+		snd_soc_dpcm_get_substream(fe, stream);
+	enum snd_soc_dpcm_trigger trigger = fe->dai_link->trigger[stream];
+	int err;
+
+	dev_dbg(fe->dev, "runtime %s close on FE %s\n",
+			stream ? "capture" : "playback", fe->dai_link->name);
+
+	if (trigger == SND_SOC_DPCM_TRIGGER_BESPOKE) {
+		/* call bespoke trigger - FE takes care of all BE triggers */
+		dev_dbg(fe->dev, "dpcm: bespoke trigger FE %s cmd stop\n",
+				fe->dai_link->name);
+
+		err = soc_pcm_bespoke_trigger(substream, SNDRV_PCM_TRIGGER_STOP);
+		if (err < 0)
+			dev_err(fe->dev,"dpcm: trigger FE failed %d\n", err);
+	} else {
+		dev_dbg(fe->dev, "dpcm: trigger FE %s cmd stop\n",
+			fe->dai_link->name);
+
+		err = dpcm_be_dai_trigger(fe, stream, SNDRV_PCM_TRIGGER_STOP);
+		if (err < 0)
+			dev_err(fe->dev,"dpcm: trigger FE failed %d\n", err);
+	}
+
+	err = dpcm_be_dai_hw_free(fe, stream);
+	if (err < 0)
+		dev_err(fe->dev,"dpcm: hw_free FE failed %d\n", err);
+
+	err = dpcm_be_dai_shutdown(fe, stream);
+	if (err < 0)
+		dev_err(fe->dev,"dpcm: shutdown FE failed %d\n", err);
+
+	/* run the stream event for each BE */
+	dpcm_dapm_stream_event(fe, stream, SND_SOC_DAPM_STREAM_NOP);
+
+	return 0;
+}
+
+static int dpcm_run_update_startup(struct snd_soc_pcm_runtime *fe, int stream)
+{
+	struct snd_pcm_substream *substream =
+		snd_soc_dpcm_get_substream(fe, stream);
+	struct snd_soc_dpcm *dpcm;
+	enum snd_soc_dpcm_trigger trigger = fe->dai_link->trigger[stream];
+	int ret;
+
+	dev_dbg(fe->dev, "runtime %s open on FE %s\n",
+			stream ? "capture" : "playback", fe->dai_link->name);
+
+	/* Only start the BE if the FE is ready */
+	if (fe->dpcm[stream].state == SND_SOC_DPCM_STATE_HW_FREE ||
+		fe->dpcm[stream].state == SND_SOC_DPCM_STATE_CLOSE)
+		return -EINVAL;
+
+	/* startup must always be called for new BEs */
+	ret = dpcm_be_dai_startup(fe, stream);
+	if (ret < 0) {
+		goto disconnect;
+		return ret;
+	}
+
+	/* keep going if FE state is > open */
+	if (fe->dpcm[stream].state == SND_SOC_DPCM_STATE_OPEN)
+		return 0;
+
+	ret = dpcm_be_dai_hw_params(fe, stream);
+	if (ret < 0) {
+		goto close;
+		return ret;
+	}
+
+	/* keep going if FE state is > hw_params */
+	if (fe->dpcm[stream].state == SND_SOC_DPCM_STATE_HW_PARAMS)
+		return 0;
+
+
+	ret = dpcm_be_dai_prepare(fe, stream);
+	if (ret < 0) {
+		goto hw_free;
+		return ret;
+	}
+
+	/* run the stream event for each BE */
+	dpcm_dapm_stream_event(fe, stream, SND_SOC_DAPM_STREAM_NOP);
+
+	/* keep going if FE state is > prepare */
+	if (fe->dpcm[stream].state == SND_SOC_DPCM_STATE_PREPARE ||
+		fe->dpcm[stream].state == SND_SOC_DPCM_STATE_STOP)
+		return 0;
+
+	if (trigger == SND_SOC_DPCM_TRIGGER_BESPOKE) {
+		/* call trigger on the frontend - FE takes care of all BE triggers */
+		dev_dbg(fe->dev, "dpcm: bespoke trigger FE %s cmd start\n",
+				fe->dai_link->name);
+
+		ret = soc_pcm_bespoke_trigger(substream, SNDRV_PCM_TRIGGER_START);
+		if (ret < 0) {
+			dev_err(fe->dev,"dpcm: bespoke trigger FE failed %d\n", ret);
+			goto hw_free;
+		}
+	} else {
+		dev_dbg(fe->dev, "dpcm: trigger FE %s cmd start\n",
+			fe->dai_link->name);
+
+		ret = dpcm_be_dai_trigger(fe, stream,
+					SNDRV_PCM_TRIGGER_START);
+		if (ret < 0) {
+			dev_err(fe->dev,"dpcm: trigger FE failed %d\n", ret);
+			goto hw_free;
+		}
+	}
+
+	return 0;
+
+hw_free:
+	dpcm_be_dai_hw_free(fe, stream);
+close:
+	dpcm_be_dai_shutdown(fe, stream);
+disconnect:
+	/* disconnect any non started BEs */
+	list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) {
+		struct snd_soc_pcm_runtime *be = dpcm->be;
+		if (be->dpcm[stream].state != SND_SOC_DPCM_STATE_START)
+				dpcm->state = SND_SOC_DPCM_LINK_STATE_FREE;
+	}
+
+	return ret;
+}
+
+static int dpcm_run_new_update(struct snd_soc_pcm_runtime *fe, int stream)
+{
+	int ret;
+
+	fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_BE;
+	ret = dpcm_run_update_startup(fe, stream);
+	if (ret < 0)
+		dev_err(fe->dev, "failed to startup some BEs\n");
+	fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO;
+
+	return ret;
+}
+
+static int dpcm_run_old_update(struct snd_soc_pcm_runtime *fe, int stream)
+{
+	int ret;
+
+	fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_BE;
+	ret = dpcm_run_update_shutdown(fe, stream);
+	if (ret < 0)
+		dev_err(fe->dev, "failed to shutdown some BEs\n");
+	fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO;
+
+	return ret;
+}
+
+/* Called by DAPM mixer/mux changes to update audio routing between PCMs and
+ * any DAI links.
+ */
+int soc_dpcm_runtime_update(struct snd_soc_dapm_widget *widget)
+{
+	struct snd_soc_card *card;
+	int i, old, new, paths;
+
+	if (widget->codec)
+		card = widget->codec->card;
+	else if (widget->platform)
+		card = widget->platform->card;
+	else
+		return -EINVAL;
+
+	mutex_lock_nested(&card->mutex, SND_SOC_CARD_CLASS_RUNTIME);
+	for (i = 0; i < card->num_rtd; i++) {
+		struct snd_soc_dapm_widget_list *list;
+		struct snd_soc_pcm_runtime *fe = &card->rtd[i];
+
+		/* make sure link is FE */
+		if (!fe->dai_link->dynamic)
+			continue;
+
+		/* only check active links */
+		if (!fe->cpu_dai->active)
+			continue;
+
+		/* DAPM sync will call this to update DSP paths */
+		dev_dbg(fe->dev, "DPCM runtime update for FE %s\n",
+			fe->dai_link->name);
+
+		/* skip if FE doesn't have playback capability */
+		if (!fe->cpu_dai->driver->playback.channels_min)
+			goto capture;
+
+		paths = dpcm_path_get(fe, SNDRV_PCM_STREAM_PLAYBACK, &list);
+		if (paths < 0) {
+			dev_warn(fe->dev, "%s no valid %s path\n",
+					fe->dai_link->name,  "playback");
+			mutex_unlock(&card->mutex);
+			return paths;
+		}
+
+		/* update any new playback paths */
+		new = dpcm_process_paths(fe, SNDRV_PCM_STREAM_PLAYBACK, &list, 1);
+		if (new) {
+			dpcm_run_new_update(fe, SNDRV_PCM_STREAM_PLAYBACK);
+			dpcm_clear_pending_state(fe, SNDRV_PCM_STREAM_PLAYBACK);
+			dpcm_be_disconnect(fe, SNDRV_PCM_STREAM_PLAYBACK);
+		}
+
+		/* update any old playback paths */
+		old = dpcm_process_paths(fe, SNDRV_PCM_STREAM_PLAYBACK, &list, 0);
+		if (old) {
+			dpcm_run_old_update(fe, SNDRV_PCM_STREAM_PLAYBACK);
+			dpcm_clear_pending_state(fe, SNDRV_PCM_STREAM_PLAYBACK);
+			dpcm_be_disconnect(fe, SNDRV_PCM_STREAM_PLAYBACK);
+		}
+
+capture:
+		/* skip if FE doesn't have capture capability */
+		if (!fe->cpu_dai->driver->capture.channels_min)
+			continue;
+
+		paths = dpcm_path_get(fe, SNDRV_PCM_STREAM_CAPTURE, &list);
+		if (paths < 0) {
+			dev_warn(fe->dev, "%s no valid %s path\n",
+					fe->dai_link->name,  "capture");
+			mutex_unlock(&card->mutex);
+			return paths;
+		}
+
+		/* update any new capture paths */
+		new = dpcm_process_paths(fe, SNDRV_PCM_STREAM_CAPTURE, &list, 1);
+		if (new) {
+			dpcm_run_new_update(fe, SNDRV_PCM_STREAM_CAPTURE);
+			dpcm_clear_pending_state(fe, SNDRV_PCM_STREAM_CAPTURE);
+			dpcm_be_disconnect(fe, SNDRV_PCM_STREAM_CAPTURE);
+		}
+
+		/* update any old capture paths */
+		old = dpcm_process_paths(fe, SNDRV_PCM_STREAM_CAPTURE, &list, 0);
+		if (old) {
+			dpcm_run_old_update(fe, SNDRV_PCM_STREAM_CAPTURE);
+			dpcm_clear_pending_state(fe, SNDRV_PCM_STREAM_CAPTURE);
+			dpcm_be_disconnect(fe, SNDRV_PCM_STREAM_CAPTURE);
+		}
+
+		dpcm_path_put(&list);
+	}
+
+	mutex_unlock(&card->mutex);
+	return 0;
+}
+int soc_dpcm_be_digital_mute(struct snd_soc_pcm_runtime *fe, int mute)
+{
+	struct snd_soc_dpcm *dpcm;
+	struct list_head *clients =
+		&fe->dpcm[SNDRV_PCM_STREAM_PLAYBACK].be_clients;
+
+	list_for_each_entry(dpcm, clients, list_be) {
+
+		struct snd_soc_pcm_runtime *be = dpcm->be;
+		struct snd_soc_dai *dai = be->codec_dai;
+		struct snd_soc_dai_driver *drv = dai->driver;
+
+		if (be->dai_link->ignore_suspend)
+			continue;
+
+		dev_dbg(be->dev, "BE digital mute %s\n", be->dai_link->name);
+
+		if (drv->ops->digital_mute && dai->playback_active)
+				drv->ops->digital_mute(dai, mute);
+	}
+
+	return 0;
+}
+
+static int dpcm_fe_dai_open(struct snd_pcm_substream *fe_substream)
+{
+	struct snd_soc_pcm_runtime *fe = fe_substream->private_data;
+	struct snd_soc_dpcm *dpcm;
+	struct snd_soc_dapm_widget_list *list;
+	int ret;
+	int stream = fe_substream->stream;
+
+	mutex_lock_nested(&fe->card->mutex, SND_SOC_CARD_CLASS_RUNTIME);
+	fe->dpcm[stream].runtime = fe_substream->runtime;
+
+	if (dpcm_path_get(fe, stream, &list) <= 0) {
+		dev_warn(fe->dev, "asoc: %s no valid %s route\n",
+			fe->dai_link->name, stream ? "capture" : "playback");
+			mutex_unlock(&fe->card->mutex);
+			return -EINVAL;
+	}
+
+	/* calculate valid and active FE <-> BE dpcms */
+	dpcm_process_paths(fe, stream, &list, 1);
+
+	ret = dpcm_fe_dai_startup(fe_substream);
+	if (ret < 0) {
+		/* clean up all links */
+		list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be)
+			dpcm->state = SND_SOC_DPCM_LINK_STATE_FREE;
+
+		dpcm_be_disconnect(fe, stream);
+		fe->dpcm[stream].runtime = NULL;
+	}
+
+	dpcm_clear_pending_state(fe, stream);
+	dpcm_path_put(&list);
+	mutex_unlock(&fe->card->mutex);
+	return ret;
+}
+
+static int dpcm_fe_dai_close(struct snd_pcm_substream *fe_substream)
+{
+	struct snd_soc_pcm_runtime *fe = fe_substream->private_data;
+	struct snd_soc_dpcm *dpcm;
+	int stream = fe_substream->stream, ret;
+
+	mutex_lock_nested(&fe->card->mutex, SND_SOC_CARD_CLASS_RUNTIME);
+	ret = dpcm_fe_dai_shutdown(fe_substream);
+
+	/* mark FE's links ready to prune */
+	list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be)
+		dpcm->state = SND_SOC_DPCM_LINK_STATE_FREE;
+
+	dpcm_be_disconnect(fe, stream);
+
+	fe->dpcm[stream].runtime = NULL;
+	mutex_unlock(&fe->card->mutex);
+	return ret;
+}
+
 /* create a new pcm */
 int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num)
 {
@@ -641,56 +2001,94 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num)
 	struct snd_soc_platform *platform = rtd->platform;
 	struct snd_soc_dai *codec_dai = rtd->codec_dai;
 	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
-	struct snd_pcm_ops *soc_pcm_ops = &rtd->ops;
 	struct snd_pcm *pcm;
 	char new_name[64];
 	int ret = 0, playback = 0, capture = 0;
 
-	soc_pcm_ops->open	= soc_pcm_open;
-	soc_pcm_ops->close	= soc_pcm_close;
-	soc_pcm_ops->hw_params	= soc_pcm_hw_params;
-	soc_pcm_ops->hw_free	= soc_pcm_hw_free;
-	soc_pcm_ops->prepare	= soc_pcm_prepare;
-	soc_pcm_ops->trigger	= soc_pcm_trigger;
-	soc_pcm_ops->pointer	= soc_pcm_pointer;
+	if (rtd->dai_link->dynamic || rtd->dai_link->no_pcm) {
+		if (cpu_dai->driver->playback.channels_min)
+			playback = 1;
+		if (cpu_dai->driver->capture.channels_min)
+			capture = 1;
+	} else {
+		if (codec_dai->driver->playback.channels_min)
+			playback = 1;
+		if (codec_dai->driver->capture.channels_min)
+			capture = 1;
+	}
 
-	/* check client and interface hw capabilities */
-	snprintf(new_name, sizeof(new_name), "%s %s-%d",
-			rtd->dai_link->stream_name, codec_dai->name, num);
+	/* create the PCM */
+	if (rtd->dai_link->no_pcm) {
+		snprintf(new_name, sizeof(new_name), "(%s)",
+			rtd->dai_link->stream_name);
 
-	if (codec_dai->driver->playback.channels_min)
-		playback = 1;
-	if (codec_dai->driver->capture.channels_min)
-		capture = 1;
+		ret = snd_pcm_new_internal(rtd->card->snd_card, new_name, num,
+				playback, capture, &pcm);
+	} else {
+		if (rtd->dai_link->dynamic)
+			snprintf(new_name, sizeof(new_name), "%s (*)",
+				rtd->dai_link->stream_name);
+		else
+			snprintf(new_name, sizeof(new_name), "%s %s-%d",
+				rtd->dai_link->stream_name, codec_dai->name, num);
 
-	dev_dbg(rtd->card->dev, "registered pcm #%d %s\n",num,new_name);
-	ret = snd_pcm_new(rtd->card->snd_card, new_name,
-			num, playback, capture, &pcm);
+		ret = snd_pcm_new(rtd->card->snd_card, new_name, num, playback,
+			capture, &pcm);
+	}
 	if (ret < 0) {
 		printk(KERN_ERR "asoc: can't create pcm for codec %s\n", codec->name);
 		return ret;
 	}
+	dev_dbg(rtd->card->dev, "registered pcm #%d %s\n",num, new_name);
 
 	/* DAPM dai link stream work */
 	INIT_DELAYED_WORK(&rtd->delayed_work, close_delayed_work);
 
 	rtd->pcm = pcm;
 	pcm->private_data = rtd;
+
+	if (rtd->dai_link->no_pcm) {
+		if (playback)
+			pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream->private_data = rtd;
+		if (capture)
+			pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream->private_data = rtd;
+		goto out;
+	}
+
+	/* ASoC PCM operations */
+	if (rtd->dai_link->dynamic) {
+		rtd->ops.open		= dpcm_fe_dai_open;
+		rtd->ops.hw_params	= dpcm_fe_dai_hw_params;
+		rtd->ops.prepare	= dpcm_fe_dai_prepare;
+		rtd->ops.trigger	= dpcm_fe_dai_trigger;
+		rtd->ops.hw_free	= dpcm_fe_dai_hw_free;
+		rtd->ops.close		= dpcm_fe_dai_close;
+		rtd->ops.pointer	= soc_pcm_pointer;
+		rtd->ops.ioctl		= soc_pcm_ioctl;
+	} else {
+		rtd->ops.open		= soc_pcm_open;
+		rtd->ops.hw_params	= soc_pcm_hw_params;
+		rtd->ops.prepare	= soc_pcm_prepare;
+		rtd->ops.trigger	= soc_pcm_trigger;
+		rtd->ops.hw_free	= soc_pcm_hw_free;
+		rtd->ops.close		= soc_pcm_close;
+		rtd->ops.pointer	= soc_pcm_pointer;
+		rtd->ops.ioctl		= soc_pcm_ioctl;
+	}
+
 	if (platform->driver->ops) {
-		soc_pcm_ops->mmap = platform->driver->ops->mmap;
-		soc_pcm_ops->pointer = platform->driver->ops->pointer;
-		soc_pcm_ops->ioctl = platform->driver->ops->ioctl;
-		soc_pcm_ops->copy = platform->driver->ops->copy;
-		soc_pcm_ops->silence = platform->driver->ops->silence;
-		soc_pcm_ops->ack = platform->driver->ops->ack;
-		soc_pcm_ops->page = platform->driver->ops->page;
+		rtd->ops.ack		= platform->driver->ops->ack;
+		rtd->ops.copy		= platform->driver->ops->copy;
+		rtd->ops.silence	= platform->driver->ops->silence;
+		rtd->ops.page		= platform->driver->ops->page;
+		rtd->ops.mmap		= platform->driver->ops->mmap;
 	}
 
 	if (playback)
-		snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, soc_pcm_ops);
+		snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &rtd->ops);
 
 	if (capture)
-		snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, soc_pcm_ops);
+		snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &rtd->ops);
 
 	if (platform->driver->pcm_new) {
 		ret = platform->driver->pcm_new(rtd);
@@ -701,7 +2099,257 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num)
 	}
 
 	pcm->private_free = platform->driver->pcm_free;
+out:
 	printk(KERN_INFO "asoc: %s <-> %s mapping ok\n", codec_dai->name,
 		cpu_dai->name);
 	return ret;
 }
+
+/* is the current PCM operation for this FE ? */
+int snd_soc_dpcm_fe_can_update(struct snd_soc_pcm_runtime *fe, int stream)
+{
+	if (fe->dpcm[stream].runtime_update == SND_SOC_DPCM_UPDATE_FE)
+		return 1;
+	return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_dpcm_fe_can_update);
+
+/* is the current PCM operation for this BE ? */
+int snd_soc_dpcm_be_can_update(struct snd_soc_pcm_runtime *fe,
+		struct snd_soc_pcm_runtime *be, int stream)
+{
+	if ((fe->dpcm[stream].runtime_update == SND_SOC_DPCM_UPDATE_FE) ||
+	   ((fe->dpcm[stream].runtime_update == SND_SOC_DPCM_UPDATE_BE) &&
+		  be->dpcm[stream].runtime_update))
+		return 1;
+	return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_dpcm_be_can_update);
+
+/* get the substream for this BE */
+struct snd_pcm_substream *
+	snd_soc_dpcm_get_substream(struct snd_soc_pcm_runtime *be, int stream)
+{
+	return be->pcm->streams[stream].substream;
+}
+EXPORT_SYMBOL_GPL(snd_soc_dpcm_get_substream);
+
+/* get the BE runtime state */
+enum snd_soc_dpcm_state
+	snd_soc_dpcm_be_get_state(struct snd_soc_pcm_runtime *be, int stream)
+{
+	return be->dpcm[stream].state;
+}
+EXPORT_SYMBOL_GPL(snd_soc_dpcm_be_get_state);
+
+/* set the BE runtime state */
+void snd_soc_dpcm_be_set_state(struct snd_soc_pcm_runtime *be,
+		int stream, enum snd_soc_dpcm_state state)
+{
+	be->dpcm[stream].state = state;
+}
+EXPORT_SYMBOL_GPL(snd_soc_dpcm_be_set_state);
+
+/*
+ * We can only hw_free, stop, pause or suspend a BE DAI if any of it's FE
+ * are not running, paused or suspended for the specified stream direction.
+ */
+int snd_soc_dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe,
+		struct snd_soc_pcm_runtime *be, int stream)
+{
+	struct snd_soc_dpcm *dpcm;
+	int state;
+
+	list_for_each_entry(dpcm, &be->dpcm[stream].fe_clients, list_fe) {
+
+		if (dpcm->fe == fe)
+			continue;
+
+		state = dpcm->fe->dpcm[stream].state;
+		if (state == SND_SOC_DPCM_STATE_START ||
+			state == SND_SOC_DPCM_STATE_PAUSED ||
+			state == SND_SOC_DPCM_STATE_SUSPEND)
+			return 0;
+	}
+
+	/* it's safe to free/stop this BE DAI */
+	return 1;
+}
+EXPORT_SYMBOL_GPL(snd_soc_dpcm_can_be_free_stop);
+
+/*
+ * We can only change hw params a BE DAI if any of it's FE are not prepared,
+ * running, paused or suspended for the specified stream direction.
+ */
+int snd_soc_dpcm_can_be_params(struct snd_soc_pcm_runtime *fe,
+		struct snd_soc_pcm_runtime *be, int stream)
+{
+	struct snd_soc_dpcm *dpcm;
+	int state;
+
+	list_for_each_entry(dpcm, &be->dpcm[stream].fe_clients, list_fe) {
+
+		if (dpcm->fe == fe)
+			continue;
+
+		state = dpcm->fe->dpcm[stream].state;
+		if (state == SND_SOC_DPCM_STATE_START ||
+			state == SND_SOC_DPCM_STATE_PAUSED ||
+			state == SND_SOC_DPCM_STATE_SUSPEND ||
+			state == SND_SOC_DPCM_STATE_PREPARE)
+			return 0;
+	}
+
+	/* it's safe to change hw_params */
+	return 1;
+}
+EXPORT_SYMBOL_GPL(snd_soc_dpcm_can_be_params);
+
+int snd_soc_platform_trigger(struct snd_pcm_substream *substream,
+		int cmd, struct snd_soc_platform *platform)
+{
+	if (platform->driver->ops->trigger)
+		return platform->driver->ops->trigger(substream, cmd);
+	return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_platform_trigger);
+
+#ifdef CONFIG_DEBUG_FS
+static char *dpcm_state_string(enum snd_soc_dpcm_state state)
+{
+	switch (state) {
+	case SND_SOC_DPCM_STATE_NEW:
+		return "new";
+	case SND_SOC_DPCM_STATE_OPEN:
+		return "open";
+	case SND_SOC_DPCM_STATE_HW_PARAMS:
+		return "hw_params";
+	case SND_SOC_DPCM_STATE_PREPARE:
+		return "prepare";
+	case SND_SOC_DPCM_STATE_START:
+		return "start";
+	case SND_SOC_DPCM_STATE_STOP:
+		return "stop";
+	case SND_SOC_DPCM_STATE_SUSPEND:
+		return "suspend";
+	case SND_SOC_DPCM_STATE_PAUSED:
+		return "paused";
+	case SND_SOC_DPCM_STATE_HW_FREE:
+		return "hw_free";
+	case SND_SOC_DPCM_STATE_CLOSE:
+		return "close";
+	}
+
+	return "unknown";
+}
+
+static ssize_t dpcm_show_state(struct snd_soc_pcm_runtime *fe,
+				int stream, char *buf, size_t size)
+{
+	struct snd_pcm_hw_params *params = &fe->dpcm[stream].hw_params;
+	struct snd_soc_dpcm *dpcm;
+	ssize_t offset = 0;
+
+	/* FE state */
+	offset += snprintf(buf + offset, size - offset,
+			"[%s - %s]\n", fe->dai_link->name,
+			stream ? "Capture" : "Playback");
+
+	offset += snprintf(buf + offset, size - offset, "State: %s\n",
+	                dpcm_state_string(fe->dpcm[stream].state));
+
+	if ((fe->dpcm[stream].state >= SND_SOC_DPCM_STATE_HW_PARAMS) &&
+	    (fe->dpcm[stream].state <= SND_SOC_DPCM_STATE_STOP))
+		offset += snprintf(buf + offset, size - offset,
+				"Hardware Params: "
+				"Format = %s, Channels = %d, Rate = %d\n",
+				snd_pcm_format_name(params_format(params)),
+				params_channels(params),
+				params_rate(params));
+
+	/* BEs state */
+	offset += snprintf(buf + offset, size - offset, "Backends:\n");
+
+	if (list_empty(&fe->dpcm[stream].be_clients)) {
+		offset += snprintf(buf + offset, size - offset,
+				" No active DSP links\n");
+		goto out;
+	}
+
+	list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) {
+		struct snd_soc_pcm_runtime *be = dpcm->be;
+		params = &dpcm->hw_params;
+
+		offset += snprintf(buf + offset, size - offset,
+				"- %s\n", be->dai_link->name);
+
+		offset += snprintf(buf + offset, size - offset,
+				"   State: %s\n",
+				dpcm_state_string(be->dpcm[stream].state));
+
+		if ((be->dpcm[stream].state >= SND_SOC_DPCM_STATE_HW_PARAMS) &&
+		    (be->dpcm[stream].state <= SND_SOC_DPCM_STATE_STOP))
+			offset += snprintf(buf + offset, size - offset,
+				"   Hardware Params: "
+				"Format = %s, Channels = %d, Rate = %d\n",
+				snd_pcm_format_name(params_format(params)),
+				params_channels(params),
+				params_rate(params));
+	}
+
+out:
+	return offset;
+}
+
+static ssize_t dpcm_state_read_file(struct file *file, char __user *user_buf,
+				size_t count, loff_t *ppos)
+{
+	struct snd_soc_pcm_runtime *fe = file->private_data;
+	ssize_t out_count = PAGE_SIZE, offset = 0, ret = 0;
+	char *buf;
+
+	buf = kmalloc(out_count, GFP_KERNEL);
+	if (!buf)
+		return -ENOMEM;
+
+	if (fe->cpu_dai->driver->playback.channels_min)
+		offset += dpcm_show_state(fe, SNDRV_PCM_STREAM_PLAYBACK,
+					buf + offset, out_count - offset);
+
+	if (fe->cpu_dai->driver->capture.channels_min)
+		offset += dpcm_show_state(fe, SNDRV_PCM_STREAM_CAPTURE,
+					buf + offset, out_count - offset);
+
+	ret = simple_read_from_buffer(user_buf, count, ppos, buf, offset);
+
+	kfree(buf);
+	return ret;
+}
+
+static const struct file_operations dpcm_state_fops = {
+	.open = simple_open,
+	.read = dpcm_state_read_file,
+	.llseek = default_llseek,
+};
+
+int soc_dpcm_debugfs_add(struct snd_soc_pcm_runtime *rtd)
+{
+	if (!rtd->dai_link)
+		return 0;
+
+	rtd->debugfs_dpcm_root = debugfs_create_dir(rtd->dai_link->name,
+			rtd->card->debugfs_card_root);
+	if (!rtd->debugfs_dpcm_root) {
+		dev_dbg(rtd->dev,
+			 "ASoC: Failed to create dpcm debugfs directory %s\n",
+			 rtd->dai_link->name);
+		return -EINVAL;
+	}
+
+	rtd->debugfs_dpcm_state = debugfs_create_file("state", 0444,
+						rtd->debugfs_dpcm_root,
+						rtd, &dpcm_state_fops);
+
+	return 0;
+}
+#endif
diff --git a/sound/soc/tegra/Kconfig b/sound/soc/tegra/Kconfig
index ce1b773c351f..c1c8e955f4d3 100644
--- a/sound/soc/tegra/Kconfig
+++ b/sound/soc/tegra/Kconfig
@@ -1,26 +1,63 @@
 config SND_SOC_TEGRA
 	tristate "SoC Audio for the Tegra System-on-Chip"
 	depends on ARCH_TEGRA && TEGRA_SYSTEM_DMA
+	select REGMAP_MMIO
 	help
 	  Say Y or M here if you want support for SoC audio on Tegra.
 
-config SND_SOC_TEGRA_I2S
+config SND_SOC_TEGRA20_DAS
 	tristate
-	depends on SND_SOC_TEGRA
+	depends on SND_SOC_TEGRA && ARCH_TEGRA_2x_SOC
+	help
+	  Say Y or M if you want to add support for the Tegra20 DAS module.
+	  You will also need to select the individual machine drivers to
+	  support below.
+
+config SND_SOC_TEGRA20_I2S
+	tristate
+	depends on SND_SOC_TEGRA && ARCH_TEGRA_2x_SOC
+	select SND_SOC_TEGRA20_DAS
 	help
 	  Say Y or M if you want to add support for codecs attached to the
-	  Tegra I2S interface. You will also need to select the individual
+	  Tegra20 I2S interface. You will also need to select the individual
 	  machine drivers to support below.
 
-config SND_SOC_TEGRA_SPDIF
+config SND_SOC_TEGRA20_SPDIF
 	tristate
-	depends on SND_SOC_TEGRA
+	depends on SND_SOC_TEGRA && ARCH_TEGRA_2x_SOC
 	default m
 	help
-	  Say Y or M if you want to add support for the SPDIF interface.
+	  Say Y or M if you want to add support for the Tegra20 SPDIF interface.
 	  You will also need to select the individual machine drivers to support
 	  below.
 
+config SND_SOC_TEGRA30_AHUB
+	tristate
+	depends on SND_SOC_TEGRA && ARCH_TEGRA_3x_SOC
+	help
+	  Say Y or M if you want to add support for the Tegra20 AHUB module.
+	  You will also need to select the individual machine drivers to
+	  support below.
+
+config SND_SOC_TEGRA30_I2S
+	tristate
+	depends on SND_SOC_TEGRA && ARCH_TEGRA_3x_SOC
+	select SND_SOC_TEGRA30_AHUB
+	help
+	  Say Y or M if you want to add support for codecs attached to the
+	  Tegra30 I2S interface. You will also need to select the individual
+	  machine drivers to support below.
+
+config SND_SOC_TEGRA_WM8753
+	tristate "SoC Audio support for Tegra boards using a WM8753 codec"
+	depends on SND_SOC_TEGRA && I2C
+	select SND_SOC_TEGRA20_I2S if ARCH_TEGRA_2x_SOC
+	select SND_SOC_TEGRA30_I2S if ARCH_TEGRA_3x_SOC
+	select SND_SOC_WM8753
+	help
+	  Say Y or M here if you want to add support for SoC audio on Tegra
+	  boards using the WM8753 codec, such as Whistler.
+
 config MACH_HAS_SND_SOC_TEGRA_WM8903
 	bool
 	help
@@ -32,7 +69,8 @@ config SND_SOC_TEGRA_WM8903
 	tristate "SoC Audio support for Tegra boards using a WM8903 codec"
 	depends on SND_SOC_TEGRA && I2C
 	depends on MACH_HAS_SND_SOC_TEGRA_WM8903
-	select SND_SOC_TEGRA_I2S
+	select SND_SOC_TEGRA20_I2S if ARCH_TEGRA_2x_SOC
+	select SND_SOC_TEGRA30_I2S if ARCH_TEGRA_3x_SOC
 	select SND_SOC_WM8903
 	help
 	  Say Y or M here if you want to add support for SoC audio on Tegra
@@ -42,17 +80,17 @@ config SND_SOC_TEGRA_WM8903
 config SND_SOC_TEGRA_TRIMSLICE
 	tristate "SoC Audio support for TrimSlice board"
 	depends on SND_SOC_TEGRA && MACH_TRIMSLICE && I2C
-	select SND_SOC_TEGRA_I2S
+	select SND_SOC_TEGRA20_I2S if ARCH_TEGRA_2x_SOC
 	select SND_SOC_TLV320AIC23
 	help
 	  Say Y or M here if you want to add support for SoC audio on the
 	  TrimSlice platform.
 
 config SND_SOC_TEGRA_ALC5632
-       tristate "SoC Audio support for Tegra boards using an ALC5632 codec"
-       depends on SND_SOC_TEGRA && I2C
-       select SND_SOC_TEGRA_I2S
-       select SND_SOC_ALC5632
-       help
-         Say Y or M here if you want to add support for SoC audio on the
-         Toshiba AC100 netbook.
+	tristate "SoC Audio support for Tegra boards using an ALC5632 codec"
+	depends on SND_SOC_TEGRA && I2C
+	select SND_SOC_TEGRA20_I2S if ARCH_TEGRA_2x_SOC
+	select SND_SOC_ALC5632
+	help
+	  Say Y or M here if you want to add support for SoC audio on the
+	  Toshiba AC100 netbook.
diff --git a/sound/soc/tegra/Makefile b/sound/soc/tegra/Makefile
index 8e584b8fcfba..391e78a34c06 100644
--- a/sound/soc/tegra/Makefile
+++ b/sound/soc/tegra/Makefile
@@ -1,21 +1,27 @@
 # Tegra platform Support
-snd-soc-tegra-das-objs := tegra_das.o
 snd-soc-tegra-pcm-objs := tegra_pcm.o
-snd-soc-tegra-i2s-objs := tegra_i2s.o
-snd-soc-tegra-spdif-objs := tegra_spdif.o
 snd-soc-tegra-utils-objs += tegra_asoc_utils.o
+snd-soc-tegra20-das-objs := tegra20_das.o
+snd-soc-tegra20-i2s-objs := tegra20_i2s.o
+snd-soc-tegra20-spdif-objs := tegra20_spdif.o
+snd-soc-tegra30-ahub-objs := tegra30_ahub.o
+snd-soc-tegra30-i2s-objs := tegra30_i2s.o
 
-obj-$(CONFIG_SND_SOC_TEGRA) += snd-soc-tegra-utils.o
-obj-$(CONFIG_SND_SOC_TEGRA) += snd-soc-tegra-das.o
 obj-$(CONFIG_SND_SOC_TEGRA) += snd-soc-tegra-pcm.o
-obj-$(CONFIG_SND_SOC_TEGRA_I2S) += snd-soc-tegra-i2s.o
-obj-$(CONFIG_SND_SOC_TEGRA_SPDIF) += snd-soc-tegra-spdif.o
+obj-$(CONFIG_SND_SOC_TEGRA) += snd-soc-tegra-utils.o
+obj-$(CONFIG_SND_SOC_TEGRA20_DAS) += snd-soc-tegra20-das.o
+obj-$(CONFIG_SND_SOC_TEGRA20_I2S) += snd-soc-tegra20-i2s.o
+obj-$(CONFIG_SND_SOC_TEGRA20_SPDIF) += snd-soc-tegra20-spdif.o
+obj-$(CONFIG_SND_SOC_TEGRA30_AHUB) += snd-soc-tegra30-ahub.o
+obj-$(CONFIG_SND_SOC_TEGRA30_I2S) += snd-soc-tegra30-i2s.o
 
 # Tegra machine Support
+snd-soc-tegra-wm8753-objs := tegra_wm8753.o
 snd-soc-tegra-wm8903-objs := tegra_wm8903.o
 snd-soc-tegra-trimslice-objs := trimslice.o
 snd-soc-tegra-alc5632-objs := tegra_alc5632.o
 
+obj-$(CONFIG_SND_SOC_TEGRA_WM8753) += snd-soc-tegra-wm8753.o
 obj-$(CONFIG_SND_SOC_TEGRA_WM8903) += snd-soc-tegra-wm8903.o
 obj-$(CONFIG_SND_SOC_TEGRA_TRIMSLICE) += snd-soc-tegra-trimslice.o
 obj-$(CONFIG_SND_SOC_TEGRA_ALC5632) += snd-soc-tegra-alc5632.o
diff --git a/sound/soc/tegra/tegra20_das.c b/sound/soc/tegra/tegra20_das.c
new file mode 100644
index 000000000000..bf99296bce95
--- /dev/null
+++ b/sound/soc/tegra/tegra20_das.c
@@ -0,0 +1,233 @@
+/*
+ * tegra20_das.c - Tegra20 DAS driver
+ *
+ * Author: Stephen Warren <swarren@nvidia.com>
+ * Copyright (C) 2010 - NVIDIA, Inc.
+ *
+ * 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
+ *
+ */
+
+#include <linux/device.h>
+#include <linux/io.h>
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <linux/regmap.h>
+#include <linux/slab.h>
+#include <sound/soc.h>
+#include "tegra20_das.h"
+
+#define DRV_NAME "tegra20-das"
+
+static struct tegra20_das *das;
+
+static inline void tegra20_das_write(u32 reg, u32 val)
+{
+	regmap_write(das->regmap, reg, val);
+}
+
+static inline u32 tegra20_das_read(u32 reg)
+{
+	u32 val;
+	regmap_read(das->regmap, reg, &val);
+	return val;
+}
+
+int tegra20_das_connect_dap_to_dac(int dap, int dac)
+{
+	u32 addr;
+	u32 reg;
+
+	if (!das)
+		return -ENODEV;
+
+	addr = TEGRA20_DAS_DAP_CTRL_SEL +
+		(dap * TEGRA20_DAS_DAP_CTRL_SEL_STRIDE);
+	reg = dac << TEGRA20_DAS_DAP_CTRL_SEL_DAP_CTRL_SEL_P;
+
+	tegra20_das_write(addr, reg);
+
+	return 0;
+}
+EXPORT_SYMBOL_GPL(tegra20_das_connect_dap_to_dac);
+
+int tegra20_das_connect_dap_to_dap(int dap, int otherdap, int master,
+				   int sdata1rx, int sdata2rx)
+{
+	u32 addr;
+	u32 reg;
+
+	if (!das)
+		return -ENODEV;
+
+	addr = TEGRA20_DAS_DAP_CTRL_SEL +
+		(dap * TEGRA20_DAS_DAP_CTRL_SEL_STRIDE);
+	reg = otherdap << TEGRA20_DAS_DAP_CTRL_SEL_DAP_CTRL_SEL_P |
+		!!sdata2rx << TEGRA20_DAS_DAP_CTRL_SEL_DAP_SDATA2_TX_RX_P |
+		!!sdata1rx << TEGRA20_DAS_DAP_CTRL_SEL_DAP_SDATA1_TX_RX_P |
+		!!master << TEGRA20_DAS_DAP_CTRL_SEL_DAP_MS_SEL_P;
+
+	tegra20_das_write(addr, reg);
+
+	return 0;
+}
+EXPORT_SYMBOL_GPL(tegra20_das_connect_dap_to_dap);
+
+int tegra20_das_connect_dac_to_dap(int dac, int dap)
+{
+	u32 addr;
+	u32 reg;
+
+	if (!das)
+		return -ENODEV;
+
+	addr = TEGRA20_DAS_DAC_INPUT_DATA_CLK_SEL +
+		(dac * TEGRA20_DAS_DAC_INPUT_DATA_CLK_SEL_STRIDE);
+	reg = dap << TEGRA20_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_CLK_SEL_P |
+		dap << TEGRA20_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_SDATA1_SEL_P |
+		dap << TEGRA20_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_SDATA2_SEL_P;
+
+	tegra20_das_write(addr, reg);
+
+	return 0;
+}
+EXPORT_SYMBOL_GPL(tegra20_das_connect_dac_to_dap);
+
+#define LAST_REG(name) \
+	(TEGRA20_DAS_##name + \
+	 (TEGRA20_DAS_##name##_STRIDE * (TEGRA20_DAS_##name##_COUNT - 1)))
+
+static bool tegra20_das_wr_rd_reg(struct device *dev, unsigned int reg)
+{
+	if ((reg >= TEGRA20_DAS_DAP_CTRL_SEL) &&
+	    (reg <= LAST_REG(DAP_CTRL_SEL)))
+		return true;
+	if ((reg >= TEGRA20_DAS_DAC_INPUT_DATA_CLK_SEL) &&
+	    (reg <= LAST_REG(DAC_INPUT_DATA_CLK_SEL)))
+		return true;
+
+	return false;
+}
+
+static const struct regmap_config tegra20_das_regmap_config = {
+	.reg_bits = 32,
+	.reg_stride = 4,
+	.val_bits = 32,
+	.max_register = LAST_REG(DAC_INPUT_DATA_CLK_SEL),
+	.writeable_reg = tegra20_das_wr_rd_reg,
+	.readable_reg = tegra20_das_wr_rd_reg,
+	.cache_type = REGCACHE_RBTREE,
+};
+
+static int __devinit tegra20_das_probe(struct platform_device *pdev)
+{
+	struct resource *res, *region;
+	void __iomem *regs;
+	int ret = 0;
+
+	if (das)
+		return -ENODEV;
+
+	das = devm_kzalloc(&pdev->dev, sizeof(struct tegra20_das), GFP_KERNEL);
+	if (!das) {
+		dev_err(&pdev->dev, "Can't allocate tegra20_das\n");
+		ret = -ENOMEM;
+		goto err;
+	}
+	das->dev = &pdev->dev;
+
+	res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+	if (!res) {
+		dev_err(&pdev->dev, "No memory resource\n");
+		ret = -ENODEV;
+		goto err;
+	}
+
+	region = devm_request_mem_region(&pdev->dev, res->start,
+					 resource_size(res), pdev->name);
+	if (!region) {
+		dev_err(&pdev->dev, "Memory region already claimed\n");
+		ret = -EBUSY;
+		goto err;
+	}
+
+	regs = devm_ioremap(&pdev->dev, res->start, resource_size(res));
+	if (!regs) {
+		dev_err(&pdev->dev, "ioremap failed\n");
+		ret = -ENOMEM;
+		goto err;
+	}
+
+	das->regmap = devm_regmap_init_mmio(&pdev->dev, regs,
+					    &tegra20_das_regmap_config);
+	if (IS_ERR(das->regmap)) {
+		dev_err(&pdev->dev, "regmap init failed\n");
+		ret = PTR_ERR(das->regmap);
+		goto err;
+	}
+
+	ret = tegra20_das_connect_dap_to_dac(TEGRA20_DAS_DAP_ID_1,
+					     TEGRA20_DAS_DAP_SEL_DAC1);
+	if (ret) {
+		dev_err(&pdev->dev, "Can't set up DAS DAP connection\n");
+		goto err;
+	}
+	ret = tegra20_das_connect_dac_to_dap(TEGRA20_DAS_DAC_ID_1,
+					     TEGRA20_DAS_DAC_SEL_DAP1);
+	if (ret) {
+		dev_err(&pdev->dev, "Can't set up DAS DAC connection\n");
+		goto err;
+	}
+
+	platform_set_drvdata(pdev, das);
+
+	return 0;
+
+err:
+	das = NULL;
+	return ret;
+}
+
+static int __devexit tegra20_das_remove(struct platform_device *pdev)
+{
+	if (!das)
+		return -ENODEV;
+
+	das = NULL;
+
+	return 0;
+}
+
+static const struct of_device_id tegra20_das_of_match[] __devinitconst = {
+	{ .compatible = "nvidia,tegra20-das", },
+	{},
+};
+
+static struct platform_driver tegra20_das_driver = {
+	.probe = tegra20_das_probe,
+	.remove = __devexit_p(tegra20_das_remove),
+	.driver = {
+		.name = DRV_NAME,
+		.owner = THIS_MODULE,
+		.of_match_table = tegra20_das_of_match,
+	},
+};
+module_platform_driver(tegra20_das_driver);
+
+MODULE_AUTHOR("Stephen Warren <swarren@nvidia.com>");
+MODULE_DESCRIPTION("Tegra20 DAS driver");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:" DRV_NAME);
+MODULE_DEVICE_TABLE(of, tegra20_das_of_match);
diff --git a/sound/soc/tegra/tegra20_das.h b/sound/soc/tegra/tegra20_das.h
new file mode 100644
index 000000000000..be217f3d3a75
--- /dev/null
+++ b/sound/soc/tegra/tegra20_das.h
@@ -0,0 +1,134 @@
+/*
+ * tegra20_das.h - Definitions for Tegra20 DAS driver
+ *
+ * Author: Stephen Warren <swarren@nvidia.com>
+ * Copyright (C) 2010,2012 - NVIDIA, Inc.
+ *
+ * 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 __TEGRA20_DAS_H__
+#define __TEGRA20_DAS_H__
+
+/* Register TEGRA20_DAS_DAP_CTRL_SEL */
+#define TEGRA20_DAS_DAP_CTRL_SEL			0x00
+#define TEGRA20_DAS_DAP_CTRL_SEL_COUNT			5
+#define TEGRA20_DAS_DAP_CTRL_SEL_STRIDE			4
+#define TEGRA20_DAS_DAP_CTRL_SEL_DAP_MS_SEL_P		31
+#define TEGRA20_DAS_DAP_CTRL_SEL_DAP_MS_SEL_S		1
+#define TEGRA20_DAS_DAP_CTRL_SEL_DAP_SDATA1_TX_RX_P	30
+#define TEGRA20_DAS_DAP_CTRL_SEL_DAP_SDATA1_TX_RX_S	1
+#define TEGRA20_DAS_DAP_CTRL_SEL_DAP_SDATA2_TX_RX_P	29
+#define TEGRA20_DAS_DAP_CTRL_SEL_DAP_SDATA2_TX_RX_S	1
+#define TEGRA20_DAS_DAP_CTRL_SEL_DAP_CTRL_SEL_P		0
+#define TEGRA20_DAS_DAP_CTRL_SEL_DAP_CTRL_SEL_S		5
+
+/* Values for field TEGRA20_DAS_DAP_CTRL_SEL_DAP_CTRL_SEL */
+#define TEGRA20_DAS_DAP_SEL_DAC1	0
+#define TEGRA20_DAS_DAP_SEL_DAC2	1
+#define TEGRA20_DAS_DAP_SEL_DAC3	2
+#define TEGRA20_DAS_DAP_SEL_DAP1	16
+#define TEGRA20_DAS_DAP_SEL_DAP2	17
+#define TEGRA20_DAS_DAP_SEL_DAP3	18
+#define TEGRA20_DAS_DAP_SEL_DAP4	19
+#define TEGRA20_DAS_DAP_SEL_DAP5	20
+
+/* Register TEGRA20_DAS_DAC_INPUT_DATA_CLK_SEL */
+#define TEGRA20_DAS_DAC_INPUT_DATA_CLK_SEL			0x40
+#define TEGRA20_DAS_DAC_INPUT_DATA_CLK_SEL_COUNT		3
+#define TEGRA20_DAS_DAC_INPUT_DATA_CLK_SEL_STRIDE		4
+#define TEGRA20_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_SDATA2_SEL_P	28
+#define TEGRA20_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_SDATA2_SEL_S	4
+#define TEGRA20_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_SDATA1_SEL_P	24
+#define TEGRA20_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_SDATA1_SEL_S	4
+#define TEGRA20_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_CLK_SEL_P	0
+#define TEGRA20_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_CLK_SEL_S	4
+
+/*
+ * Values for:
+ * TEGRA20_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_SDATA2_SEL
+ * TEGRA20_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_SDATA1_SEL
+ * TEGRA20_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_CLK_SEL
+ */
+#define TEGRA20_DAS_DAC_SEL_DAP1	0
+#define TEGRA20_DAS_DAC_SEL_DAP2	1
+#define TEGRA20_DAS_DAC_SEL_DAP3	2
+#define TEGRA20_DAS_DAC_SEL_DAP4	3
+#define TEGRA20_DAS_DAC_SEL_DAP5	4
+
+/*
+ * Names/IDs of the DACs/DAPs.
+ */
+
+#define TEGRA20_DAS_DAP_ID_1 0
+#define TEGRA20_DAS_DAP_ID_2 1
+#define TEGRA20_DAS_DAP_ID_3 2
+#define TEGRA20_DAS_DAP_ID_4 3
+#define TEGRA20_DAS_DAP_ID_5 4
+
+#define TEGRA20_DAS_DAC_ID_1 0
+#define TEGRA20_DAS_DAC_ID_2 1
+#define TEGRA20_DAS_DAC_ID_3 2
+
+struct tegra20_das {
+	struct device *dev;
+	struct regmap *regmap;
+};
+
+/*
+ * Terminology:
+ * DAS: Digital audio switch (HW module controlled by this driver)
+ * DAP: Digital audio port (port/pins on Tegra device)
+ * DAC: Digital audio controller (e.g. I2S or AC97 controller elsewhere)
+ *
+ * The Tegra DAS is a mux/cross-bar which can connect each DAP to a specific
+ * DAC, or another DAP. When DAPs are connected, one must be the master and
+ * one the slave. Each DAC allows selection of a specific DAP for input, to
+ * cater for the case where N DAPs are connected to 1 DAC for broadcast
+ * output.
+ *
+ * This driver is dumb; no attempt is made to ensure that a valid routing
+ * configuration is programmed.
+ */
+
+/*
+ * Connect a DAP to to a DAC
+ * dap_id: DAP to connect: TEGRA20_DAS_DAP_ID_*
+ * dac_sel: DAC to connect to: TEGRA20_DAS_DAP_SEL_DAC*
+ */
+extern int tegra20_das_connect_dap_to_dac(int dap_id, int dac_sel);
+
+/*
+ * Connect a DAP to to another DAP
+ * dap_id: DAP to connect: TEGRA20_DAS_DAP_ID_*
+ * other_dap_sel: DAP to connect to: TEGRA20_DAS_DAP_SEL_DAP*
+ * master: Is this DAP the master (1) or slave (0)
+ * sdata1rx: Is this DAP's SDATA1 pin RX (1) or TX (0)
+ * sdata2rx: Is this DAP's SDATA2 pin RX (1) or TX (0)
+ */
+extern int tegra20_das_connect_dap_to_dap(int dap_id, int other_dap_sel,
+					  int master, int sdata1rx,
+					  int sdata2rx);
+
+/*
+ * Connect a DAC's input to a DAP
+ * (DAC outputs are selected by the DAP)
+ * dac_id: DAC ID to connect: TEGRA20_DAS_DAC_ID_*
+ * dap_sel: DAP to receive input from: TEGRA20_DAS_DAC_SEL_DAP*
+ */
+extern int tegra20_das_connect_dac_to_dap(int dac_id, int dap_sel);
+
+#endif
diff --git a/sound/soc/tegra/tegra20_i2s.c b/sound/soc/tegra/tegra20_i2s.c
new file mode 100644
index 000000000000..0c7af63d444b
--- /dev/null
+++ b/sound/soc/tegra/tegra20_i2s.c
@@ -0,0 +1,494 @@
+/*
+ * tegra20_i2s.c - Tegra20 I2S driver
+ *
+ * Author: Stephen Warren <swarren@nvidia.com>
+ * Copyright (C) 2010,2012 - NVIDIA, Inc.
+ *
+ * Based on code copyright/by:
+ *
+ * Copyright (c) 2009-2010, NVIDIA Corporation.
+ * Scott Peterson <speterson@nvidia.com>
+ *
+ * Copyright (C) 2010 Google, Inc.
+ * Iliyan Malchev <malchev@google.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
+ *
+ */
+
+#include <linux/clk.h>
+#include <linux/device.h>
+#include <linux/io.h>
+#include <linux/module.h>
+#include <linux/of.h>
+#include <linux/platform_device.h>
+#include <linux/pm_runtime.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 "tegra20_i2s.h"
+
+#define DRV_NAME "tegra20-i2s"
+
+static inline void tegra20_i2s_write(struct tegra20_i2s *i2s, u32 reg, u32 val)
+{
+	regmap_write(i2s->regmap, reg, val);
+}
+
+static inline u32 tegra20_i2s_read(struct tegra20_i2s *i2s, u32 reg)
+{
+	u32 val;
+	regmap_read(i2s->regmap, reg, &val);
+	return val;
+}
+
+static int tegra20_i2s_runtime_suspend(struct device *dev)
+{
+	struct tegra20_i2s *i2s = dev_get_drvdata(dev);
+
+	clk_disable(i2s->clk_i2s);
+
+	return 0;
+}
+
+static int tegra20_i2s_runtime_resume(struct device *dev)
+{
+	struct tegra20_i2s *i2s = dev_get_drvdata(dev);
+	int ret;
+
+	ret = clk_enable(i2s->clk_i2s);
+	if (ret) {
+		dev_err(dev, "clk_enable failed: %d\n", ret);
+		return ret;
+	}
+
+	return 0;
+}
+
+static int tegra20_i2s_set_fmt(struct snd_soc_dai *dai,
+				unsigned int fmt)
+{
+	struct tegra20_i2s *i2s = snd_soc_dai_get_drvdata(dai);
+
+	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+	case SND_SOC_DAIFMT_NB_NF:
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	i2s->reg_ctrl &= ~TEGRA20_I2S_CTRL_MASTER_ENABLE;
+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBS_CFS:
+		i2s->reg_ctrl |= TEGRA20_I2S_CTRL_MASTER_ENABLE;
+		break;
+	case SND_SOC_DAIFMT_CBM_CFM:
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	i2s->reg_ctrl &= ~(TEGRA20_I2S_CTRL_BIT_FORMAT_MASK |
+			   TEGRA20_I2S_CTRL_LRCK_MASK);
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_DSP_A:
+		i2s->reg_ctrl |= TEGRA20_I2S_CTRL_BIT_FORMAT_DSP;
+		i2s->reg_ctrl |= TEGRA20_I2S_CTRL_LRCK_L_LOW;
+		break;
+	case SND_SOC_DAIFMT_DSP_B:
+		i2s->reg_ctrl |= TEGRA20_I2S_CTRL_BIT_FORMAT_DSP;
+		i2s->reg_ctrl |= TEGRA20_I2S_CTRL_LRCK_R_LOW;
+		break;
+	case SND_SOC_DAIFMT_I2S:
+		i2s->reg_ctrl |= TEGRA20_I2S_CTRL_BIT_FORMAT_I2S;
+		i2s->reg_ctrl |= TEGRA20_I2S_CTRL_LRCK_L_LOW;
+		break;
+	case SND_SOC_DAIFMT_RIGHT_J:
+		i2s->reg_ctrl |= TEGRA20_I2S_CTRL_BIT_FORMAT_RJM;
+		i2s->reg_ctrl |= TEGRA20_I2S_CTRL_LRCK_L_LOW;
+		break;
+	case SND_SOC_DAIFMT_LEFT_J:
+		i2s->reg_ctrl |= TEGRA20_I2S_CTRL_BIT_FORMAT_LJM;
+		i2s->reg_ctrl |= TEGRA20_I2S_CTRL_LRCK_L_LOW;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	return 0;
+}
+
+static int tegra20_i2s_hw_params(struct snd_pcm_substream *substream,
+				 struct snd_pcm_hw_params *params,
+				 struct snd_soc_dai *dai)
+{
+	struct device *dev = substream->pcm->card->dev;
+	struct tegra20_i2s *i2s = snd_soc_dai_get_drvdata(dai);
+	u32 reg;
+	int ret, sample_size, srate, i2sclock, bitcnt;
+
+	i2s->reg_ctrl &= ~TEGRA20_I2S_CTRL_BIT_SIZE_MASK;
+	switch (params_format(params)) {
+	case SNDRV_PCM_FORMAT_S16_LE:
+		i2s->reg_ctrl |= TEGRA20_I2S_CTRL_BIT_SIZE_16;
+		sample_size = 16;
+		break;
+	case SNDRV_PCM_FORMAT_S24_LE:
+		i2s->reg_ctrl |= TEGRA20_I2S_CTRL_BIT_SIZE_24;
+		sample_size = 24;
+		break;
+	case SNDRV_PCM_FORMAT_S32_LE:
+		i2s->reg_ctrl |= TEGRA20_I2S_CTRL_BIT_SIZE_32;
+		sample_size = 32;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	srate = params_rate(params);
+
+	/* Final "* 2" required by Tegra hardware */
+	i2sclock = srate * params_channels(params) * sample_size * 2;
+
+	ret = clk_set_rate(i2s->clk_i2s, i2sclock);
+	if (ret) {
+		dev_err(dev, "Can't set I2S clock rate: %d\n", ret);
+		return ret;
+	}
+
+	bitcnt = (i2sclock / (2 * srate)) - 1;
+	if (bitcnt < 0 || bitcnt > TEGRA20_I2S_TIMING_CHANNEL_BIT_COUNT_MASK_US)
+		return -EINVAL;
+	reg = bitcnt << TEGRA20_I2S_TIMING_CHANNEL_BIT_COUNT_SHIFT;
+
+	if (i2sclock % (2 * srate))
+		reg |= TEGRA20_I2S_TIMING_NON_SYM_ENABLE;
+
+	tegra20_i2s_write(i2s, TEGRA20_I2S_TIMING, reg);
+
+	tegra20_i2s_write(i2s, TEGRA20_I2S_FIFO_SCR,
+		TEGRA20_I2S_FIFO_SCR_FIFO2_ATN_LVL_FOUR_SLOTS |
+		TEGRA20_I2S_FIFO_SCR_FIFO1_ATN_LVL_FOUR_SLOTS);
+
+	return 0;
+}
+
+static void tegra20_i2s_start_playback(struct tegra20_i2s *i2s)
+{
+	i2s->reg_ctrl |= TEGRA20_I2S_CTRL_FIFO1_ENABLE;
+	tegra20_i2s_write(i2s, TEGRA20_I2S_CTRL, i2s->reg_ctrl);
+}
+
+static void tegra20_i2s_stop_playback(struct tegra20_i2s *i2s)
+{
+	i2s->reg_ctrl &= ~TEGRA20_I2S_CTRL_FIFO1_ENABLE;
+	tegra20_i2s_write(i2s, TEGRA20_I2S_CTRL, i2s->reg_ctrl);
+}
+
+static void tegra20_i2s_start_capture(struct tegra20_i2s *i2s)
+{
+	i2s->reg_ctrl |= TEGRA20_I2S_CTRL_FIFO2_ENABLE;
+	tegra20_i2s_write(i2s, TEGRA20_I2S_CTRL, i2s->reg_ctrl);
+}
+
+static void tegra20_i2s_stop_capture(struct tegra20_i2s *i2s)
+{
+	i2s->reg_ctrl &= ~TEGRA20_I2S_CTRL_FIFO2_ENABLE;
+	tegra20_i2s_write(i2s, TEGRA20_I2S_CTRL, i2s->reg_ctrl);
+}
+
+static int tegra20_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
+			       struct snd_soc_dai *dai)
+{
+	struct tegra20_i2s *i2s = snd_soc_dai_get_drvdata(dai);
+
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_START:
+	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+	case SNDRV_PCM_TRIGGER_RESUME:
+		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+			tegra20_i2s_start_playback(i2s);
+		else
+			tegra20_i2s_start_capture(i2s);
+		break;
+	case SNDRV_PCM_TRIGGER_STOP:
+	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+	case SNDRV_PCM_TRIGGER_SUSPEND:
+		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+			tegra20_i2s_stop_playback(i2s);
+		else
+			tegra20_i2s_stop_capture(i2s);
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	return 0;
+}
+
+static int tegra20_i2s_probe(struct snd_soc_dai *dai)
+{
+	struct tegra20_i2s *i2s = snd_soc_dai_get_drvdata(dai);
+
+	dai->capture_dma_data = &i2s->capture_dma_data;
+	dai->playback_dma_data = &i2s->playback_dma_data;
+
+	return 0;
+}
+
+static const struct snd_soc_dai_ops tegra20_i2s_dai_ops = {
+	.set_fmt	= tegra20_i2s_set_fmt,
+	.hw_params	= tegra20_i2s_hw_params,
+	.trigger	= tegra20_i2s_trigger,
+};
+
+static const struct snd_soc_dai_driver tegra20_i2s_dai_template = {
+	.probe = tegra20_i2s_probe,
+	.playback = {
+		.channels_min = 2,
+		.channels_max = 2,
+		.rates = SNDRV_PCM_RATE_8000_96000,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE,
+	},
+	.capture = {
+		.channels_min = 2,
+		.channels_max = 2,
+		.rates = SNDRV_PCM_RATE_8000_96000,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE,
+	},
+	.ops = &tegra20_i2s_dai_ops,
+	.symmetric_rates = 1,
+};
+
+static bool tegra20_i2s_wr_rd_reg(struct device *dev, unsigned int reg)
+{
+	switch (reg) {
+	case TEGRA20_I2S_CTRL:
+	case TEGRA20_I2S_STATUS:
+	case TEGRA20_I2S_TIMING:
+	case TEGRA20_I2S_FIFO_SCR:
+	case TEGRA20_I2S_PCM_CTRL:
+	case TEGRA20_I2S_NW_CTRL:
+	case TEGRA20_I2S_TDM_CTRL:
+	case TEGRA20_I2S_TDM_TX_RX_CTRL:
+	case TEGRA20_I2S_FIFO1:
+	case TEGRA20_I2S_FIFO2:
+		return true;
+	default:
+		return false;
+	};
+}
+
+static bool tegra20_i2s_volatile_reg(struct device *dev, unsigned int reg)
+{
+	switch (reg) {
+	case TEGRA20_I2S_STATUS:
+	case TEGRA20_I2S_FIFO_SCR:
+	case TEGRA20_I2S_FIFO1:
+	case TEGRA20_I2S_FIFO2:
+		return true;
+	default:
+		return false;
+	};
+}
+
+static bool tegra20_i2s_precious_reg(struct device *dev, unsigned int reg)
+{
+	switch (reg) {
+	case TEGRA20_I2S_FIFO1:
+	case TEGRA20_I2S_FIFO2:
+		return true;
+	default:
+		return false;
+	};
+}
+
+static const struct regmap_config tegra20_i2s_regmap_config = {
+	.reg_bits = 32,
+	.reg_stride = 4,
+	.val_bits = 32,
+	.max_register = TEGRA20_I2S_FIFO2,
+	.writeable_reg = tegra20_i2s_wr_rd_reg,
+	.readable_reg = tegra20_i2s_wr_rd_reg,
+	.volatile_reg = tegra20_i2s_volatile_reg,
+	.precious_reg = tegra20_i2s_precious_reg,
+	.cache_type = REGCACHE_RBTREE,
+};
+
+static __devinit int tegra20_i2s_platform_probe(struct platform_device *pdev)
+{
+	struct tegra20_i2s *i2s;
+	struct resource *mem, *memregion, *dmareq;
+	u32 of_dma[2];
+	u32 dma_ch;
+	void __iomem *regs;
+	int ret;
+
+	i2s = devm_kzalloc(&pdev->dev, sizeof(struct tegra20_i2s), GFP_KERNEL);
+	if (!i2s) {
+		dev_err(&pdev->dev, "Can't allocate tegra20_i2s\n");
+		ret = -ENOMEM;
+		goto err;
+	}
+	dev_set_drvdata(&pdev->dev, i2s);
+
+	i2s->dai = tegra20_i2s_dai_template;
+	i2s->dai.name = dev_name(&pdev->dev);
+
+	i2s->clk_i2s = clk_get(&pdev->dev, NULL);
+	if (IS_ERR(i2s->clk_i2s)) {
+		dev_err(&pdev->dev, "Can't retrieve i2s clock\n");
+		ret = PTR_ERR(i2s->clk_i2s);
+		goto err;
+	}
+
+	mem = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+	if (!mem) {
+		dev_err(&pdev->dev, "No memory resource\n");
+		ret = -ENODEV;
+		goto err_clk_put;
+	}
+
+	dmareq = platform_get_resource(pdev, IORESOURCE_DMA, 0);
+	if (!dmareq) {
+		if (of_property_read_u32_array(pdev->dev.of_node,
+					"nvidia,dma-request-selector",
+					of_dma, 2) < 0) {
+			dev_err(&pdev->dev, "No DMA resource\n");
+			ret = -ENODEV;
+			goto err_clk_put;
+		}
+		dma_ch = of_dma[1];
+	} else {
+		dma_ch = dmareq->start;
+	}
+
+	memregion = devm_request_mem_region(&pdev->dev, mem->start,
+					    resource_size(mem), DRV_NAME);
+	if (!memregion) {
+		dev_err(&pdev->dev, "Memory region already claimed\n");
+		ret = -EBUSY;
+		goto err_clk_put;
+	}
+
+	regs = devm_ioremap(&pdev->dev, mem->start, resource_size(mem));
+	if (!regs) {
+		dev_err(&pdev->dev, "ioremap failed\n");
+		ret = -ENOMEM;
+		goto err_clk_put;
+	}
+
+	i2s->regmap = devm_regmap_init_mmio(&pdev->dev, regs,
+					    &tegra20_i2s_regmap_config);
+	if (IS_ERR(i2s->regmap)) {
+		dev_err(&pdev->dev, "regmap init failed\n");
+		ret = PTR_ERR(i2s->regmap);
+		goto err_clk_put;
+	}
+
+	i2s->capture_dma_data.addr = mem->start + TEGRA20_I2S_FIFO2;
+	i2s->capture_dma_data.wrap = 4;
+	i2s->capture_dma_data.width = 32;
+	i2s->capture_dma_data.req_sel = dma_ch;
+
+	i2s->playback_dma_data.addr = mem->start + TEGRA20_I2S_FIFO1;
+	i2s->playback_dma_data.wrap = 4;
+	i2s->playback_dma_data.width = 32;
+	i2s->playback_dma_data.req_sel = dma_ch;
+
+	i2s->reg_ctrl = TEGRA20_I2S_CTRL_FIFO_FORMAT_PACKED;
+
+	pm_runtime_enable(&pdev->dev);
+	if (!pm_runtime_enabled(&pdev->dev)) {
+		ret = tegra20_i2s_runtime_resume(&pdev->dev);
+		if (ret)
+			goto err_pm_disable;
+	}
+
+	ret = snd_soc_register_dai(&pdev->dev, &i2s->dai);
+	if (ret) {
+		dev_err(&pdev->dev, "Could not register DAI: %d\n", ret);
+		ret = -ENOMEM;
+		goto err_suspend;
+	}
+
+	ret = tegra_pcm_platform_register(&pdev->dev);
+	if (ret) {
+		dev_err(&pdev->dev, "Could not register PCM: %d\n", ret);
+		goto err_unregister_dai;
+	}
+
+	return 0;
+
+err_unregister_dai:
+	snd_soc_unregister_dai(&pdev->dev);
+err_suspend:
+	if (!pm_runtime_status_suspended(&pdev->dev))
+		tegra20_i2s_runtime_suspend(&pdev->dev);
+err_pm_disable:
+	pm_runtime_disable(&pdev->dev);
+err_clk_put:
+	clk_put(i2s->clk_i2s);
+err:
+	return ret;
+}
+
+static int __devexit tegra20_i2s_platform_remove(struct platform_device *pdev)
+{
+	struct tegra20_i2s *i2s = dev_get_drvdata(&pdev->dev);
+
+	pm_runtime_disable(&pdev->dev);
+	if (!pm_runtime_status_suspended(&pdev->dev))
+		tegra20_i2s_runtime_suspend(&pdev->dev);
+
+	tegra_pcm_platform_unregister(&pdev->dev);
+	snd_soc_unregister_dai(&pdev->dev);
+
+	clk_put(i2s->clk_i2s);
+
+	return 0;
+}
+
+static const struct of_device_id tegra20_i2s_of_match[] __devinitconst = {
+	{ .compatible = "nvidia,tegra20-i2s", },
+	{},
+};
+
+static const struct dev_pm_ops tegra20_i2s_pm_ops __devinitconst = {
+	SET_RUNTIME_PM_OPS(tegra20_i2s_runtime_suspend,
+			   tegra20_i2s_runtime_resume, NULL)
+};
+
+static struct platform_driver tegra20_i2s_driver = {
+	.driver = {
+		.name = DRV_NAME,
+		.owner = THIS_MODULE,
+		.of_match_table = tegra20_i2s_of_match,
+		.pm = &tegra20_i2s_pm_ops,
+	},
+	.probe = tegra20_i2s_platform_probe,
+	.remove = __devexit_p(tegra20_i2s_platform_remove),
+};
+module_platform_driver(tegra20_i2s_driver);
+
+MODULE_AUTHOR("Stephen Warren <swarren@nvidia.com>");
+MODULE_DESCRIPTION("Tegra20 I2S ASoC driver");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:" DRV_NAME);
+MODULE_DEVICE_TABLE(of, tegra20_i2s_of_match);
diff --git a/sound/soc/tegra/tegra20_i2s.h b/sound/soc/tegra/tegra20_i2s.h
new file mode 100644
index 000000000000..a57efc6a597e
--- /dev/null
+++ b/sound/soc/tegra/tegra20_i2s.h
@@ -0,0 +1,164 @@
+/*
+ * tegra20_i2s.h - Definitions for Tegra20 I2S driver
+ *
+ * Author: Stephen Warren <swarren@nvidia.com>
+ * Copyright (C) 2010,2012 - NVIDIA, Inc.
+ *
+ * Based on code copyright/by:
+ *
+ * Copyright (c) 2009-2010, NVIDIA Corporation.
+ * Scott Peterson <speterson@nvidia.com>
+ *
+ * Copyright (C) 2010 Google, Inc.
+ * Iliyan Malchev <malchev@google.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 __TEGRA20_I2S_H__
+#define __TEGRA20_I2S_H__
+
+#include "tegra_pcm.h"
+
+/* Register offsets from TEGRA20_I2S1_BASE and TEGRA20_I2S2_BASE */
+
+#define TEGRA20_I2S_CTRL				0x00
+#define TEGRA20_I2S_STATUS				0x04
+#define TEGRA20_I2S_TIMING				0x08
+#define TEGRA20_I2S_FIFO_SCR				0x0c
+#define TEGRA20_I2S_PCM_CTRL				0x10
+#define TEGRA20_I2S_NW_CTRL				0x14
+#define TEGRA20_I2S_TDM_CTRL				0x20
+#define TEGRA20_I2S_TDM_TX_RX_CTRL			0x24
+#define TEGRA20_I2S_FIFO1				0x40
+#define TEGRA20_I2S_FIFO2				0x80
+
+/* Fields in TEGRA20_I2S_CTRL */
+
+#define TEGRA20_I2S_CTRL_FIFO2_TX_ENABLE		(1 << 30)
+#define TEGRA20_I2S_CTRL_FIFO1_ENABLE			(1 << 29)
+#define TEGRA20_I2S_CTRL_FIFO2_ENABLE			(1 << 28)
+#define TEGRA20_I2S_CTRL_FIFO1_RX_ENABLE		(1 << 27)
+#define TEGRA20_I2S_CTRL_FIFO_LPBK_ENABLE		(1 << 26)
+#define TEGRA20_I2S_CTRL_MASTER_ENABLE			(1 << 25)
+
+#define TEGRA20_I2S_LRCK_LEFT_LOW				0
+#define TEGRA20_I2S_LRCK_RIGHT_LOW			1
+
+#define TEGRA20_I2S_CTRL_LRCK_SHIFT			24
+#define TEGRA20_I2S_CTRL_LRCK_MASK			(1                          << TEGRA20_I2S_CTRL_LRCK_SHIFT)
+#define TEGRA20_I2S_CTRL_LRCK_L_LOW			(TEGRA20_I2S_LRCK_LEFT_LOW  << TEGRA20_I2S_CTRL_LRCK_SHIFT)
+#define TEGRA20_I2S_CTRL_LRCK_R_LOW			(TEGRA20_I2S_LRCK_RIGHT_LOW << TEGRA20_I2S_CTRL_LRCK_SHIFT)
+
+#define TEGRA20_I2S_BIT_FORMAT_I2S			0
+#define TEGRA20_I2S_BIT_FORMAT_RJM			1
+#define TEGRA20_I2S_BIT_FORMAT_LJM			2
+#define TEGRA20_I2S_BIT_FORMAT_DSP			3
+
+#define TEGRA20_I2S_CTRL_BIT_FORMAT_SHIFT		10
+#define TEGRA20_I2S_CTRL_BIT_FORMAT_MASK		(3                          << TEGRA20_I2S_CTRL_BIT_FORMAT_SHIFT)
+#define TEGRA20_I2S_CTRL_BIT_FORMAT_I2S			(TEGRA20_I2S_BIT_FORMAT_I2S << TEGRA20_I2S_CTRL_BIT_FORMAT_SHIFT)
+#define TEGRA20_I2S_CTRL_BIT_FORMAT_RJM			(TEGRA20_I2S_BIT_FORMAT_RJM << TEGRA20_I2S_CTRL_BIT_FORMAT_SHIFT)
+#define TEGRA20_I2S_CTRL_BIT_FORMAT_LJM			(TEGRA20_I2S_BIT_FORMAT_LJM << TEGRA20_I2S_CTRL_BIT_FORMAT_SHIFT)
+#define TEGRA20_I2S_CTRL_BIT_FORMAT_DSP			(TEGRA20_I2S_BIT_FORMAT_DSP << TEGRA20_I2S_CTRL_BIT_FORMAT_SHIFT)
+
+#define TEGRA20_I2S_BIT_SIZE_16				0
+#define TEGRA20_I2S_BIT_SIZE_20				1
+#define TEGRA20_I2S_BIT_SIZE_24				2
+#define TEGRA20_I2S_BIT_SIZE_32				3
+
+#define TEGRA20_I2S_CTRL_BIT_SIZE_SHIFT			8
+#define TEGRA20_I2S_CTRL_BIT_SIZE_MASK			(3                       << TEGRA20_I2S_CTRL_BIT_SIZE_SHIFT)
+#define TEGRA20_I2S_CTRL_BIT_SIZE_16			(TEGRA20_I2S_BIT_SIZE_16 << TEGRA20_I2S_CTRL_BIT_SIZE_SHIFT)
+#define TEGRA20_I2S_CTRL_BIT_SIZE_20			(TEGRA20_I2S_BIT_SIZE_20 << TEGRA20_I2S_CTRL_BIT_SIZE_SHIFT)
+#define TEGRA20_I2S_CTRL_BIT_SIZE_24			(TEGRA20_I2S_BIT_SIZE_24 << TEGRA20_I2S_CTRL_BIT_SIZE_SHIFT)
+#define TEGRA20_I2S_CTRL_BIT_SIZE_32			(TEGRA20_I2S_BIT_SIZE_32 << TEGRA20_I2S_CTRL_BIT_SIZE_SHIFT)
+
+#define TEGRA20_I2S_FIFO_16_LSB				0
+#define TEGRA20_I2S_FIFO_20_LSB				1
+#define TEGRA20_I2S_FIFO_24_LSB				2
+#define TEGRA20_I2S_FIFO_32				3
+#define TEGRA20_I2S_FIFO_PACKED				7
+
+#define TEGRA20_I2S_CTRL_FIFO_FORMAT_SHIFT		4
+#define TEGRA20_I2S_CTRL_FIFO_FORMAT_MASK		(7                       << TEGRA20_I2S_CTRL_FIFO_FORMAT_SHIFT)
+#define TEGRA20_I2S_CTRL_FIFO_FORMAT_16_LSB		(TEGRA20_I2S_FIFO_16_LSB << TEGRA20_I2S_CTRL_FIFO_FORMAT_SHIFT)
+#define TEGRA20_I2S_CTRL_FIFO_FORMAT_20_LSB		(TEGRA20_I2S_FIFO_20_LSB << TEGRA20_I2S_CTRL_FIFO_FORMAT_SHIFT)
+#define TEGRA20_I2S_CTRL_FIFO_FORMAT_24_LSB		(TEGRA20_I2S_FIFO_24_LSB << TEGRA20_I2S_CTRL_FIFO_FORMAT_SHIFT)
+#define TEGRA20_I2S_CTRL_FIFO_FORMAT_32			(TEGRA20_I2S_FIFO_32     << TEGRA20_I2S_CTRL_FIFO_FORMAT_SHIFT)
+#define TEGRA20_I2S_CTRL_FIFO_FORMAT_PACKED		(TEGRA20_I2S_FIFO_PACKED << TEGRA20_I2S_CTRL_FIFO_FORMAT_SHIFT)
+
+#define TEGRA20_I2S_CTRL_IE_FIFO1_ERR			(1 << 3)
+#define TEGRA20_I2S_CTRL_IE_FIFO2_ERR			(1 << 2)
+#define TEGRA20_I2S_CTRL_QE_FIFO1			(1 << 1)
+#define TEGRA20_I2S_CTRL_QE_FIFO2			(1 << 0)
+
+/* Fields in TEGRA20_I2S_STATUS */
+
+#define TEGRA20_I2S_STATUS_FIFO1_RDY			(1 << 31)
+#define TEGRA20_I2S_STATUS_FIFO2_RDY			(1 << 30)
+#define TEGRA20_I2S_STATUS_FIFO1_BSY			(1 << 29)
+#define TEGRA20_I2S_STATUS_FIFO2_BSY			(1 << 28)
+#define TEGRA20_I2S_STATUS_FIFO1_ERR			(1 << 3)
+#define TEGRA20_I2S_STATUS_FIFO2_ERR			(1 << 2)
+#define TEGRA20_I2S_STATUS_QS_FIFO1			(1 << 1)
+#define TEGRA20_I2S_STATUS_QS_FIFO2			(1 << 0)
+
+/* Fields in TEGRA20_I2S_TIMING */
+
+#define TEGRA20_I2S_TIMING_NON_SYM_ENABLE		(1 << 12)
+#define TEGRA20_I2S_TIMING_CHANNEL_BIT_COUNT_SHIFT	0
+#define TEGRA20_I2S_TIMING_CHANNEL_BIT_COUNT_MASK_US	0x7fff
+#define TEGRA20_I2S_TIMING_CHANNEL_BIT_COUNT_MASK	(TEGRA20_I2S_TIMING_CHANNEL_BIT_COUNT_MASK_US << TEGRA20_I2S_TIMING_CHANNEL_BIT_COUNT_SHIFT)
+
+/* Fields in TEGRA20_I2S_FIFO_SCR */
+
+#define TEGRA20_I2S_FIFO_SCR_FIFO2_FULL_EMPTY_COUNT_SHIFT	24
+#define TEGRA20_I2S_FIFO_SCR_FIFO1_FULL_EMPTY_COUNT_SHIFT	16
+#define TEGRA20_I2S_FIFO_SCR_FIFO_FULL_EMPTY_COUNT_MASK		0x3f
+
+#define TEGRA20_I2S_FIFO_SCR_FIFO2_CLR			(1 << 12)
+#define TEGRA20_I2S_FIFO_SCR_FIFO1_CLR			(1 << 8)
+
+#define TEGRA20_I2S_FIFO_ATN_LVL_ONE_SLOT		0
+#define TEGRA20_I2S_FIFO_ATN_LVL_FOUR_SLOTS		1
+#define TEGRA20_I2S_FIFO_ATN_LVL_EIGHT_SLOTS		2
+#define TEGRA20_I2S_FIFO_ATN_LVL_TWELVE_SLOTS		3
+
+#define TEGRA20_I2S_FIFO_SCR_FIFO2_ATN_LVL_SHIFT	4
+#define TEGRA20_I2S_FIFO_SCR_FIFO2_ATN_LVL_MASK		(3 << TEGRA20_I2S_FIFO_SCR_FIFO2_ATN_LVL_SHIFT)
+#define TEGRA20_I2S_FIFO_SCR_FIFO2_ATN_LVL_ONE_SLOT	(TEGRA20_I2S_FIFO_ATN_LVL_ONE_SLOT     << TEGRA20_I2S_FIFO_SCR_FIFO2_ATN_LVL_SHIFT)
+#define TEGRA20_I2S_FIFO_SCR_FIFO2_ATN_LVL_FOUR_SLOTS	(TEGRA20_I2S_FIFO_ATN_LVL_FOUR_SLOTS   << TEGRA20_I2S_FIFO_SCR_FIFO2_ATN_LVL_SHIFT)
+#define TEGRA20_I2S_FIFO_SCR_FIFO2_ATN_LVL_EIGHT_SLOTS	(TEGRA20_I2S_FIFO_ATN_LVL_EIGHT_SLOTS  << TEGRA20_I2S_FIFO_SCR_FIFO2_ATN_LVL_SHIFT)
+#define TEGRA20_I2S_FIFO_SCR_FIFO2_ATN_LVL_TWELVE_SLOTS	(TEGRA20_I2S_FIFO_ATN_LVL_TWELVE_SLOTS << TEGRA20_I2S_FIFO_SCR_FIFO2_ATN_LVL_SHIFT)
+
+#define TEGRA20_I2S_FIFO_SCR_FIFO1_ATN_LVL_SHIFT	0
+#define TEGRA20_I2S_FIFO_SCR_FIFO1_ATN_LVL_MASK		(3 << TEGRA20_I2S_FIFO_SCR_FIFO1_ATN_LVL_SHIFT)
+#define TEGRA20_I2S_FIFO_SCR_FIFO1_ATN_LVL_ONE_SLOT	(TEGRA20_I2S_FIFO_ATN_LVL_ONE_SLOT     << TEGRA20_I2S_FIFO_SCR_FIFO1_ATN_LVL_SHIFT)
+#define TEGRA20_I2S_FIFO_SCR_FIFO1_ATN_LVL_FOUR_SLOTS	(TEGRA20_I2S_FIFO_ATN_LVL_FOUR_SLOTS   << TEGRA20_I2S_FIFO_SCR_FIFO1_ATN_LVL_SHIFT)
+#define TEGRA20_I2S_FIFO_SCR_FIFO1_ATN_LVL_EIGHT_SLOTS	(TEGRA20_I2S_FIFO_ATN_LVL_EIGHT_SLOTS  << TEGRA20_I2S_FIFO_SCR_FIFO1_ATN_LVL_SHIFT)
+#define TEGRA20_I2S_FIFO_SCR_FIFO1_ATN_LVL_TWELVE_SLOTS	(TEGRA20_I2S_FIFO_ATN_LVL_TWELVE_SLOTS << TEGRA20_I2S_FIFO_SCR_FIFO1_ATN_LVL_SHIFT)
+
+struct tegra20_i2s {
+	struct snd_soc_dai_driver dai;
+	struct clk *clk_i2s;
+	struct tegra_pcm_dma_params capture_dma_data;
+	struct tegra_pcm_dma_params playback_dma_data;
+	struct regmap *regmap;
+	u32 reg_ctrl;
+};
+
+#endif
diff --git a/sound/soc/tegra/tegra20_spdif.c b/sound/soc/tegra/tegra20_spdif.c
new file mode 100644
index 000000000000..f9b57418bd08
--- /dev/null
+++ b/sound/soc/tegra/tegra20_spdif.c
@@ -0,0 +1,404 @@
+/*
+ * tegra20_spdif.c - Tegra20 SPDIF driver
+ *
+ * Author: Stephen Warren <swarren@nvidia.com>
+ * Copyright (C) 2011-2012 - NVIDIA, Inc.
+ *
+ * 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
+ *
+ */
+
+#include <linux/clk.h>
+#include <linux/device.h>
+#include <linux/io.h>
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <linux/pm_runtime.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 "tegra20_spdif.h"
+
+#define DRV_NAME "tegra20-spdif"
+
+static inline void tegra20_spdif_write(struct tegra20_spdif *spdif, u32 reg,
+					u32 val)
+{
+	regmap_write(spdif->regmap, reg, val);
+}
+
+static inline u32 tegra20_spdif_read(struct tegra20_spdif *spdif, u32 reg)
+{
+	u32 val;
+	regmap_read(spdif->regmap, reg, &val);
+	return val;
+}
+
+static int tegra20_spdif_runtime_suspend(struct device *dev)
+{
+	struct tegra20_spdif *spdif = dev_get_drvdata(dev);
+
+	clk_disable(spdif->clk_spdif_out);
+
+	return 0;
+}
+
+static int tegra20_spdif_runtime_resume(struct device *dev)
+{
+	struct tegra20_spdif *spdif = dev_get_drvdata(dev);
+	int ret;
+
+	ret = clk_enable(spdif->clk_spdif_out);
+	if (ret) {
+		dev_err(dev, "clk_enable failed: %d\n", ret);
+		return ret;
+	}
+
+	return 0;
+}
+
+static int tegra20_spdif_hw_params(struct snd_pcm_substream *substream,
+				struct snd_pcm_hw_params *params,
+				struct snd_soc_dai *dai)
+{
+	struct device *dev = substream->pcm->card->dev;
+	struct tegra20_spdif *spdif = snd_soc_dai_get_drvdata(dai);
+	int ret, spdifclock;
+
+	spdif->reg_ctrl &= ~TEGRA20_SPDIF_CTRL_PACK;
+	spdif->reg_ctrl &= ~TEGRA20_SPDIF_CTRL_BIT_MODE_MASK;
+	switch (params_format(params)) {
+	case SNDRV_PCM_FORMAT_S16_LE:
+		spdif->reg_ctrl |= TEGRA20_SPDIF_CTRL_PACK;
+		spdif->reg_ctrl |= TEGRA20_SPDIF_CTRL_BIT_MODE_16BIT;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	switch (params_rate(params)) {
+	case 32000:
+		spdifclock = 4096000;
+		break;
+	case 44100:
+		spdifclock = 5644800;
+		break;
+	case 48000:
+		spdifclock = 6144000;
+		break;
+	case 88200:
+		spdifclock = 11289600;
+		break;
+	case 96000:
+		spdifclock = 12288000;
+		break;
+	case 176400:
+		spdifclock = 22579200;
+		break;
+	case 192000:
+		spdifclock = 24576000;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	ret = clk_set_rate(spdif->clk_spdif_out, spdifclock);
+	if (ret) {
+		dev_err(dev, "Can't set SPDIF clock rate: %d\n", ret);
+		return ret;
+	}
+
+	return 0;
+}
+
+static void tegra20_spdif_start_playback(struct tegra20_spdif *spdif)
+{
+	spdif->reg_ctrl |= TEGRA20_SPDIF_CTRL_TX_EN;
+	tegra20_spdif_write(spdif, TEGRA20_SPDIF_CTRL, spdif->reg_ctrl);
+}
+
+static void tegra20_spdif_stop_playback(struct tegra20_spdif *spdif)
+{
+	spdif->reg_ctrl &= ~TEGRA20_SPDIF_CTRL_TX_EN;
+	tegra20_spdif_write(spdif, TEGRA20_SPDIF_CTRL, spdif->reg_ctrl);
+}
+
+static int tegra20_spdif_trigger(struct snd_pcm_substream *substream, int cmd,
+				struct snd_soc_dai *dai)
+{
+	struct tegra20_spdif *spdif = snd_soc_dai_get_drvdata(dai);
+
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_START:
+	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+	case SNDRV_PCM_TRIGGER_RESUME:
+		tegra20_spdif_start_playback(spdif);
+		break;
+	case SNDRV_PCM_TRIGGER_STOP:
+	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+	case SNDRV_PCM_TRIGGER_SUSPEND:
+		tegra20_spdif_stop_playback(spdif);
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	return 0;
+}
+
+static int tegra20_spdif_probe(struct snd_soc_dai *dai)
+{
+	struct tegra20_spdif *spdif = snd_soc_dai_get_drvdata(dai);
+
+	dai->capture_dma_data = NULL;
+	dai->playback_dma_data = &spdif->playback_dma_data;
+
+	return 0;
+}
+
+static const struct snd_soc_dai_ops tegra20_spdif_dai_ops = {
+	.hw_params	= tegra20_spdif_hw_params,
+	.trigger	= tegra20_spdif_trigger,
+};
+
+static struct snd_soc_dai_driver tegra20_spdif_dai = {
+	.name = DRV_NAME,
+	.probe = tegra20_spdif_probe,
+	.playback = {
+		.channels_min = 2,
+		.channels_max = 2,
+		.rates = SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |
+				SNDRV_PCM_RATE_48000,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE,
+	},
+	.ops = &tegra20_spdif_dai_ops,
+};
+
+static bool tegra20_spdif_wr_rd_reg(struct device *dev, unsigned int reg)
+{
+	switch (reg) {
+	case TEGRA20_SPDIF_CTRL:
+	case TEGRA20_SPDIF_STATUS:
+	case TEGRA20_SPDIF_STROBE_CTRL:
+	case TEGRA20_SPDIF_DATA_FIFO_CSR:
+	case TEGRA20_SPDIF_DATA_OUT:
+	case TEGRA20_SPDIF_DATA_IN:
+	case TEGRA20_SPDIF_CH_STA_RX_A:
+	case TEGRA20_SPDIF_CH_STA_RX_B:
+	case TEGRA20_SPDIF_CH_STA_RX_C:
+	case TEGRA20_SPDIF_CH_STA_RX_D:
+	case TEGRA20_SPDIF_CH_STA_RX_E:
+	case TEGRA20_SPDIF_CH_STA_RX_F:
+	case TEGRA20_SPDIF_CH_STA_TX_A:
+	case TEGRA20_SPDIF_CH_STA_TX_B:
+	case TEGRA20_SPDIF_CH_STA_TX_C:
+	case TEGRA20_SPDIF_CH_STA_TX_D:
+	case TEGRA20_SPDIF_CH_STA_TX_E:
+	case TEGRA20_SPDIF_CH_STA_TX_F:
+	case TEGRA20_SPDIF_USR_STA_RX_A:
+	case TEGRA20_SPDIF_USR_DAT_TX_A:
+		return true;
+	default:
+		return false;
+	};
+}
+
+static bool tegra20_spdif_volatile_reg(struct device *dev, unsigned int reg)
+{
+	switch (reg) {
+	case TEGRA20_SPDIF_STATUS:
+	case TEGRA20_SPDIF_DATA_FIFO_CSR:
+	case TEGRA20_SPDIF_DATA_OUT:
+	case TEGRA20_SPDIF_DATA_IN:
+	case TEGRA20_SPDIF_CH_STA_RX_A:
+	case TEGRA20_SPDIF_CH_STA_RX_B:
+	case TEGRA20_SPDIF_CH_STA_RX_C:
+	case TEGRA20_SPDIF_CH_STA_RX_D:
+	case TEGRA20_SPDIF_CH_STA_RX_E:
+	case TEGRA20_SPDIF_CH_STA_RX_F:
+	case TEGRA20_SPDIF_USR_STA_RX_A:
+	case TEGRA20_SPDIF_USR_DAT_TX_A:
+		return true;
+	default:
+		return false;
+	};
+}
+
+static bool tegra20_spdif_precious_reg(struct device *dev, unsigned int reg)
+{
+	switch (reg) {
+	case TEGRA20_SPDIF_DATA_OUT:
+	case TEGRA20_SPDIF_DATA_IN:
+	case TEGRA20_SPDIF_USR_STA_RX_A:
+	case TEGRA20_SPDIF_USR_DAT_TX_A:
+		return true;
+	default:
+		return false;
+	};
+}
+
+static const struct regmap_config tegra20_spdif_regmap_config = {
+	.reg_bits = 32,
+	.reg_stride = 4,
+	.val_bits = 32,
+	.max_register = TEGRA20_SPDIF_USR_DAT_TX_A,
+	.writeable_reg = tegra20_spdif_wr_rd_reg,
+	.readable_reg = tegra20_spdif_wr_rd_reg,
+	.volatile_reg = tegra20_spdif_volatile_reg,
+	.precious_reg = tegra20_spdif_precious_reg,
+	.cache_type = REGCACHE_RBTREE,
+};
+
+static __devinit int tegra20_spdif_platform_probe(struct platform_device *pdev)
+{
+	struct tegra20_spdif *spdif;
+	struct resource *mem, *memregion, *dmareq;
+	void __iomem *regs;
+	int ret;
+
+	spdif = devm_kzalloc(&pdev->dev, sizeof(struct tegra20_spdif),
+			     GFP_KERNEL);
+	if (!spdif) {
+		dev_err(&pdev->dev, "Can't allocate tegra20_spdif\n");
+		ret = -ENOMEM;
+		goto err;
+	}
+	dev_set_drvdata(&pdev->dev, spdif);
+
+	spdif->clk_spdif_out = clk_get(&pdev->dev, "spdif_out");
+	if (IS_ERR(spdif->clk_spdif_out)) {
+		pr_err("Can't retrieve spdif clock\n");
+		ret = PTR_ERR(spdif->clk_spdif_out);
+		goto err;
+	}
+
+	mem = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+	if (!mem) {
+		dev_err(&pdev->dev, "No memory resource\n");
+		ret = -ENODEV;
+		goto err_clk_put;
+	}
+
+	dmareq = platform_get_resource(pdev, IORESOURCE_DMA, 0);
+	if (!dmareq) {
+		dev_err(&pdev->dev, "No DMA resource\n");
+		ret = -ENODEV;
+		goto err_clk_put;
+	}
+
+	memregion = devm_request_mem_region(&pdev->dev, mem->start,
+					    resource_size(mem), DRV_NAME);
+	if (!memregion) {
+		dev_err(&pdev->dev, "Memory region already claimed\n");
+		ret = -EBUSY;
+		goto err_clk_put;
+	}
+
+	regs = devm_ioremap(&pdev->dev, mem->start, resource_size(mem));
+	if (!regs) {
+		dev_err(&pdev->dev, "ioremap failed\n");
+		ret = -ENOMEM;
+		goto err_clk_put;
+	}
+
+	spdif->regmap = devm_regmap_init_mmio(&pdev->dev, regs,
+					    &tegra20_spdif_regmap_config);
+	if (IS_ERR(spdif->regmap)) {
+		dev_err(&pdev->dev, "regmap init failed\n");
+		ret = PTR_ERR(spdif->regmap);
+		goto err_clk_put;
+	}
+
+	spdif->playback_dma_data.addr = mem->start + TEGRA20_SPDIF_DATA_OUT;
+	spdif->playback_dma_data.wrap = 4;
+	spdif->playback_dma_data.width = 32;
+	spdif->playback_dma_data.req_sel = dmareq->start;
+
+	pm_runtime_enable(&pdev->dev);
+	if (!pm_runtime_enabled(&pdev->dev)) {
+		ret = tegra20_spdif_runtime_resume(&pdev->dev);
+		if (ret)
+			goto err_pm_disable;
+	}
+
+	ret = snd_soc_register_dai(&pdev->dev, &tegra20_spdif_dai);
+	if (ret) {
+		dev_err(&pdev->dev, "Could not register DAI: %d\n", ret);
+		ret = -ENOMEM;
+		goto err_suspend;
+	}
+
+	ret = tegra_pcm_platform_register(&pdev->dev);
+	if (ret) {
+		dev_err(&pdev->dev, "Could not register PCM: %d\n", ret);
+		goto err_unregister_dai;
+	}
+
+	return 0;
+
+err_unregister_dai:
+	snd_soc_unregister_dai(&pdev->dev);
+err_suspend:
+	if (!pm_runtime_status_suspended(&pdev->dev))
+		tegra20_spdif_runtime_suspend(&pdev->dev);
+err_pm_disable:
+	pm_runtime_disable(&pdev->dev);
+err_clk_put:
+	clk_put(spdif->clk_spdif_out);
+err:
+	return ret;
+}
+
+static int __devexit tegra20_spdif_platform_remove(struct platform_device *pdev)
+{
+	struct tegra20_spdif *spdif = dev_get_drvdata(&pdev->dev);
+
+	pm_runtime_disable(&pdev->dev);
+	if (!pm_runtime_status_suspended(&pdev->dev))
+		tegra20_spdif_runtime_suspend(&pdev->dev);
+
+	tegra_pcm_platform_unregister(&pdev->dev);
+	snd_soc_unregister_dai(&pdev->dev);
+
+	clk_put(spdif->clk_spdif_out);
+
+	return 0;
+}
+
+static const struct dev_pm_ops tegra20_spdif_pm_ops __devinitconst = {
+	SET_RUNTIME_PM_OPS(tegra20_spdif_runtime_suspend,
+			   tegra20_spdif_runtime_resume, NULL)
+};
+
+static struct platform_driver tegra20_spdif_driver = {
+	.driver = {
+		.name = DRV_NAME,
+		.owner = THIS_MODULE,
+		.pm = &tegra20_spdif_pm_ops,
+	},
+	.probe = tegra20_spdif_platform_probe,
+	.remove = __devexit_p(tegra20_spdif_platform_remove),
+};
+
+module_platform_driver(tegra20_spdif_driver);
+
+MODULE_AUTHOR("Stephen Warren <swarren@nvidia.com>");
+MODULE_DESCRIPTION("Tegra20 SPDIF ASoC driver");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:" DRV_NAME);
diff --git a/sound/soc/tegra/tegra20_spdif.h b/sound/soc/tegra/tegra20_spdif.h
new file mode 100644
index 000000000000..ed756527efea
--- /dev/null
+++ b/sound/soc/tegra/tegra20_spdif.h
@@ -0,0 +1,471 @@
+/*
+ * tegra20_spdif.h - Definitions for Tegra20 SPDIF driver
+ *
+ * Author: Stephen Warren <swarren@nvidia.com>
+ * Copyright (C) 2011 - NVIDIA, Inc.
+ *
+ * Based on code copyright/by:
+ * Copyright (c) 2008-2009, NVIDIA Corporation
+ *
+ * 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 __TEGRA20_SPDIF_H__
+#define __TEGRA20_SPDIF_H__
+
+#include "tegra_pcm.h"
+
+/* Offsets from TEGRA20_SPDIF_BASE */
+
+#define TEGRA20_SPDIF_CTRL					0x0
+#define TEGRA20_SPDIF_STATUS					0x4
+#define TEGRA20_SPDIF_STROBE_CTRL				0x8
+#define TEGRA20_SPDIF_DATA_FIFO_CSR				0x0C
+#define TEGRA20_SPDIF_DATA_OUT					0x40
+#define TEGRA20_SPDIF_DATA_IN					0x80
+#define TEGRA20_SPDIF_CH_STA_RX_A				0x100
+#define TEGRA20_SPDIF_CH_STA_RX_B				0x104
+#define TEGRA20_SPDIF_CH_STA_RX_C				0x108
+#define TEGRA20_SPDIF_CH_STA_RX_D				0x10C
+#define TEGRA20_SPDIF_CH_STA_RX_E				0x110
+#define TEGRA20_SPDIF_CH_STA_RX_F				0x114
+#define TEGRA20_SPDIF_CH_STA_TX_A				0x140
+#define TEGRA20_SPDIF_CH_STA_TX_B				0x144
+#define TEGRA20_SPDIF_CH_STA_TX_C				0x148
+#define TEGRA20_SPDIF_CH_STA_TX_D				0x14C
+#define TEGRA20_SPDIF_CH_STA_TX_E				0x150
+#define TEGRA20_SPDIF_CH_STA_TX_F				0x154
+#define TEGRA20_SPDIF_USR_STA_RX_A				0x180
+#define TEGRA20_SPDIF_USR_DAT_TX_A				0x1C0
+
+/* Fields in TEGRA20_SPDIF_CTRL */
+
+/* Start capturing from 0=right, 1=left channel */
+#define TEGRA20_SPDIF_CTRL_CAP_LC				(1 << 30)
+
+/* SPDIF receiver(RX) enable */
+#define TEGRA20_SPDIF_CTRL_RX_EN				(1 << 29)
+
+/* SPDIF Transmitter(TX) enable */
+#define TEGRA20_SPDIF_CTRL_TX_EN				(1 << 28)
+
+/* Transmit Channel status */
+#define TEGRA20_SPDIF_CTRL_TC_EN				(1 << 27)
+
+/* Transmit user Data */
+#define TEGRA20_SPDIF_CTRL_TU_EN				(1 << 26)
+
+/* Interrupt on transmit error */
+#define TEGRA20_SPDIF_CTRL_IE_TXE				(1 << 25)
+
+/* Interrupt on receive error */
+#define TEGRA20_SPDIF_CTRL_IE_RXE				(1 << 24)
+
+/* Interrupt on invalid preamble */
+#define TEGRA20_SPDIF_CTRL_IE_P					(1 << 23)
+
+/* Interrupt on "B" preamble */
+#define TEGRA20_SPDIF_CTRL_IE_B					(1 << 22)
+
+/* Interrupt when block of channel status received */
+#define TEGRA20_SPDIF_CTRL_IE_C					(1 << 21)
+
+/* Interrupt when a valid information unit (IU) is received */
+#define TEGRA20_SPDIF_CTRL_IE_U					(1 << 20)
+
+/* Interrupt when RX user FIFO attention level is reached */
+#define TEGRA20_SPDIF_CTRL_QE_RU				(1 << 19)
+
+/* Interrupt when TX user FIFO attention level is reached */
+#define TEGRA20_SPDIF_CTRL_QE_TU				(1 << 18)
+
+/* Interrupt when RX data FIFO attention level is reached */
+#define TEGRA20_SPDIF_CTRL_QE_RX				(1 << 17)
+
+/* Interrupt when TX data FIFO attention level is reached */
+#define TEGRA20_SPDIF_CTRL_QE_TX				(1 << 16)
+
+/* Loopback test mode enable */
+#define TEGRA20_SPDIF_CTRL_LBK_EN				(1 << 15)
+
+/*
+ * Pack data mode:
+ * 0 = Single data (16 bit needs to be  padded to match the
+ *     interface data bit size).
+ * 1 = Packeted left/right channel data into a single word.
+ */
+#define TEGRA20_SPDIF_CTRL_PACK					(1 << 14)
+
+/*
+ * 00 = 16bit data
+ * 01 = 20bit data
+ * 10 = 24bit data
+ * 11 = raw data
+ */
+#define TEGRA20_SPDIF_BIT_MODE_16BIT				0
+#define TEGRA20_SPDIF_BIT_MODE_20BIT				1
+#define TEGRA20_SPDIF_BIT_MODE_24BIT				2
+#define TEGRA20_SPDIF_BIT_MODE_RAW				3
+
+#define TEGRA20_SPDIF_CTRL_BIT_MODE_SHIFT			12
+#define TEGRA20_SPDIF_CTRL_BIT_MODE_MASK			(3                            << TEGRA20_SPDIF_CTRL_BIT_MODE_SHIFT)
+#define TEGRA20_SPDIF_CTRL_BIT_MODE_16BIT			(TEGRA20_SPDIF_BIT_MODE_16BIT << TEGRA20_SPDIF_CTRL_BIT_MODE_SHIFT)
+#define TEGRA20_SPDIF_CTRL_BIT_MODE_20BIT			(TEGRA20_SPDIF_BIT_MODE_20BIT << TEGRA20_SPDIF_CTRL_BIT_MODE_SHIFT)
+#define TEGRA20_SPDIF_CTRL_BIT_MODE_24BIT			(TEGRA20_SPDIF_BIT_MODE_24BIT << TEGRA20_SPDIF_CTRL_BIT_MODE_SHIFT)
+#define TEGRA20_SPDIF_CTRL_BIT_MODE_RAW				(TEGRA20_SPDIF_BIT_MODE_RAW   << TEGRA20_SPDIF_CTRL_BIT_MODE_SHIFT)
+
+/* Fields in TEGRA20_SPDIF_STATUS */
+
+/*
+ * Note: IS_P, IS_B, IS_C, and IS_U are sticky bits. Software must
+ * write a 1 to the corresponding bit location to clear the status.
+ */
+
+/*
+ * Receiver(RX) shifter is busy receiving data.
+ * This bit is asserted when the receiver first locked onto the
+ * preamble of the data stream after RX_EN is asserted. This bit is
+ * deasserted when either,
+ * (a) the end of a frame is reached after RX_EN is deeasserted, or
+ * (b) the SPDIF data stream becomes inactive.
+ */
+#define TEGRA20_SPDIF_STATUS_RX_BSY				(1 << 29)
+
+/*
+ * Transmitter(TX) shifter is busy transmitting data.
+ * This bit is asserted when TX_EN is asserted.
+ * This bit is deasserted when the end of a frame is reached after
+ * TX_EN is deasserted.
+ */
+#define TEGRA20_SPDIF_STATUS_TX_BSY				(1 << 28)
+
+/*
+ * TX is busy shifting out channel status.
+ * This bit is asserted when both TX_EN and TC_EN are asserted and
+ * data from CH_STA_TX_A register is loaded into the internal shifter.
+ * This bit is deasserted when either,
+ * (a) the end of a frame is reached after TX_EN is deasserted, or
+ * (b) CH_STA_TX_F register is loaded into the internal shifter.
+ */
+#define TEGRA20_SPDIF_STATUS_TC_BSY				(1 << 27)
+
+/*
+ * TX User data FIFO busy.
+ * This bit is asserted when TX_EN and TXU_EN are asserted and
+ * there's data in the TX user FIFO.  This bit is deassert when either,
+ * (a) the end of a frame is reached after TX_EN is deasserted, or
+ * (b) there's no data left in the TX user FIFO.
+ */
+#define TEGRA20_SPDIF_STATUS_TU_BSY				(1 << 26)
+
+/* TX FIFO Underrun error status */
+#define TEGRA20_SPDIF_STATUS_TX_ERR				(1 << 25)
+
+/* RX FIFO Overrun error status */
+#define TEGRA20_SPDIF_STATUS_RX_ERR				(1 << 24)
+
+/* Preamble status: 0=Preamble OK, 1=bad/missing preamble */
+#define TEGRA20_SPDIF_STATUS_IS_P				(1 << 23)
+
+/* B-preamble detection status: 0=not detected, 1=B-preamble detected */
+#define TEGRA20_SPDIF_STATUS_IS_B				(1 << 22)
+
+/*
+ * RX channel block data receive status:
+ * 0=entire block not recieved yet.
+ * 1=received entire block of channel status,
+ */
+#define TEGRA20_SPDIF_STATUS_IS_C				(1 << 21)
+
+/* RX User Data Valid flag:  1=valid IU detected, 0 = no IU detected. */
+#define TEGRA20_SPDIF_STATUS_IS_U				(1 << 20)
+
+/*
+ * RX User FIFO Status:
+ * 1=attention level reached, 0=attention level not reached.
+ */
+#define TEGRA20_SPDIF_STATUS_QS_RU				(1 << 19)
+
+/*
+ * TX User FIFO Status:
+ * 1=attention level reached, 0=attention level not reached.
+ */
+#define TEGRA20_SPDIF_STATUS_QS_TU				(1 << 18)
+
+/*
+ * RX Data FIFO Status:
+ * 1=attention level reached, 0=attention level not reached.
+ */
+#define TEGRA20_SPDIF_STATUS_QS_RX				(1 << 17)
+
+/*
+ * TX Data FIFO Status:
+ * 1=attention level reached, 0=attention level not reached.
+ */
+#define TEGRA20_SPDIF_STATUS_QS_TX				(1 << 16)
+
+/* Fields in TEGRA20_SPDIF_STROBE_CTRL */
+
+/*
+ * Indicates the approximate number of detected SPDIFIN clocks within a
+ * bi-phase period.
+ */
+#define TEGRA20_SPDIF_STROBE_CTRL_PERIOD_SHIFT			16
+#define TEGRA20_SPDIF_STROBE_CTRL_PERIOD_MASK			(0xff << TEGRA20_SPDIF_STROBE_CTRL_PERIOD_SHIFT)
+
+/* Data strobe mode: 0=Auto-locked 1=Manual locked */
+#define TEGRA20_SPDIF_STROBE_CTRL_STROBE			(1 << 15)
+
+/*
+ * Manual data strobe time within the bi-phase clock period (in terms of
+ * the number of over-sampling clocks).
+ */
+#define TEGRA20_SPDIF_STROBE_CTRL_DATA_STROBES_SHIFT		8
+#define TEGRA20_SPDIF_STROBE_CTRL_DATA_STROBES_MASK		(0x1f << TEGRA20_SPDIF_STROBE_CTRL_DATA_STROBES_SHIFT)
+
+/*
+ * Manual SPDIFIN bi-phase clock period (in terms of the number of
+ * over-sampling clocks).
+ */
+#define TEGRA20_SPDIF_STROBE_CTRL_CLOCK_PERIOD_SHIFT		0
+#define TEGRA20_SPDIF_STROBE_CTRL_CLOCK_PERIOD_MASK		(0x3f << TEGRA20_SPDIF_STROBE_CTRL_CLOCK_PERIOD_SHIFT)
+
+/* Fields in SPDIF_DATA_FIFO_CSR */
+
+/* Clear Receiver User FIFO (RX USR.FIFO) */
+#define TEGRA20_SPDIF_DATA_FIFO_CSR_RU_CLR			(1 << 31)
+
+#define TEGRA20_SPDIF_FIFO_ATN_LVL_U_ONE_SLOT			0
+#define TEGRA20_SPDIF_FIFO_ATN_LVL_U_TWO_SLOTS			1
+#define TEGRA20_SPDIF_FIFO_ATN_LVL_U_THREE_SLOTS		2
+#define TEGRA20_SPDIF_FIFO_ATN_LVL_U_FOUR_SLOTS			3
+
+/* RU FIFO attention level */
+#define TEGRA20_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_SHIFT		29
+#define TEGRA20_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_MASK		\
+		(0x3                                      << TEGRA20_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_SHIFT)
+#define TEGRA20_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_RU1_WORD_FULL	\
+		(TEGRA20_SPDIF_FIFO_ATN_LVL_U_ONE_SLOT    << TEGRA20_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_SHIFT)
+#define TEGRA20_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_RU2_WORD_FULL	\
+		(TEGRA20_SPDIF_FIFO_ATN_LVL_U_TWO_SLOTS   << TEGRA20_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_SHIFT)
+#define TEGRA20_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_RU3_WORD_FULL	\
+		(TEGRA20_SPDIF_FIFO_ATN_LVL_U_THREE_SLOTS << TEGRA20_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_SHIFT)
+#define TEGRA20_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_RU4_WORD_FULL	\
+		(TEGRA20_SPDIF_FIFO_ATN_LVL_U_FOUR_SLOTS  << TEGRA20_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_SHIFT)
+
+/* Number of RX USR.FIFO levels with valid data. */
+#define TEGRA20_SPDIF_DATA_FIFO_CSR_RU_FULL_COUNT_SHIFT		24
+#define TEGRA20_SPDIF_DATA_FIFO_CSR_RU_FULL_COUNT_MASK		(0x1f << TEGRA20_SPDIF_DATA_FIFO_CSR_RU_FULL_COUNT_SHIFT)
+
+/* Clear Transmitter User FIFO (TX USR.FIFO) */
+#define TEGRA20_SPDIF_DATA_FIFO_CSR_TU_CLR			(1 << 23)
+
+/* TU FIFO attention level */
+#define TEGRA20_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_SHIFT		21
+#define TEGRA20_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_MASK		\
+		(0x3                                      << TEGRA20_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_SHIFT)
+#define TEGRA20_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_TU1_WORD_FULL	\
+		(TEGRA20_SPDIF_FIFO_ATN_LVL_U_ONE_SLOT    << TEGRA20_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_SHIFT)
+#define TEGRA20_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_TU2_WORD_FULL	\
+		(TEGRA20_SPDIF_FIFO_ATN_LVL_U_TWO_SLOTS   << TEGRA20_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_SHIFT)
+#define TEGRA20_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_TU3_WORD_FULL	\
+		(TEGRA20_SPDIF_FIFO_ATN_LVL_U_THREE_SLOTS << TEGRA20_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_SHIFT)
+#define TEGRA20_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_TU4_WORD_FULL	\
+		(TEGRA20_SPDIF_FIFO_ATN_LVL_U_FOUR_SLOTS  << TEGRA20_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_SHIFT)
+
+/* Number of TX USR.FIFO levels that could be filled. */
+#define TEGRA20_SPDIF_DATA_FIFO_CSR_TU_EMPTY_COUNT_SHIFT	16
+#define TEGRA20_SPDIF_DATA_FIFO_CSR_TU_EMPTY_COUNT_MASK		(0x1f << SPDIF_DATA_FIFO_CSR_TU_EMPTY_COUNT_SHIFT)
+
+/* Clear Receiver Data FIFO (RX DATA.FIFO) */
+#define TEGRA20_SPDIF_DATA_FIFO_CSR_RX_CLR			(1 << 15)
+
+#define TEGRA20_SPDIF_FIFO_ATN_LVL_D_ONE_SLOT			0
+#define TEGRA20_SPDIF_FIFO_ATN_LVL_D_FOUR_SLOTS			1
+#define TEGRA20_SPDIF_FIFO_ATN_LVL_D_EIGHT_SLOTS		2
+#define TEGRA20_SPDIF_FIFO_ATN_LVL_D_TWELVE_SLOTS		3
+
+/* RU FIFO attention level */
+#define TEGRA20_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_SHIFT		13
+#define TEGRA20_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_MASK		\
+		(0x3                                       << TEGRA20_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_SHIFT)
+#define TEGRA20_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_RU1_WORD_FULL	\
+		(TEGRA20_SPDIF_FIFO_ATN_LVL_D_ONE_SLOT     << TEGRA20_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_SHIFT)
+#define TEGRA20_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_RU4_WORD_FULL	\
+		(TEGRA20_SPDIF_FIFO_ATN_LVL_D_FOUR_SLOTS   << TEGRA20_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_SHIFT)
+#define TEGRA20_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_RU8_WORD_FULL	\
+		(TEGRA20_SPDIF_FIFO_ATN_LVL_D_EIGHT_SLOTS  << TEGRA20_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_SHIFT)
+#define TEGRA20_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_RU12_WORD_FULL	\
+		(TEGRA20_SPDIF_FIFO_ATN_LVL_D_TWELVE_SLOTS << TEGRA20_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_SHIFT)
+
+/* Number of RX DATA.FIFO levels with valid data. */
+#define TEGRA20_SPDIF_DATA_FIFO_CSR_RX_FULL_COUNT_SHIFT		8
+#define TEGRA20_SPDIF_DATA_FIFO_CSR_RX_FULL_COUNT_MASK		(0x1f << TEGRA20_SPDIF_DATA_FIFO_CSR_RX_FULL_COUNT_SHIFT)
+
+/* Clear Transmitter Data FIFO (TX DATA.FIFO) */
+#define TEGRA20_SPDIF_DATA_FIFO_CSR_TX_CLR			(1 << 7)
+
+/* TU FIFO attention level */
+#define TEGRA20_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_SHIFT		5
+#define TEGRA20_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_MASK		\
+		(0x3                                       << TEGRA20_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_SHIFT)
+#define TEGRA20_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_TU1_WORD_FULL	\
+		(TEGRA20_SPDIF_FIFO_ATN_LVL_D_ONE_SLOT     << TEGRA20_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_SHIFT)
+#define TEGRA20_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_TU4_WORD_FULL	\
+		(TEGRA20_SPDIF_FIFO_ATN_LVL_D_FOUR_SLOTS   << TEGRA20_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_SHIFT)
+#define TEGRA20_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_TU8_WORD_FULL	\
+		(TEGRA20_SPDIF_FIFO_ATN_LVL_D_EIGHT_SLOTS  << TEGRA20_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_SHIFT)
+#define TEGRA20_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_TU12_WORD_FULL	\
+		(TEGRA20_SPDIF_FIFO_ATN_LVL_D_TWELVE_SLOTS << TEGRA20_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_SHIFT)
+
+/* Number of TX DATA.FIFO levels that could be filled. */
+#define TEGRA20_SPDIF_DATA_FIFO_CSR_TX_EMPTY_COUNT_SHIFT	0
+#define TEGRA20_SPDIF_DATA_FIFO_CSR_TX_EMPTY_COUNT_MASK		(0x1f << SPDIF_DATA_FIFO_CSR_TX_EMPTY_COUNT_SHIFT)
+
+/* Fields in TEGRA20_SPDIF_DATA_OUT */
+
+/*
+ * This register has 5 different formats:
+ * 16-bit        (BIT_MODE=00, PACK=0)
+ * 20-bit        (BIT_MODE=01, PACK=0)
+ * 24-bit        (BIT_MODE=10, PACK=0)
+ * raw           (BIT_MODE=11, PACK=0)
+ * 16-bit packed (BIT_MODE=00, PACK=1)
+ */
+
+#define TEGRA20_SPDIF_DATA_OUT_DATA_16_SHIFT			0
+#define TEGRA20_SPDIF_DATA_OUT_DATA_16_MASK			(0xffff << TEGRA20_SPDIF_DATA_OUT_DATA_16_SHIFT)
+
+#define TEGRA20_SPDIF_DATA_OUT_DATA_20_SHIFT			0
+#define TEGRA20_SPDIF_DATA_OUT_DATA_20_MASK			(0xfffff << TEGRA20_SPDIF_DATA_OUT_DATA_20_SHIFT)
+
+#define TEGRA20_SPDIF_DATA_OUT_DATA_24_SHIFT			0
+#define TEGRA20_SPDIF_DATA_OUT_DATA_24_MASK			(0xffffff << TEGRA20_SPDIF_DATA_OUT_DATA_24_SHIFT)
+
+#define TEGRA20_SPDIF_DATA_OUT_DATA_RAW_P			(1 << 31)
+#define TEGRA20_SPDIF_DATA_OUT_DATA_RAW_C			(1 << 30)
+#define TEGRA20_SPDIF_DATA_OUT_DATA_RAW_U			(1 << 29)
+#define TEGRA20_SPDIF_DATA_OUT_DATA_RAW_V			(1 << 28)
+
+#define TEGRA20_SPDIF_DATA_OUT_DATA_RAW_DATA_SHIFT		8
+#define TEGRA20_SPDIF_DATA_OUT_DATA_RAW_DATA_MASK		(0xfffff << TEGRA20_SPDIF_DATA_OUT_DATA_RAW_DATA_SHIFT)
+
+#define TEGRA20_SPDIF_DATA_OUT_DATA_RAW_AUX_SHIFT		4
+#define TEGRA20_SPDIF_DATA_OUT_DATA_RAW_AUX_MASK		(0xf << TEGRA20_SPDIF_DATA_OUT_DATA_RAW_AUX_SHIFT)
+
+#define TEGRA20_SPDIF_DATA_OUT_DATA_RAW_PREAMBLE_SHIFT		0
+#define TEGRA20_SPDIF_DATA_OUT_DATA_RAW_PREAMBLE_MASK		(0xf << TEGRA20_SPDIF_DATA_OUT_DATA_RAW_PREAMBLE_SHIFT)
+
+#define TEGRA20_SPDIF_DATA_OUT_DATA_16_PACKED_RIGHT_SHIFT	16
+#define TEGRA20_SPDIF_DATA_OUT_DATA_16_PACKED_RIGHT_MASK	(0xffff << TEGRA20_SPDIF_DATA_OUT_DATA_16_PACKED_RIGHT_SHIFT)
+
+#define TEGRA20_SPDIF_DATA_OUT_DATA_16_PACKED_LEFT_SHIFT	0
+#define TEGRA20_SPDIF_DATA_OUT_DATA_16_PACKED_LEFT_MASK		(0xffff << TEGRA20_SPDIF_DATA_OUT_DATA_16_PACKED_LEFT_SHIFT)
+
+/* Fields in TEGRA20_SPDIF_DATA_IN */
+
+/*
+ * This register has 5 different formats:
+ * 16-bit        (BIT_MODE=00, PACK=0)
+ * 20-bit        (BIT_MODE=01, PACK=0)
+ * 24-bit        (BIT_MODE=10, PACK=0)
+ * raw           (BIT_MODE=11, PACK=0)
+ * 16-bit packed (BIT_MODE=00, PACK=1)
+ *
+ * Bits 31:24 are common to all modes except 16-bit packed
+ */
+
+#define TEGRA20_SPDIF_DATA_IN_DATA_P				(1 << 31)
+#define TEGRA20_SPDIF_DATA_IN_DATA_C				(1 << 30)
+#define TEGRA20_SPDIF_DATA_IN_DATA_U				(1 << 29)
+#define TEGRA20_SPDIF_DATA_IN_DATA_V				(1 << 28)
+
+#define TEGRA20_SPDIF_DATA_IN_DATA_PREAMBLE_SHIFT		24
+#define TEGRA20_SPDIF_DATA_IN_DATA_PREAMBLE_MASK		(0xf << TEGRA20_SPDIF_DATA_IN_DATA_PREAMBLE_SHIFT)
+
+#define TEGRA20_SPDIF_DATA_IN_DATA_16_SHIFT			0
+#define TEGRA20_SPDIF_DATA_IN_DATA_16_MASK			(0xffff << TEGRA20_SPDIF_DATA_IN_DATA_16_SHIFT)
+
+#define TEGRA20_SPDIF_DATA_IN_DATA_20_SHIFT			0
+#define TEGRA20_SPDIF_DATA_IN_DATA_20_MASK			(0xfffff << TEGRA20_SPDIF_DATA_IN_DATA_20_SHIFT)
+
+#define TEGRA20_SPDIF_DATA_IN_DATA_24_SHIFT			0
+#define TEGRA20_SPDIF_DATA_IN_DATA_24_MASK			(0xffffff << TEGRA20_SPDIF_DATA_IN_DATA_24_SHIFT)
+
+#define TEGRA20_SPDIF_DATA_IN_DATA_RAW_DATA_SHIFT		8
+#define TEGRA20_SPDIF_DATA_IN_DATA_RAW_DATA_MASK		(0xfffff << TEGRA20_SPDIF_DATA_IN_DATA_RAW_DATA_SHIFT)
+
+#define TEGRA20_SPDIF_DATA_IN_DATA_RAW_AUX_SHIFT		4
+#define TEGRA20_SPDIF_DATA_IN_DATA_RAW_AUX_MASK			(0xf << TEGRA20_SPDIF_DATA_IN_DATA_RAW_AUX_SHIFT)
+
+#define TEGRA20_SPDIF_DATA_IN_DATA_RAW_PREAMBLE_SHIFT		0
+#define TEGRA20_SPDIF_DATA_IN_DATA_RAW_PREAMBLE_MASK		(0xf << TEGRA20_SPDIF_DATA_IN_DATA_RAW_PREAMBLE_SHIFT)
+
+#define TEGRA20_SPDIF_DATA_IN_DATA_16_PACKED_RIGHT_SHIFT	16
+#define TEGRA20_SPDIF_DATA_IN_DATA_16_PACKED_RIGHT_MASK		(0xffff << TEGRA20_SPDIF_DATA_IN_DATA_16_PACKED_RIGHT_SHIFT)
+
+#define TEGRA20_SPDIF_DATA_IN_DATA_16_PACKED_LEFT_SHIFT		0
+#define TEGRA20_SPDIF_DATA_IN_DATA_16_PACKED_LEFT_MASK		(0xffff << TEGRA20_SPDIF_DATA_IN_DATA_16_PACKED_LEFT_SHIFT)
+
+/* Fields in TEGRA20_SPDIF_CH_STA_RX_A */
+/* Fields in TEGRA20_SPDIF_CH_STA_RX_B */
+/* Fields in TEGRA20_SPDIF_CH_STA_RX_C */
+/* Fields in TEGRA20_SPDIF_CH_STA_RX_D */
+/* Fields in TEGRA20_SPDIF_CH_STA_RX_E */
+/* Fields in TEGRA20_SPDIF_CH_STA_RX_F */
+
+/*
+ * The 6-word receive channel data page buffer holds a block (192 frames) of
+ * channel status information. The order of receive is from LSB to MSB
+ * bit, and from CH_STA_RX_A to CH_STA_RX_F then back to CH_STA_RX_A.
+ */
+
+/* Fields in TEGRA20_SPDIF_CH_STA_TX_A */
+/* Fields in TEGRA20_SPDIF_CH_STA_TX_B */
+/* Fields in TEGRA20_SPDIF_CH_STA_TX_C */
+/* Fields in TEGRA20_SPDIF_CH_STA_TX_D */
+/* Fields in TEGRA20_SPDIF_CH_STA_TX_E */
+/* Fields in TEGRA20_SPDIF_CH_STA_TX_F */
+
+/*
+ * The 6-word transmit channel data page buffer holds a block (192 frames) of
+ * channel status information. The order of transmission is from LSB to MSB
+ * bit, and from CH_STA_TX_A to CH_STA_TX_F then back to CH_STA_TX_A.
+ */
+
+/* Fields in TEGRA20_SPDIF_USR_STA_RX_A */
+
+/*
+ * This 4-word deep FIFO receives user FIFO field information. The order of
+ * receive is from LSB to MSB bit.
+ */
+
+/* Fields in TEGRA20_SPDIF_USR_DAT_TX_A */
+
+/*
+ * This 4-word deep FIFO transmits user FIFO field information. The order of
+ * transmission is from LSB to MSB bit.
+ */
+
+struct tegra20_spdif {
+	struct clk *clk_spdif_out;
+	struct tegra_pcm_dma_params capture_dma_data;
+	struct tegra_pcm_dma_params playback_dma_data;
+	struct regmap *regmap;
+	u32 reg_ctrl;
+};
+
+#endif
diff --git a/sound/soc/tegra/tegra30_ahub.c b/sound/soc/tegra/tegra30_ahub.c
new file mode 100644
index 000000000000..57cd419f743e
--- /dev/null
+++ b/sound/soc/tegra/tegra30_ahub.c
@@ -0,0 +1,631 @@
+/*
+ * tegra30_ahub.c - Tegra30 AHUB driver
+ *
+ * Copyright (c) 2011,2012, NVIDIA CORPORATION.  All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms and conditions of the GNU General Public License,
+ * version 2, as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope 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, see <http://www.gnu.org/licenses/>.
+ */
+
+#include <linux/clk.h>
+#include <linux/device.h>
+#include <linux/io.h>
+#include <linux/module.h>
+#include <linux/of_platform.h>
+#include <linux/platform_device.h>
+#include <linux/pm_runtime.h>
+#include <linux/regmap.h>
+#include <linux/slab.h>
+#include <mach/clk.h>
+#include <mach/dma.h>
+#include <sound/soc.h>
+#include "tegra30_ahub.h"
+
+#define DRV_NAME "tegra30-ahub"
+
+static struct tegra30_ahub *ahub;
+
+static inline void tegra30_apbif_write(u32 reg, u32 val)
+{
+	regmap_write(ahub->regmap_apbif, reg, val);
+}
+
+static inline u32 tegra30_apbif_read(u32 reg)
+{
+	u32 val;
+	regmap_read(ahub->regmap_apbif, reg, &val);
+	return val;
+}
+
+static inline void tegra30_audio_write(u32 reg, u32 val)
+{
+	regmap_write(ahub->regmap_ahub, reg, val);
+}
+
+static int tegra30_ahub_runtime_suspend(struct device *dev)
+{
+	regcache_cache_only(ahub->regmap_apbif, true);
+	regcache_cache_only(ahub->regmap_ahub, true);
+
+	clk_disable(ahub->clk_apbif);
+	clk_disable(ahub->clk_d_audio);
+
+	return 0;
+}
+
+/*
+ * clk_apbif isn't required for an I2S<->I2S configuration where no PCM data
+ * is read from or sent to memory. However, that's not something the rest of
+ * the driver supports right now, so we'll just treat the two clocks as one
+ * for now.
+ *
+ * These functions should not be a plain ref-count. Instead, each active stream
+ * contributes some requirement to the minimum clock rate, so starting or
+ * stopping streams should dynamically adjust the clock as required.  However,
+ * this is not yet implemented.
+ */
+static int tegra30_ahub_runtime_resume(struct device *dev)
+{
+	int ret;
+
+	ret = clk_enable(ahub->clk_d_audio);
+	if (ret) {
+		dev_err(dev, "clk_enable d_audio failed: %d\n", ret);
+		return ret;
+	}
+	ret = clk_enable(ahub->clk_apbif);
+	if (ret) {
+		dev_err(dev, "clk_enable apbif failed: %d\n", ret);
+		clk_disable(ahub->clk_d_audio);
+		return ret;
+	}
+
+	regcache_cache_only(ahub->regmap_apbif, false);
+	regcache_cache_only(ahub->regmap_ahub, false);
+
+	return 0;
+}
+
+int tegra30_ahub_allocate_rx_fifo(enum tegra30_ahub_rxcif *rxcif,
+				  unsigned long *fiforeg,
+				  unsigned long *reqsel)
+{
+	int channel;
+	u32 reg, val;
+
+	channel = find_first_zero_bit(ahub->rx_usage,
+				      TEGRA30_AHUB_CHANNEL_CTRL_COUNT);
+	if (channel >= TEGRA30_AHUB_CHANNEL_CTRL_COUNT)
+		return -EBUSY;
+
+	__set_bit(channel, ahub->rx_usage);
+
+	*rxcif = TEGRA30_AHUB_RXCIF_APBIF_RX0 + channel;
+	*fiforeg = ahub->apbif_addr + TEGRA30_AHUB_CHANNEL_RXFIFO +
+		   (channel * TEGRA30_AHUB_CHANNEL_RXFIFO_STRIDE);
+	*reqsel = ahub->dma_sel + channel;
+
+	reg = TEGRA30_AHUB_CHANNEL_CTRL +
+	      (channel * TEGRA30_AHUB_CHANNEL_CTRL_STRIDE);
+	val = tegra30_apbif_read(reg);
+	val &= ~(TEGRA30_AHUB_CHANNEL_CTRL_RX_THRESHOLD_MASK |
+		 TEGRA30_AHUB_CHANNEL_CTRL_RX_PACK_MASK);
+	val |= (7 << TEGRA30_AHUB_CHANNEL_CTRL_RX_THRESHOLD_SHIFT) |
+	       TEGRA30_AHUB_CHANNEL_CTRL_RX_PACK_EN |
+	       TEGRA30_AHUB_CHANNEL_CTRL_RX_PACK_16;
+	tegra30_apbif_write(reg, val);
+
+	reg = TEGRA30_AHUB_CIF_RX_CTRL +
+	      (channel * TEGRA30_AHUB_CIF_RX_CTRL_STRIDE);
+	val = (0 << TEGRA30_AUDIOCIF_CTRL_FIFO_THRESHOLD_SHIFT) |
+	      (1 << TEGRA30_AUDIOCIF_CTRL_AUDIO_CHANNELS_SHIFT) |
+	      (1 << TEGRA30_AUDIOCIF_CTRL_CLIENT_CHANNELS_SHIFT) |
+	      TEGRA30_AUDIOCIF_CTRL_AUDIO_BITS_16 |
+	      TEGRA30_AUDIOCIF_CTRL_CLIENT_BITS_16 |
+	      TEGRA30_AUDIOCIF_CTRL_DIRECTION_RX;
+	tegra30_apbif_write(reg, val);
+
+	return 0;
+}
+EXPORT_SYMBOL_GPL(tegra30_ahub_allocate_rx_fifo);
+
+int tegra30_ahub_enable_rx_fifo(enum tegra30_ahub_rxcif rxcif)
+{
+	int channel = rxcif - TEGRA30_AHUB_RXCIF_APBIF_RX0;
+	int reg, val;
+
+	reg = TEGRA30_AHUB_CHANNEL_CTRL +
+	      (channel * TEGRA30_AHUB_CHANNEL_CTRL_STRIDE);
+	val = tegra30_apbif_read(reg);
+	val |= TEGRA30_AHUB_CHANNEL_CTRL_RX_EN;
+	tegra30_apbif_write(reg, val);
+
+	return 0;
+}
+EXPORT_SYMBOL_GPL(tegra30_ahub_enable_rx_fifo);
+
+int tegra30_ahub_disable_rx_fifo(enum tegra30_ahub_rxcif rxcif)
+{
+	int channel = rxcif - TEGRA30_AHUB_RXCIF_APBIF_RX0;
+	int reg, val;
+
+	reg = TEGRA30_AHUB_CHANNEL_CTRL +
+	      (channel * TEGRA30_AHUB_CHANNEL_CTRL_STRIDE);
+	val = tegra30_apbif_read(reg);
+	val &= ~TEGRA30_AHUB_CHANNEL_CTRL_RX_EN;
+	tegra30_apbif_write(reg, val);
+
+	return 0;
+}
+EXPORT_SYMBOL_GPL(tegra30_ahub_disable_rx_fifo);
+
+int tegra30_ahub_free_rx_fifo(enum tegra30_ahub_rxcif rxcif)
+{
+	int channel = rxcif - TEGRA30_AHUB_RXCIF_APBIF_RX0;
+
+	__clear_bit(channel, ahub->rx_usage);
+
+	return 0;
+}
+EXPORT_SYMBOL_GPL(tegra30_ahub_free_rx_fifo);
+
+int tegra30_ahub_allocate_tx_fifo(enum tegra30_ahub_txcif *txcif,
+				  unsigned long *fiforeg,
+				  unsigned long *reqsel)
+{
+	int channel;
+	u32 reg, val;
+
+	channel = find_first_zero_bit(ahub->tx_usage,
+				      TEGRA30_AHUB_CHANNEL_CTRL_COUNT);
+	if (channel >= TEGRA30_AHUB_CHANNEL_CTRL_COUNT)
+		return -EBUSY;
+
+	__set_bit(channel, ahub->tx_usage);
+
+	*txcif = TEGRA30_AHUB_TXCIF_APBIF_TX0 + channel;
+	*fiforeg = ahub->apbif_addr + TEGRA30_AHUB_CHANNEL_TXFIFO +
+		   (channel * TEGRA30_AHUB_CHANNEL_TXFIFO_STRIDE);
+	*reqsel = ahub->dma_sel + channel;
+
+	reg = TEGRA30_AHUB_CHANNEL_CTRL +
+	      (channel * TEGRA30_AHUB_CHANNEL_CTRL_STRIDE);
+	val = tegra30_apbif_read(reg);
+	val &= ~(TEGRA30_AHUB_CHANNEL_CTRL_TX_THRESHOLD_MASK |
+		 TEGRA30_AHUB_CHANNEL_CTRL_TX_PACK_MASK);
+	val |= (7 << TEGRA30_AHUB_CHANNEL_CTRL_TX_THRESHOLD_SHIFT) |
+	       TEGRA30_AHUB_CHANNEL_CTRL_TX_PACK_EN |
+	       TEGRA30_AHUB_CHANNEL_CTRL_TX_PACK_16;
+	tegra30_apbif_write(reg, val);
+
+	reg = TEGRA30_AHUB_CIF_TX_CTRL +
+	      (channel * TEGRA30_AHUB_CIF_TX_CTRL_STRIDE);
+	val = (0 << TEGRA30_AUDIOCIF_CTRL_FIFO_THRESHOLD_SHIFT) |
+	      (1 << TEGRA30_AUDIOCIF_CTRL_AUDIO_CHANNELS_SHIFT) |
+	      (1 << TEGRA30_AUDIOCIF_CTRL_CLIENT_CHANNELS_SHIFT) |
+	      TEGRA30_AUDIOCIF_CTRL_AUDIO_BITS_16 |
+	      TEGRA30_AUDIOCIF_CTRL_CLIENT_BITS_16 |
+	      TEGRA30_AUDIOCIF_CTRL_DIRECTION_TX;
+	tegra30_apbif_write(reg, val);
+
+	return 0;
+}
+EXPORT_SYMBOL_GPL(tegra30_ahub_allocate_tx_fifo);
+
+int tegra30_ahub_enable_tx_fifo(enum tegra30_ahub_txcif txcif)
+{
+	int channel = txcif - TEGRA30_AHUB_TXCIF_APBIF_TX0;
+	int reg, val;
+
+	reg = TEGRA30_AHUB_CHANNEL_CTRL +
+	      (channel * TEGRA30_AHUB_CHANNEL_CTRL_STRIDE);
+	val = tegra30_apbif_read(reg);
+	val |= TEGRA30_AHUB_CHANNEL_CTRL_TX_EN;
+	tegra30_apbif_write(reg, val);
+
+	return 0;
+}
+EXPORT_SYMBOL_GPL(tegra30_ahub_enable_tx_fifo);
+
+int tegra30_ahub_disable_tx_fifo(enum tegra30_ahub_txcif txcif)
+{
+	int channel = txcif - TEGRA30_AHUB_TXCIF_APBIF_TX0;
+	int reg, val;
+
+	reg = TEGRA30_AHUB_CHANNEL_CTRL +
+	      (channel * TEGRA30_AHUB_CHANNEL_CTRL_STRIDE);
+	val = tegra30_apbif_read(reg);
+	val &= ~TEGRA30_AHUB_CHANNEL_CTRL_TX_EN;
+	tegra30_apbif_write(reg, val);
+
+	return 0;
+}
+EXPORT_SYMBOL_GPL(tegra30_ahub_disable_tx_fifo);
+
+int tegra30_ahub_free_tx_fifo(enum tegra30_ahub_txcif txcif)
+{
+	int channel = txcif - TEGRA30_AHUB_TXCIF_APBIF_TX0;
+
+	__clear_bit(channel, ahub->tx_usage);
+
+	return 0;
+}
+EXPORT_SYMBOL_GPL(tegra30_ahub_free_tx_fifo);
+
+int tegra30_ahub_set_rx_cif_source(enum tegra30_ahub_rxcif rxcif,
+				   enum tegra30_ahub_txcif txcif)
+{
+	int channel = rxcif - TEGRA30_AHUB_RXCIF_APBIF_RX0;
+	int reg;
+
+	reg = TEGRA30_AHUB_AUDIO_RX +
+	      (channel * TEGRA30_AHUB_AUDIO_RX_STRIDE);
+	tegra30_audio_write(reg, 1 << txcif);
+
+	return 0;
+}
+EXPORT_SYMBOL_GPL(tegra30_ahub_set_rx_cif_source);
+
+int tegra30_ahub_unset_rx_cif_source(enum tegra30_ahub_rxcif rxcif)
+{
+	int channel = rxcif - TEGRA30_AHUB_RXCIF_APBIF_RX0;
+	int reg;
+
+	reg = TEGRA30_AHUB_AUDIO_RX +
+	      (channel * TEGRA30_AHUB_AUDIO_RX_STRIDE);
+	tegra30_audio_write(reg, 0);
+
+	return 0;
+}
+EXPORT_SYMBOL_GPL(tegra30_ahub_unset_rx_cif_source);
+
+static const char * const configlink_clocks[] __devinitconst = {
+	"i2s0",
+	"i2s1",
+	"i2s2",
+	"i2s3",
+	"i2s4",
+	"dam0",
+	"dam1",
+	"dam2",
+	"spdif_in",
+};
+
+struct of_dev_auxdata ahub_auxdata[] __devinitdata = {
+	OF_DEV_AUXDATA("nvidia,tegra30-i2s", 0x70080300, "tegra30-i2s.0", NULL),
+	OF_DEV_AUXDATA("nvidia,tegra30-i2s", 0x70080400, "tegra30-i2s.1", NULL),
+	OF_DEV_AUXDATA("nvidia,tegra30-i2s", 0x70080500, "tegra30-i2s.2", NULL),
+	OF_DEV_AUXDATA("nvidia,tegra30-i2s", 0x70080600, "tegra30-i2s.3", NULL),
+	OF_DEV_AUXDATA("nvidia,tegra30-i2s", 0x70080700, "tegra30-i2s.4", NULL),
+	{}
+};
+
+#define LAST_REG(name) \
+	(TEGRA30_AHUB_##name + \
+	 (TEGRA30_AHUB_##name##_STRIDE * TEGRA30_AHUB_##name##_COUNT) - 4)
+
+#define REG_IN_ARRAY(reg, name) \
+	((reg >= TEGRA30_AHUB_##name) && \
+	 (reg <= LAST_REG(name) && \
+	 (!((reg - TEGRA30_AHUB_##name) % TEGRA30_AHUB_##name##_STRIDE))))
+
+static bool tegra30_ahub_apbif_wr_rd_reg(struct device *dev, unsigned int reg)
+{
+	switch (reg) {
+	case TEGRA30_AHUB_CONFIG_LINK_CTRL:
+	case TEGRA30_AHUB_MISC_CTRL:
+	case TEGRA30_AHUB_APBDMA_LIVE_STATUS:
+	case TEGRA30_AHUB_I2S_LIVE_STATUS:
+	case TEGRA30_AHUB_SPDIF_LIVE_STATUS:
+	case TEGRA30_AHUB_I2S_INT_MASK:
+	case TEGRA30_AHUB_DAM_INT_MASK:
+	case TEGRA30_AHUB_SPDIF_INT_MASK:
+	case TEGRA30_AHUB_APBIF_INT_MASK:
+	case TEGRA30_AHUB_I2S_INT_STATUS:
+	case TEGRA30_AHUB_DAM_INT_STATUS:
+	case TEGRA30_AHUB_SPDIF_INT_STATUS:
+	case TEGRA30_AHUB_APBIF_INT_STATUS:
+	case TEGRA30_AHUB_I2S_INT_SOURCE:
+	case TEGRA30_AHUB_DAM_INT_SOURCE:
+	case TEGRA30_AHUB_SPDIF_INT_SOURCE:
+	case TEGRA30_AHUB_APBIF_INT_SOURCE:
+	case TEGRA30_AHUB_I2S_INT_SET:
+	case TEGRA30_AHUB_DAM_INT_SET:
+	case TEGRA30_AHUB_SPDIF_INT_SET:
+	case TEGRA30_AHUB_APBIF_INT_SET:
+		return true;
+	default:
+		break;
+	};
+
+	if (REG_IN_ARRAY(reg, CHANNEL_CTRL) ||
+	    REG_IN_ARRAY(reg, CHANNEL_CLEAR) ||
+	    REG_IN_ARRAY(reg, CHANNEL_STATUS) ||
+	    REG_IN_ARRAY(reg, CHANNEL_TXFIFO) ||
+	    REG_IN_ARRAY(reg, CHANNEL_RXFIFO) ||
+	    REG_IN_ARRAY(reg, CIF_TX_CTRL) ||
+	    REG_IN_ARRAY(reg, CIF_RX_CTRL) ||
+	    REG_IN_ARRAY(reg, DAM_LIVE_STATUS))
+		return true;
+
+	return false;
+}
+
+static bool tegra30_ahub_apbif_volatile_reg(struct device *dev,
+					    unsigned int reg)
+{
+	switch (reg) {
+	case TEGRA30_AHUB_CONFIG_LINK_CTRL:
+	case TEGRA30_AHUB_MISC_CTRL:
+	case TEGRA30_AHUB_APBDMA_LIVE_STATUS:
+	case TEGRA30_AHUB_I2S_LIVE_STATUS:
+	case TEGRA30_AHUB_SPDIF_LIVE_STATUS:
+	case TEGRA30_AHUB_I2S_INT_STATUS:
+	case TEGRA30_AHUB_DAM_INT_STATUS:
+	case TEGRA30_AHUB_SPDIF_INT_STATUS:
+	case TEGRA30_AHUB_APBIF_INT_STATUS:
+	case TEGRA30_AHUB_I2S_INT_SET:
+	case TEGRA30_AHUB_DAM_INT_SET:
+	case TEGRA30_AHUB_SPDIF_INT_SET:
+	case TEGRA30_AHUB_APBIF_INT_SET:
+		return true;
+	default:
+		break;
+	};
+
+	if (REG_IN_ARRAY(reg, CHANNEL_CLEAR) ||
+	    REG_IN_ARRAY(reg, CHANNEL_STATUS) ||
+	    REG_IN_ARRAY(reg, CHANNEL_TXFIFO) ||
+	    REG_IN_ARRAY(reg, CHANNEL_RXFIFO) ||
+	    REG_IN_ARRAY(reg, DAM_LIVE_STATUS))
+		return true;
+
+	return false;
+}
+
+static bool tegra30_ahub_apbif_precious_reg(struct device *dev,
+					    unsigned int reg)
+{
+	if (REG_IN_ARRAY(reg, CHANNEL_TXFIFO) ||
+	    REG_IN_ARRAY(reg, CHANNEL_RXFIFO))
+		return true;
+
+	return false;
+}
+
+static const struct regmap_config tegra30_ahub_apbif_regmap_config = {
+	.name = "apbif",
+	.reg_bits = 32,
+	.val_bits = 32,
+	.reg_stride = 4,
+	.max_register = TEGRA30_AHUB_APBIF_INT_SET,
+	.writeable_reg = tegra30_ahub_apbif_wr_rd_reg,
+	.readable_reg = tegra30_ahub_apbif_wr_rd_reg,
+	.volatile_reg = tegra30_ahub_apbif_volatile_reg,
+	.precious_reg = tegra30_ahub_apbif_precious_reg,
+	.cache_type = REGCACHE_RBTREE,
+};
+
+static bool tegra30_ahub_ahub_wr_rd_reg(struct device *dev, unsigned int reg)
+{
+	if (REG_IN_ARRAY(reg, AUDIO_RX))
+		return true;
+
+	return false;
+}
+
+static const struct regmap_config tegra30_ahub_ahub_regmap_config = {
+	.name = "ahub",
+	.reg_bits = 32,
+	.val_bits = 32,
+	.reg_stride = 4,
+	.max_register = LAST_REG(AUDIO_RX),
+	.writeable_reg = tegra30_ahub_ahub_wr_rd_reg,
+	.readable_reg = tegra30_ahub_ahub_wr_rd_reg,
+	.cache_type = REGCACHE_RBTREE,
+};
+
+static int __devinit tegra30_ahub_probe(struct platform_device *pdev)
+{
+	struct clk *clk;
+	int i;
+	struct resource *res0, *res1, *region;
+	u32 of_dma[2];
+	void __iomem *regs_apbif, *regs_ahub;
+	int ret = 0;
+
+	if (ahub)
+		return -ENODEV;
+
+	/*
+	 * The AHUB hosts a register bus: the "configlink". For this to
+	 * operate correctly, all devices on this bus must be out of reset.
+	 * Ensure that here.
+	 */
+	for (i = 0; i < ARRAY_SIZE(configlink_clocks); i++) {
+		clk = clk_get_sys(NULL, configlink_clocks[i]);
+		if (IS_ERR(clk)) {
+			dev_err(&pdev->dev, "Can't get clock %s\n",
+				configlink_clocks[i]);
+			ret = PTR_ERR(clk);
+			goto err;
+		}
+		tegra_periph_reset_deassert(clk);
+		clk_put(clk);
+	}
+
+	ahub = devm_kzalloc(&pdev->dev, sizeof(struct tegra30_ahub),
+			    GFP_KERNEL);
+	if (!ahub) {
+		dev_err(&pdev->dev, "Can't allocate tegra30_ahub\n");
+		ret = -ENOMEM;
+		goto err;
+	}
+	dev_set_drvdata(&pdev->dev, ahub);
+
+	ahub->dev = &pdev->dev;
+
+	ahub->clk_d_audio = clk_get(&pdev->dev, "d_audio");
+	if (IS_ERR(ahub->clk_d_audio)) {
+		dev_err(&pdev->dev, "Can't retrieve ahub d_audio clock\n");
+		ret = PTR_ERR(ahub->clk_d_audio);
+		goto err;
+	}
+
+	ahub->clk_apbif = clk_get(&pdev->dev, "apbif");
+	if (IS_ERR(ahub->clk_apbif)) {
+		dev_err(&pdev->dev, "Can't retrieve ahub apbif clock\n");
+		ret = PTR_ERR(ahub->clk_apbif);
+		goto err_clk_put_d_audio;
+	}
+
+	if (of_property_read_u32_array(pdev->dev.of_node,
+				"nvidia,dma-request-selector",
+				of_dma, 2) < 0) {
+		dev_err(&pdev->dev,
+			"Missing property nvidia,dma-request-selector\n");
+		ret = -ENODEV;
+		goto err_clk_put_d_audio;
+	}
+	ahub->dma_sel = of_dma[1];
+
+	res0 = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+	if (!res0) {
+		dev_err(&pdev->dev, "No apbif memory resource\n");
+		ret = -ENODEV;
+		goto err_clk_put_apbif;
+	}
+
+	region = devm_request_mem_region(&pdev->dev, res0->start,
+					 resource_size(res0), DRV_NAME);
+	if (!region) {
+		dev_err(&pdev->dev, "request region apbif failed\n");
+		ret = -EBUSY;
+		goto err_clk_put_apbif;
+	}
+	ahub->apbif_addr = res0->start;
+
+	regs_apbif = devm_ioremap(&pdev->dev, res0->start,
+				  resource_size(res0));
+	if (!regs_apbif) {
+		dev_err(&pdev->dev, "ioremap apbif failed\n");
+		ret = -ENOMEM;
+		goto err_clk_put_apbif;
+	}
+
+	ahub->regmap_apbif = devm_regmap_init_mmio(&pdev->dev, regs_apbif,
+					&tegra30_ahub_apbif_regmap_config);
+	if (IS_ERR(ahub->regmap_apbif)) {
+		dev_err(&pdev->dev, "apbif regmap init failed\n");
+		ret = PTR_ERR(ahub->regmap_apbif);
+		goto err_clk_put_apbif;
+	}
+	regcache_cache_only(ahub->regmap_apbif, true);
+
+	res1 = platform_get_resource(pdev, IORESOURCE_MEM, 1);
+	if (!res1) {
+		dev_err(&pdev->dev, "No ahub memory resource\n");
+		ret = -ENODEV;
+		goto err_clk_put_apbif;
+	}
+
+	region = devm_request_mem_region(&pdev->dev, res1->start,
+					 resource_size(res1), DRV_NAME);
+	if (!region) {
+		dev_err(&pdev->dev, "request region ahub failed\n");
+		ret = -EBUSY;
+		goto err_clk_put_apbif;
+	}
+
+	regs_ahub = devm_ioremap(&pdev->dev, res1->start,
+				 resource_size(res1));
+	if (!regs_ahub) {
+		dev_err(&pdev->dev, "ioremap ahub failed\n");
+		ret = -ENOMEM;
+		goto err_clk_put_apbif;
+	}
+
+	ahub->regmap_ahub = devm_regmap_init_mmio(&pdev->dev, regs_ahub,
+					&tegra30_ahub_ahub_regmap_config);
+	if (IS_ERR(ahub->regmap_ahub)) {
+		dev_err(&pdev->dev, "ahub regmap init failed\n");
+		ret = PTR_ERR(ahub->regmap_ahub);
+		goto err_clk_put_apbif;
+	}
+	regcache_cache_only(ahub->regmap_ahub, true);
+
+	pm_runtime_enable(&pdev->dev);
+	if (!pm_runtime_enabled(&pdev->dev)) {
+		ret = tegra30_ahub_runtime_resume(&pdev->dev);
+		if (ret)
+			goto err_pm_disable;
+	}
+
+	of_platform_populate(pdev->dev.of_node, NULL, ahub_auxdata,
+			     &pdev->dev);
+
+	return 0;
+
+err_pm_disable:
+	pm_runtime_disable(&pdev->dev);
+err_clk_put_apbif:
+	clk_put(ahub->clk_apbif);
+err_clk_put_d_audio:
+	clk_put(ahub->clk_d_audio);
+	ahub = 0;
+err:
+	return ret;
+}
+
+static int __devexit tegra30_ahub_remove(struct platform_device *pdev)
+{
+	if (!ahub)
+		return -ENODEV;
+
+	pm_runtime_disable(&pdev->dev);
+	if (!pm_runtime_status_suspended(&pdev->dev))
+		tegra30_ahub_runtime_suspend(&pdev->dev);
+
+	clk_put(ahub->clk_apbif);
+	clk_put(ahub->clk_d_audio);
+
+	ahub = 0;
+
+	return 0;
+}
+
+static const struct of_device_id tegra30_ahub_of_match[] __devinitconst = {
+	{ .compatible = "nvidia,tegra30-ahub", },
+	{},
+};
+
+static const struct dev_pm_ops tegra30_ahub_pm_ops __devinitconst = {
+	SET_RUNTIME_PM_OPS(tegra30_ahub_runtime_suspend,
+			   tegra30_ahub_runtime_resume, NULL)
+};
+
+static struct platform_driver tegra30_ahub_driver = {
+	.probe = tegra30_ahub_probe,
+	.remove = __devexit_p(tegra30_ahub_remove),
+	.driver = {
+		.name = DRV_NAME,
+		.owner = THIS_MODULE,
+		.of_match_table = tegra30_ahub_of_match,
+		.pm = &tegra30_ahub_pm_ops,
+	},
+};
+module_platform_driver(tegra30_ahub_driver);
+
+MODULE_AUTHOR("Stephen Warren <swarren@nvidia.com>");
+MODULE_DESCRIPTION("Tegra30 AHUB driver");
+MODULE_LICENSE("GPL v2");
+MODULE_ALIAS("platform:" DRV_NAME);
diff --git a/sound/soc/tegra/tegra30_ahub.h b/sound/soc/tegra/tegra30_ahub.h
new file mode 100644
index 000000000000..e690e2eecc92
--- /dev/null
+++ b/sound/soc/tegra/tegra30_ahub.h
@@ -0,0 +1,483 @@
+/*
+ * tegra30_ahub.h - Definitions for Tegra30 AHUB driver
+ *
+ * Copyright (c) 2011,2012, NVIDIA CORPORATION.  All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms and conditions of the GNU General Public License,
+ * version 2, as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope 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, see <http://www.gnu.org/licenses/>.
+ */
+
+#ifndef __TEGRA30_AHUB_H__
+#define __TEGRA30_AHUB_H__
+
+/* Fields in *_CIF_RX/TX_CTRL; used by AHUB FIFOs, and all other audio modules */
+
+#define TEGRA30_AUDIOCIF_CTRL_FIFO_THRESHOLD_SHIFT	28
+#define TEGRA30_AUDIOCIF_CTRL_FIFO_THRESHOLD_MASK_US	0xf
+#define TEGRA30_AUDIOCIF_CTRL_FIFO_THRESHOLD_MASK	(TEGRA30_AUDIOCIF_CTRL_FIFO_THRESHOLD_MASK_US << TEGRA30_AUDIOCIF_CTRL_FIFO_THRESHOLD_SHIFT)
+
+/* Channel count minus 1 */
+#define TEGRA30_AUDIOCIF_CTRL_AUDIO_CHANNELS_SHIFT	24
+#define TEGRA30_AUDIOCIF_CTRL_AUDIO_CHANNELS_MASK_US	7
+#define TEGRA30_AUDIOCIF_CTRL_AUDIO_CHANNELS_MASK	(TEGRA30_AUDIOCIF_CTRL_AUDIO_CHANNELS_MASK_US << TEGRA30_AUDIOCIF_CTRL_AUDIO_CHANNELS_SHIFT)
+
+/* Channel count minus 1 */
+#define TEGRA30_AUDIOCIF_CTRL_CLIENT_CHANNELS_SHIFT	16
+#define TEGRA30_AUDIOCIF_CTRL_CLIENT_CHANNELS_MASK_US	7
+#define TEGRA30_AUDIOCIF_CTRL_CLIENT_CHANNELS_MASK	(TEGRA30_AUDIOCIF_CTRL_CLIENT_CHANNELS_MASK_US << TEGRA30_AUDIOCIF_CTRL_CLIENT_CHANNELS_SHIFT)
+
+#define TEGRA30_AUDIOCIF_BITS_4				0
+#define TEGRA30_AUDIOCIF_BITS_8				1
+#define TEGRA30_AUDIOCIF_BITS_12			2
+#define TEGRA30_AUDIOCIF_BITS_16			3
+#define TEGRA30_AUDIOCIF_BITS_20			4
+#define TEGRA30_AUDIOCIF_BITS_24			5
+#define TEGRA30_AUDIOCIF_BITS_28			6
+#define TEGRA30_AUDIOCIF_BITS_32			7
+
+#define TEGRA30_AUDIOCIF_CTRL_AUDIO_BITS_SHIFT		12
+#define TEGRA30_AUDIOCIF_CTRL_AUDIO_BITS_MASK		(7                        << TEGRA30_AUDIOCIF_CTRL_AUDIO_BITS_SHIFT)
+#define TEGRA30_AUDIOCIF_CTRL_AUDIO_BITS_4		(TEGRA30_AUDIOCIF_BITS_4  << TEGRA30_AUDIOCIF_CTRL_AUDIO_BITS_SHIFT)
+#define TEGRA30_AUDIOCIF_CTRL_AUDIO_BITS_8		(TEGRA30_AUDIOCIF_BITS_8  << TEGRA30_AUDIOCIF_CTRL_AUDIO_BITS_SHIFT)
+#define TEGRA30_AUDIOCIF_CTRL_AUDIO_BITS_12		(TEGRA30_AUDIOCIF_BITS_12 << TEGRA30_AUDIOCIF_CTRL_AUDIO_BITS_SHIFT)
+#define TEGRA30_AUDIOCIF_CTRL_AUDIO_BITS_16		(TEGRA30_AUDIOCIF_BITS_16 << TEGRA30_AUDIOCIF_CTRL_AUDIO_BITS_SHIFT)
+#define TEGRA30_AUDIOCIF_CTRL_AUDIO_BITS_20		(TEGRA30_AUDIOCIF_BITS_20 << TEGRA30_AUDIOCIF_CTRL_AUDIO_BITS_SHIFT)
+#define TEGRA30_AUDIOCIF_CTRL_AUDIO_BITS_24		(TEGRA30_AUDIOCIF_BITS_24 << TEGRA30_AUDIOCIF_CTRL_AUDIO_BITS_SHIFT)
+#define TEGRA30_AUDIOCIF_CTRL_AUDIO_BITS_28		(TEGRA30_AUDIOCIF_BITS_28 << TEGRA30_AUDIOCIF_CTRL_AUDIO_BITS_SHIFT)
+#define TEGRA30_AUDIOCIF_CTRL_AUDIO_BITS_32		(TEGRA30_AUDIOCIF_BITS_32 << TEGRA30_AUDIOCIF_CTRL_AUDIO_BITS_SHIFT)
+
+#define TEGRA30_AUDIOCIF_CTRL_CLIENT_BITS_SHIFT		8
+#define TEGRA30_AUDIOCIF_CTRL_CLIENT_BITS_MASK		(7                        << TEGRA30_AUDIOCIF_CTRL_CLIENT_BITS_SHIFT)
+#define TEGRA30_AUDIOCIF_CTRL_CLIENT_BITS_4		(TEGRA30_AUDIOCIF_BITS_4  << TEGRA30_AUDIOCIF_CTRL_CLIENT_BITS_SHIFT)
+#define TEGRA30_AUDIOCIF_CTRL_CLIENT_BITS_8		(TEGRA30_AUDIOCIF_BITS_8  << TEGRA30_AUDIOCIF_CTRL_CLIENT_BITS_SHIFT)
+#define TEGRA30_AUDIOCIF_CTRL_CLIENT_BITS_12		(TEGRA30_AUDIOCIF_BITS_12 << TEGRA30_AUDIOCIF_CTRL_CLIENT_BITS_SHIFT)
+#define TEGRA30_AUDIOCIF_CTRL_CLIENT_BITS_16		(TEGRA30_AUDIOCIF_BITS_16 << TEGRA30_AUDIOCIF_CTRL_CLIENT_BITS_SHIFT)
+#define TEGRA30_AUDIOCIF_CTRL_CLIENT_BITS_20		(TEGRA30_AUDIOCIF_BITS_20 << TEGRA30_AUDIOCIF_CTRL_CLIENT_BITS_SHIFT)
+#define TEGRA30_AUDIOCIF_CTRL_CLIENT_BITS_24		(TEGRA30_AUDIOCIF_BITS_24 << TEGRA30_AUDIOCIF_CTRL_CLIENT_BITS_SHIFT)
+#define TEGRA30_AUDIOCIF_CTRL_CLIENT_BITS_28		(TEGRA30_AUDIOCIF_BITS_28 << TEGRA30_AUDIOCIF_CTRL_CLIENT_BITS_SHIFT)
+#define TEGRA30_AUDIOCIF_CTRL_CLIENT_BITS_32		(TEGRA30_AUDIOCIF_BITS_32 << TEGRA30_AUDIOCIF_CTRL_CLIENT_BITS_SHIFT)
+
+#define TEGRA30_AUDIOCIF_EXPAND_ZERO			0
+#define TEGRA30_AUDIOCIF_EXPAND_ONE			1
+#define TEGRA30_AUDIOCIF_EXPAND_LFSR			2
+
+#define TEGRA30_AUDIOCIF_CTRL_EXPAND_SHIFT		6
+#define TEGRA30_AUDIOCIF_CTRL_EXPAND_MASK		(3                            << TEGRA30_AUDIOCIF_CTRL_EXPAND_SHIFT)
+#define TEGRA30_AUDIOCIF_CTRL_EXPAND_ZERO		(TEGRA30_AUDIOCIF_EXPAND_ZERO << TEGRA30_AUDIOCIF_CTRL_EXPAND_SHIFT)
+#define TEGRA30_AUDIOCIF_CTRL_EXPAND_ONE		(TEGRA30_AUDIOCIF_EXPAND_ONE  << TEGRA30_AUDIOCIF_CTRL_EXPAND_SHIFT)
+#define TEGRA30_AUDIOCIF_CTRL_EXPAND_LFSR		(TEGRA30_AUDIOCIF_EXPAND_LFSR << TEGRA30_AUDIOCIF_CTRL_EXPAND_SHIFT)
+
+#define TEGRA30_AUDIOCIF_STEREO_CONV_CH0		0
+#define TEGRA30_AUDIOCIF_STEREO_CONV_CH1		1
+#define TEGRA30_AUDIOCIF_STEREO_CONV_AVG		2
+
+#define TEGRA30_AUDIOCIF_CTRL_STEREO_CONV_SHIFT		4
+#define TEGRA30_AUDIOCIF_CTRL_STEREO_CONV_MASK		(3                                << TEGRA30_AUDIOCIF_CTRL_STEREO_CONV_SHIFT)
+#define TEGRA30_AUDIOCIF_CTRL_STEREO_CONV_CH0		(TEGRA30_AUDIOCIF_STEREO_CONV_CH0 << TEGRA30_AUDIOCIF_CTRL_STEREO_CONV_SHIFT)
+#define TEGRA30_AUDIOCIF_CTRL_STEREO_CONV_CH1		(TEGRA30_AUDIOCIF_STEREO_CONV_CH1 << TEGRA30_AUDIOCIF_CTRL_STEREO_CONV_SHIFT)
+#define TEGRA30_AUDIOCIF_CTRL_STEREO_CONV_AVG		(TEGRA30_AUDIOCIF_STEREO_CONV_AVG << TEGRA30_AUDIOCIF_CTRL_STEREO_CONV_SHIFT)
+
+#define TEGRA30_AUDIOCIF_CTRL_REPLICATE			3
+
+#define TEGRA30_AUDIOCIF_DIRECTION_TX			0
+#define TEGRA30_AUDIOCIF_DIRECTION_RX			1
+
+#define TEGRA30_AUDIOCIF_CTRL_DIRECTION_SHIFT		2
+#define TEGRA30_AUDIOCIF_CTRL_DIRECTION_MASK		(1                             << TEGRA30_AUDIOCIF_CTRL_DIRECTION_SHIFT)
+#define TEGRA30_AUDIOCIF_CTRL_DIRECTION_TX		(TEGRA30_AUDIOCIF_DIRECTION_TX << TEGRA30_AUDIOCIF_CTRL_DIRECTION_SHIFT)
+#define TEGRA30_AUDIOCIF_CTRL_DIRECTION_RX		(TEGRA30_AUDIOCIF_DIRECTION_RX << TEGRA30_AUDIOCIF_CTRL_DIRECTION_SHIFT)
+
+#define TEGRA30_AUDIOCIF_TRUNCATE_ROUND			0
+#define TEGRA30_AUDIOCIF_TRUNCATE_CHOP			1
+
+#define TEGRA30_AUDIOCIF_CTRL_TRUNCATE_SHIFT		1
+#define TEGRA30_AUDIOCIF_CTRL_TRUNCATE_MASK		(1                               << TEGRA30_AUDIOCIF_CTRL_TRUNCATE_SHIFT)
+#define TEGRA30_AUDIOCIF_CTRL_TRUNCATE_ROUND		(TEGRA30_AUDIOCIF_TRUNCATE_ROUND << TEGRA30_AUDIOCIF_CTRL_TRUNCATE_SHIFT)
+#define TEGRA30_AUDIOCIF_CTRL_TRUNCATE_CHOP		(TEGRA30_AUDIOCIF_TRUNCATE_CHOP  << TEGRA30_AUDIOCIF_CTRL_TRUNCATE_SHIFT)
+
+#define TEGRA30_AUDIOCIF_MONO_CONV_ZERO			0
+#define TEGRA30_AUDIOCIF_MONO_CONV_COPY			1
+
+#define TEGRA30_AUDIOCIF_CTRL_MONO_CONV_SHIFT		0
+#define TEGRA30_AUDIOCIF_CTRL_MONO_CONV_MASK		(1                               << TEGRA30_AUDIOCIF_CTRL_MONO_CONV_SHIFT)
+#define TEGRA30_AUDIOCIF_CTRL_MONO_CONV_ZERO		(TEGRA30_AUDIOCIF_MONO_CONV_ZERO << TEGRA30_AUDIOCIF_CTRL_MONO_CONV_SHIFT)
+#define TEGRA30_AUDIOCIF_CTRL_MONO_CONV_COPY		(TEGRA30_AUDIOCIF_MONO_CONV_COPY << TEGRA30_AUDIOCIF_CTRL_MONO_CONV_SHIFT)
+
+/* Registers within TEGRA30_AUDIO_CLUSTER_BASE */
+
+/* TEGRA30_AHUB_CHANNEL_CTRL */
+
+#define TEGRA30_AHUB_CHANNEL_CTRL			0x0
+#define TEGRA30_AHUB_CHANNEL_CTRL_STRIDE		0x20
+#define TEGRA30_AHUB_CHANNEL_CTRL_COUNT			4
+#define TEGRA30_AHUB_CHANNEL_CTRL_TX_EN			(1 << 31)
+#define TEGRA30_AHUB_CHANNEL_CTRL_RX_EN			(1 << 30)
+#define TEGRA30_AHUB_CHANNEL_CTRL_LOOPBACK		(1 << 29)
+
+#define TEGRA30_AHUB_CHANNEL_CTRL_TX_THRESHOLD_SHIFT	16
+#define TEGRA30_AHUB_CHANNEL_CTRL_TX_THRESHOLD_MASK_US	0xff
+#define TEGRA30_AHUB_CHANNEL_CTRL_TX_THRESHOLD_MASK	(TEGRA30_AHUB_CHANNEL_CTRL_TX_THRESHOLD_MASK_US << TEGRA30_AHUB_CHANNEL_CTRL_TX_THRESHOLD_SHIFT)
+
+#define TEGRA30_AHUB_CHANNEL_CTRL_RX_THRESHOLD_SHIFT	8
+#define TEGRA30_AHUB_CHANNEL_CTRL_RX_THRESHOLD_MASK_US	0xff
+#define TEGRA30_AHUB_CHANNEL_CTRL_RX_THRESHOLD_MASK	(TEGRA30_AHUB_CHANNEL_CTRL_RX_THRESHOLD_MASK_US << TEGRA30_AHUB_CHANNEL_CTRL_RX_THRESHOLD_SHIFT)
+
+#define TEGRA30_AHUB_CHANNEL_CTRL_TX_PACK_EN		(1 << 6)
+
+#define TEGRA30_PACK_8_4				2
+#define TEGRA30_PACK_16					3
+
+#define TEGRA30_AHUB_CHANNEL_CTRL_TX_PACK_SHIFT		4
+#define TEGRA30_AHUB_CHANNEL_CTRL_TX_PACK_MASK_US	3
+#define TEGRA30_AHUB_CHANNEL_CTRL_TX_PACK_MASK		(TEGRA30_AHUB_CHANNEL_CTRL_TX_PACK_MASK_US << TEGRA30_AHUB_CHANNEL_CTRL_TX_PACK_SHIFT)
+#define TEGRA30_AHUB_CHANNEL_CTRL_TX_PACK_8_4		(TEGRA30_PACK_8_4                          << TEGRA30_AHUB_CHANNEL_CTRL_TX_PACK_SHIFT)
+#define TEGRA30_AHUB_CHANNEL_CTRL_TX_PACK_16		(TEGRA30_PACK_16                           << TEGRA30_AHUB_CHANNEL_CTRL_TX_PACK_SHIFT)
+
+#define TEGRA30_AHUB_CHANNEL_CTRL_RX_PACK_EN		(1 << 2)
+
+#define TEGRA30_AHUB_CHANNEL_CTRL_RX_PACK_SHIFT		0
+#define TEGRA30_AHUB_CHANNEL_CTRL_RX_PACK_MASK_US	3
+#define TEGRA30_AHUB_CHANNEL_CTRL_RX_PACK_MASK		(TEGRA30_AHUB_CHANNEL_CTRL_RX_PACK_MASK_US << TEGRA30_AHUB_CHANNEL_CTRL_RX_PACK_SHIFT)
+#define TEGRA30_AHUB_CHANNEL_CTRL_RX_PACK_8_4		(TEGRA30_PACK_8_4                          << TEGRA30_AHUB_CHANNEL_CTRL_RX_PACK_SHIFT)
+#define TEGRA30_AHUB_CHANNEL_CTRL_RX_PACK_16		(TEGRA30_PACK_16                           << TEGRA30_AHUB_CHANNEL_CTRL_RX_PACK_SHIFT)
+
+/* TEGRA30_AHUB_CHANNEL_CLEAR */
+
+#define TEGRA30_AHUB_CHANNEL_CLEAR			0x4
+#define TEGRA30_AHUB_CHANNEL_CLEAR_STRIDE		0x20
+#define TEGRA30_AHUB_CHANNEL_CLEAR_COUNT		4
+#define TEGRA30_AHUB_CHANNEL_CLEAR_TX_SOFT_RESET	(1 << 31)
+#define TEGRA30_AHUB_CHANNEL_CLEAR_RX_SOFT_RESET	(1 << 30)
+
+/* TEGRA30_AHUB_CHANNEL_STATUS */
+
+#define TEGRA30_AHUB_CHANNEL_STATUS			0x8
+#define TEGRA30_AHUB_CHANNEL_STATUS_STRIDE		0x20
+#define TEGRA30_AHUB_CHANNEL_STATUS_COUNT		4
+#define TEGRA30_AHUB_CHANNEL_STATUS_TX_FREE_SHIFT	24
+#define TEGRA30_AHUB_CHANNEL_STATUS_TX_FREE_MASK_US	0xff
+#define TEGRA30_AHUB_CHANNEL_STATUS_TX_FREE_MASK	(TEGRA30_AHUB_CHANNEL_STATUS_TX_FREE_MASK_US << TEGRA30_AHUB_CHANNEL_STATUS_TX_FREE_SHIFT)
+#define TEGRA30_AHUB_CHANNEL_STATUS_RX_FREE_SHIFT	16
+#define TEGRA30_AHUB_CHANNEL_STATUS_RX_FREE_MASK_US	0xff
+#define TEGRA30_AHUB_CHANNEL_STATUS_RX_FREE_MASK	(TEGRA30_AHUB_CHANNEL_STATUS_RX_FREE_MASK_US << TEGRA30_AHUB_CHANNEL_STATUS_RX_FREE_SHIFT)
+#define TEGRA30_AHUB_CHANNEL_STATUS_TX_TRIG		(1 << 1)
+#define TEGRA30_AHUB_CHANNEL_STATUS_RX_TRIG		(1 << 0)
+
+/* TEGRA30_AHUB_CHANNEL_TXFIFO */
+
+#define TEGRA30_AHUB_CHANNEL_TXFIFO			0xc
+#define TEGRA30_AHUB_CHANNEL_TXFIFO_STRIDE		0x20
+#define TEGRA30_AHUB_CHANNEL_TXFIFO_COUNT		4
+
+/* TEGRA30_AHUB_CHANNEL_RXFIFO */
+
+#define TEGRA30_AHUB_CHANNEL_RXFIFO			0x10
+#define TEGRA30_AHUB_CHANNEL_RXFIFO_STRIDE		0x20
+#define TEGRA30_AHUB_CHANNEL_RXFIFO_COUNT		4
+
+/* TEGRA30_AHUB_CIF_TX_CTRL */
+
+#define TEGRA30_AHUB_CIF_TX_CTRL			0x14
+#define TEGRA30_AHUB_CIF_TX_CTRL_STRIDE			0x20
+#define TEGRA30_AHUB_CIF_TX_CTRL_COUNT			4
+/* Uses field from TEGRA30_AUDIOCIF_CTRL_* */
+
+/* TEGRA30_AHUB_CIF_RX_CTRL */
+
+#define TEGRA30_AHUB_CIF_RX_CTRL			0x18
+#define TEGRA30_AHUB_CIF_RX_CTRL_STRIDE			0x20
+#define TEGRA30_AHUB_CIF_RX_CTRL_COUNT			4
+/* Uses field from TEGRA30_AUDIOCIF_CTRL_* */
+
+/* TEGRA30_AHUB_CONFIG_LINK_CTRL */
+
+#define TEGRA30_AHUB_CONFIG_LINK_CTRL					0x80
+#define TEGRA30_AHUB_CONFIG_LINK_CTRL_MASTER_FIFO_FULL_CNT_SHIFT	28
+#define TEGRA30_AHUB_CONFIG_LINK_CTRL_MASTER_FIFO_FULL_CNT_MASK_US	0xf
+#define TEGRA30_AHUB_CONFIG_LINK_CTRL_MASTER_FIFO_FULL_CNT_MASK		(TEGRA30_AHUB_CONFIG_LINK_CTRL_MASTER_FIFO_FULL_CNT_MASK_US << TEGRA30_AHUB_CONFIG_LINK_CTRL_MASTER_FIFO_FULL_CNT_SHIFT)
+#define TEGRA30_AHUB_CONFIG_LINK_CTRL_TIMEOUT_CNT_SHIFT			16
+#define TEGRA30_AHUB_CONFIG_LINK_CTRL_TIMEOUT_CNT_MASK_US		0xfff
+#define TEGRA30_AHUB_CONFIG_LINK_CTRL_TIMEOUT_CNT_MASK			(TEGRA30_AHUB_CONFIG_LINK_CTRL_TIMEOUT_CNT_MASK_US << TEGRA30_AHUB_CONFIG_LINK_CTRL_TIMEOUT_CNT_SHIFT)
+#define TEGRA30_AHUB_CONFIG_LINK_CTRL_IDLE_CNT_SHIFT			4
+#define TEGRA30_AHUB_CONFIG_LINK_CTRL_IDLE_CNT_MASK_US			0xfff
+#define TEGRA30_AHUB_CONFIG_LINK_CTRL_IDLE_CNT_MASK			(TEGRA30_AHUB_CONFIG_LINK_CTRL_IDLE_CNT_MASK_US << TEGRA30_AHUB_CONFIG_LINK_CTRL_IDLE_CNT_SHIFT)
+#define TEGRA30_AHUB_CONFIG_LINK_CTRL_CG_EN				(1 << 2)
+#define TEGRA30_AHUB_CONFIG_LINK_CTRL_CLEAR_TIMEOUT_CNTR		(1 << 1)
+#define TEGRA30_AHUB_CONFIG_LINK_CTRL_SOFT_RESET			(1 << 0)
+
+/* TEGRA30_AHUB_MISC_CTRL */
+
+#define TEGRA30_AHUB_MISC_CTRL				0x84
+#define TEGRA30_AHUB_MISC_CTRL_AUDIO_ACTIVE		(1 << 31)
+#define TEGRA30_AHUB_MISC_CTRL_AUDIO_CG_EN		(1 << 8)
+#define TEGRA30_AHUB_MISC_CTRL_AUDIO_OBS_SEL_SHIFT	0
+#define TEGRA30_AHUB_MISC_CTRL_AUDIO_OBS_SEL_MASK	(0x1f << TEGRA30_AHUB_MISC_CTRL_AUDIO_OBS_SEL_SHIFT)
+
+/* TEGRA30_AHUB_APBDMA_LIVE_STATUS */
+
+#define TEGRA30_AHUB_APBDMA_LIVE_STATUS				0x88
+#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH3_RX_CIF_FIFO_FULL	(1 << 31)
+#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH3_TX_CIF_FIFO_FULL	(1 << 30)
+#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH2_RX_CIF_FIFO_FULL	(1 << 29)
+#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH2_TX_CIF_FIFO_FULL	(1 << 28)
+#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH1_RX_CIF_FIFO_FULL	(1 << 27)
+#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH1_TX_CIF_FIFO_FULL	(1 << 26)
+#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH0_RX_CIF_FIFO_FULL	(1 << 25)
+#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH0_TX_CIF_FIFO_FULL	(1 << 24)
+#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH3_RX_CIF_FIFO_EMPTY	(1 << 23)
+#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH3_TX_CIF_FIFO_EMPTY	(1 << 22)
+#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH2_RX_CIF_FIFO_EMPTY	(1 << 21)
+#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH2_TX_CIF_FIFO_EMPTY	(1 << 20)
+#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH1_RX_CIF_FIFO_EMPTY	(1 << 19)
+#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH1_TX_CIF_FIFO_EMPTY	(1 << 18)
+#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH0_RX_CIF_FIFO_EMPTY	(1 << 17)
+#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH0_TX_CIF_FIFO_EMPTY	(1 << 16)
+#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH3_RX_DMA_FIFO_FULL	(1 << 15)
+#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH3_TX_DMA_FIFO_FULL	(1 << 14)
+#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH2_RX_DMA_FIFO_FULL	(1 << 13)
+#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH2_TX_DMA_FIFO_FULL	(1 << 12)
+#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH1_RX_DMA_FIFO_FULL	(1 << 11)
+#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH1_TX_DMA_FIFO_FULL	(1 << 10)
+#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH0_RX_DMA_FIFO_FULL	(1 << 9)
+#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH0_TX_DMA_FIFO_FULL	(1 << 8)
+#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH3_RX_DMA_FIFO_EMPTY	(1 << 7)
+#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH3_TX_DMA_FIFO_EMPTY	(1 << 6)
+#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH2_RX_DMA_FIFO_EMPTY	(1 << 5)
+#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH2_TX_DMA_FIFO_EMPTY	(1 << 4)
+#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH1_RX_DMA_FIFO_EMPTY	(1 << 3)
+#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH1_TX_DMA_FIFO_EMPTY	(1 << 2)
+#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH0_RX_DMA_FIFO_EMPTY	(1 << 1)
+#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH0_TX_DMA_FIFO_EMPTY	(1 << 0)
+
+/* TEGRA30_AHUB_I2S_LIVE_STATUS */
+
+#define TEGRA30_AHUB_I2S_LIVE_STATUS				0x8c
+#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S4_RX_FIFO_FULL		(1 << 29)
+#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S4_TX_FIFO_FULL		(1 << 28)
+#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S3_RX_FIFO_FULL		(1 << 27)
+#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S3_TX_FIFO_FULL		(1 << 26)
+#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S2_RX_FIFO_FULL		(1 << 25)
+#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S2_TX_FIFO_FULL		(1 << 24)
+#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S1_RX_FIFO_FULL		(1 << 23)
+#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S1_TX_FIFO_FULL		(1 << 22)
+#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S0_RX_FIFO_FULL		(1 << 21)
+#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S0_TX_FIFO_FULL		(1 << 20)
+#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S4_RX_FIFO_ENABLED	(1 << 19)
+#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S4_TX_FIFO_ENABLED	(1 << 18)
+#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S3_RX_FIFO_ENABLED	(1 << 17)
+#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S3_TX_FIFO_ENABLED	(1 << 16)
+#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S2_RX_FIFO_ENABLED	(1 << 15)
+#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S2_TX_FIFO_ENABLED	(1 << 14)
+#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S1_RX_FIFO_ENABLED	(1 << 13)
+#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S1_TX_FIFO_ENABLED	(1 << 12)
+#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S0_RX_FIFO_ENABLED	(1 << 11)
+#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S0_TX_FIFO_ENABLED	(1 << 10)
+#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S4_RX_FIFO_EMPTY		(1 << 9)
+#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S4_TX_FIFO_EMPTY		(1 << 8)
+#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S3_RX_FIFO_EMPTY		(1 << 7)
+#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S3_TX_FIFO_EMPTY		(1 << 6)
+#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S2_RX_FIFO_EMPTY		(1 << 5)
+#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S2_TX_FIFO_EMPTY		(1 << 4)
+#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S1_RX_FIFO_EMPTY		(1 << 3)
+#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S1_TX_FIFO_EMPTY		(1 << 2)
+#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S0_RX_FIFO_EMPTY		(1 << 1)
+#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S0_TX_FIFO_EMPTY		(1 << 0)
+
+/* TEGRA30_AHUB_DAM0_LIVE_STATUS */
+
+#define TEGRA30_AHUB_DAM_LIVE_STATUS				0x90
+#define TEGRA30_AHUB_DAM_LIVE_STATUS_STRIDE			0x8
+#define TEGRA30_AHUB_DAM_LIVE_STATUS_COUNT			3
+#define TEGRA30_AHUB_DAM_LIVE_STATUS_TX_ENABLED			(1 << 26)
+#define TEGRA30_AHUB_DAM_LIVE_STATUS_RX1_ENABLED		(1 << 25)
+#define TEGRA30_AHUB_DAM_LIVE_STATUS_RX0_ENABLED		(1 << 24)
+#define TEGRA30_AHUB_DAM_LIVE_STATUS_TXFIFO_FULL		(1 << 15)
+#define TEGRA30_AHUB_DAM_LIVE_STATUS_RX1FIFO_FULL		(1 << 9)
+#define TEGRA30_AHUB_DAM_LIVE_STATUS_RX0FIFO_FULL		(1 << 8)
+#define TEGRA30_AHUB_DAM_LIVE_STATUS_TXFIFO_EMPTY		(1 << 7)
+#define TEGRA30_AHUB_DAM_LIVE_STATUS_RX1FIFO_EMPTY		(1 << 1)
+#define TEGRA30_AHUB_DAM_LIVE_STATUS_RX0FIFO_EMPTY		(1 << 0)
+
+/* TEGRA30_AHUB_SPDIF_LIVE_STATUS */
+
+#define TEGRA30_AHUB_SPDIF_LIVE_STATUS				0xa8
+#define TEGRA30_AHUB_SPDIF_LIVE_STATUS_USER_TX_ENABLED		(1 << 11)
+#define TEGRA30_AHUB_SPDIF_LIVE_STATUS_USER_RX_ENABLED		(1 << 10)
+#define TEGRA30_AHUB_SPDIF_LIVE_STATUS_DATA_TX_ENABLED		(1 << 9)
+#define TEGRA30_AHUB_SPDIF_LIVE_STATUS_DATA_RX_ENABLED		(1 << 8)
+#define TEGRA30_AHUB_SPDIF_LIVE_STATUS_USER_TXFIFO_FULL		(1 << 7)
+#define TEGRA30_AHUB_SPDIF_LIVE_STATUS_USER_RXFIFO_FULL		(1 << 6)
+#define TEGRA30_AHUB_SPDIF_LIVE_STATUS_DATA_TXFIFO_FULL		(1 << 5)
+#define TEGRA30_AHUB_SPDIF_LIVE_STATUS_DATA_RXFIFO_FULL		(1 << 4)
+#define TEGRA30_AHUB_SPDIF_LIVE_STATUS_USER_TXFIFO_EMPTY	(1 << 3)
+#define TEGRA30_AHUB_SPDIF_LIVE_STATUS_USER_RXFIFO_EMPTY	(1 << 2)
+#define TEGRA30_AHUB_SPDIF_LIVE_STATUS_DATA_TXFIFO_EMPTY	(1 << 1)
+#define TEGRA30_AHUB_SPDIF_LIVE_STATUS_DATA_RXFIFO_EMPTY	(1 << 0)
+
+/* TEGRA30_AHUB_I2S_INT_MASK */
+
+#define TEGRA30_AHUB_I2S_INT_MASK				0xb0
+
+/* TEGRA30_AHUB_DAM_INT_MASK */
+
+#define TEGRA30_AHUB_DAM_INT_MASK				0xb4
+
+/* TEGRA30_AHUB_SPDIF_INT_MASK */
+
+#define TEGRA30_AHUB_SPDIF_INT_MASK				0xbc
+
+/* TEGRA30_AHUB_APBIF_INT_MASK */
+
+#define TEGRA30_AHUB_APBIF_INT_MASK				0xc0
+
+/* TEGRA30_AHUB_I2S_INT_STATUS */
+
+#define TEGRA30_AHUB_I2S_INT_STATUS				0xc8
+
+/* TEGRA30_AHUB_DAM_INT_STATUS */
+
+#define TEGRA30_AHUB_DAM_INT_STATUS				0xcc
+
+/* TEGRA30_AHUB_SPDIF_INT_STATUS */
+
+#define TEGRA30_AHUB_SPDIF_INT_STATUS				0xd4
+
+/* TEGRA30_AHUB_APBIF_INT_STATUS */
+
+#define TEGRA30_AHUB_APBIF_INT_STATUS				0xd8
+
+/* TEGRA30_AHUB_I2S_INT_SOURCE */
+
+#define TEGRA30_AHUB_I2S_INT_SOURCE				0xe0
+
+/* TEGRA30_AHUB_DAM_INT_SOURCE */
+
+#define TEGRA30_AHUB_DAM_INT_SOURCE				0xe4
+
+/* TEGRA30_AHUB_SPDIF_INT_SOURCE */
+
+#define TEGRA30_AHUB_SPDIF_INT_SOURCE				0xec
+
+/* TEGRA30_AHUB_APBIF_INT_SOURCE */
+
+#define TEGRA30_AHUB_APBIF_INT_SOURCE				0xf0
+
+/* TEGRA30_AHUB_I2S_INT_SET */
+
+#define TEGRA30_AHUB_I2S_INT_SET				0xf8
+
+/* TEGRA30_AHUB_DAM_INT_SET */
+
+#define TEGRA30_AHUB_DAM_INT_SET				0xfc
+
+/* TEGRA30_AHUB_SPDIF_INT_SET */
+
+#define TEGRA30_AHUB_SPDIF_INT_SET				0x100
+
+/* TEGRA30_AHUB_APBIF_INT_SET */
+
+#define TEGRA30_AHUB_APBIF_INT_SET				0x104
+
+/* Registers within TEGRA30_AHUB_BASE */
+
+#define TEGRA30_AHUB_AUDIO_RX					0x0
+#define TEGRA30_AHUB_AUDIO_RX_STRIDE				0x4
+#define TEGRA30_AHUB_AUDIO_RX_COUNT				17
+/* This register repeats once for each entry in enum tegra30_ahub_rxcif */
+/* The fields in this register are 1 bit per entry in tegra30_ahub_txcif */
+
+/*
+ * Terminology:
+ * AHUB: Audio Hub; a cross-bar switch between the audio devices: DMA FIFOs,
+ *       I2S controllers, SPDIF controllers, and DAMs.
+ * XBAR: The core cross-bar component of the AHUB.
+ * CIF:  Client Interface; the HW module connecting an audio device to the
+ *       XBAR.
+ * DAM:  Digital Audio Mixer: A HW module that mixes multiple audio streams,
+ *       possibly including sample-rate conversion.
+ *
+ * Each TX CIF transmits data into the XBAR. Each RX CIF can receive audio
+ * transmitted by a particular TX CIF.
+ *
+ * This driver is currently very simplistic; many HW features are not
+ * exposed; DAMs are not supported, only 16-bit stereo audio is supported,
+ * etc.
+ */
+
+enum tegra30_ahub_txcif {
+	TEGRA30_AHUB_TXCIF_APBIF_TX0,
+	TEGRA30_AHUB_TXCIF_APBIF_TX1,
+	TEGRA30_AHUB_TXCIF_APBIF_TX2,
+	TEGRA30_AHUB_TXCIF_APBIF_TX3,
+	TEGRA30_AHUB_TXCIF_I2S0_TX0,
+	TEGRA30_AHUB_TXCIF_I2S1_TX0,
+	TEGRA30_AHUB_TXCIF_I2S2_TX0,
+	TEGRA30_AHUB_TXCIF_I2S3_TX0,
+	TEGRA30_AHUB_TXCIF_I2S4_TX0,
+	TEGRA30_AHUB_TXCIF_DAM0_TX0,
+	TEGRA30_AHUB_TXCIF_DAM1_TX0,
+	TEGRA30_AHUB_TXCIF_DAM2_TX0,
+	TEGRA30_AHUB_TXCIF_SPDIF_TX0,
+	TEGRA30_AHUB_TXCIF_SPDIF_TX1,
+};
+
+enum tegra30_ahub_rxcif {
+	TEGRA30_AHUB_RXCIF_APBIF_RX0,
+	TEGRA30_AHUB_RXCIF_APBIF_RX1,
+	TEGRA30_AHUB_RXcIF_APBIF_RX2,
+	TEGRA30_AHUB_RXCIF_APBIF_RX3,
+	TEGRA30_AHUB_RXCIF_I2S0_RX0,
+	TEGRA30_AHUB_RXCIF_I2S1_RX0,
+	TEGRA30_AHUB_RXCIF_I2S2_RX0,
+	TEGRA30_AHUB_RXCIF_I2S3_RX0,
+	TEGRA30_AHUB_RXCIF_I2S4_RX0,
+	TEGRA30_AHUB_RXCIF_DAM0_RX0,
+	TEGRA30_AHUB_RXCIF_DAM0_RX1,
+	TEGRA30_AHUB_RXCIF_DAM1_RX0,
+	TEGRA30_AHUB_RXCIF_DAM2_RX1,
+	TEGRA30_AHUB_RXCIF_DAM3_RX0,
+	TEGRA30_AHUB_RXCIF_DAM3_RX1,
+	TEGRA30_AHUB_RXCIF_SPDIF_RX0,
+	TEGRA30_AHUB_RXCIF_SPDIF_RX1,
+};
+
+extern int tegra30_ahub_allocate_rx_fifo(enum tegra30_ahub_rxcif *rxcif,
+					 unsigned long *fiforeg,
+					 unsigned long *reqsel);
+extern int tegra30_ahub_enable_rx_fifo(enum tegra30_ahub_rxcif rxcif);
+extern int tegra30_ahub_disable_rx_fifo(enum tegra30_ahub_rxcif rxcif);
+extern int tegra30_ahub_free_rx_fifo(enum tegra30_ahub_rxcif rxcif);
+
+extern int tegra30_ahub_allocate_tx_fifo(enum tegra30_ahub_txcif *txcif,
+					 unsigned long *fiforeg,
+					 unsigned long *reqsel);
+extern int tegra30_ahub_enable_tx_fifo(enum tegra30_ahub_txcif txcif);
+extern int tegra30_ahub_disable_tx_fifo(enum tegra30_ahub_txcif txcif);
+extern int tegra30_ahub_free_tx_fifo(enum tegra30_ahub_txcif txcif);
+
+extern int tegra30_ahub_set_rx_cif_source(enum tegra30_ahub_rxcif rxcif,
+					  enum tegra30_ahub_txcif txcif);
+extern int tegra30_ahub_unset_rx_cif_source(enum tegra30_ahub_rxcif rxcif);
+
+struct tegra30_ahub {
+	struct device *dev;
+	struct clk *clk_d_audio;
+	struct clk *clk_apbif;
+	int dma_sel;
+	resource_size_t apbif_addr;
+	struct regmap *regmap_apbif;
+	struct regmap *regmap_ahub;
+	DECLARE_BITMAP(rx_usage, TEGRA30_AHUB_CHANNEL_CTRL_COUNT);
+	DECLARE_BITMAP(tx_usage, TEGRA30_AHUB_CHANNEL_CTRL_COUNT);
+};
+
+#endif
diff --git a/sound/soc/tegra/tegra30_i2s.c b/sound/soc/tegra/tegra30_i2s.c
new file mode 100644
index 000000000000..8596032985dc
--- /dev/null
+++ b/sound/soc/tegra/tegra30_i2s.c
@@ -0,0 +1,536 @@
+/*
+ * tegra30_i2s.c - Tegra30 I2S driver
+ *
+ * Author: Stephen Warren <swarren@nvidia.com>
+ * Copyright (c) 2010-2012, NVIDIA CORPORATION.  All rights reserved.
+ *
+ * Based on code copyright/by:
+ *
+ * Copyright (c) 2009-2010, NVIDIA Corporation.
+ * Scott Peterson <speterson@nvidia.com>
+ *
+ * Copyright (C) 2010 Google, Inc.
+ * Iliyan Malchev <malchev@google.com>
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms and conditions of the GNU General Public License,
+ * version 2, as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope 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, see <http://www.gnu.org/licenses/>.
+ */
+
+#include <linux/clk.h>
+#include <linux/device.h>
+#include <linux/io.h>
+#include <linux/module.h>
+#include <linux/of.h>
+#include <linux/platform_device.h>
+#include <linux/pm_runtime.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 "tegra30_ahub.h"
+#include "tegra30_i2s.h"
+
+#define DRV_NAME "tegra30-i2s"
+
+static inline void tegra30_i2s_write(struct tegra30_i2s *i2s, u32 reg, u32 val)
+{
+	regmap_write(i2s->regmap, reg, val);
+}
+
+static inline u32 tegra30_i2s_read(struct tegra30_i2s *i2s, u32 reg)
+{
+	u32 val;
+	regmap_read(i2s->regmap, reg, &val);
+	return val;
+}
+
+static int tegra30_i2s_runtime_suspend(struct device *dev)
+{
+	struct tegra30_i2s *i2s = dev_get_drvdata(dev);
+
+	regcache_cache_only(i2s->regmap, true);
+
+	clk_disable(i2s->clk_i2s);
+
+	return 0;
+}
+
+static int tegra30_i2s_runtime_resume(struct device *dev)
+{
+	struct tegra30_i2s *i2s = dev_get_drvdata(dev);
+	int ret;
+
+	ret = clk_enable(i2s->clk_i2s);
+	if (ret) {
+		dev_err(dev, "clk_enable failed: %d\n", ret);
+		return ret;
+	}
+
+	regcache_cache_only(i2s->regmap, false);
+
+	return 0;
+}
+
+int tegra30_i2s_startup(struct snd_pcm_substream *substream,
+			struct snd_soc_dai *dai)
+{
+	struct tegra30_i2s *i2s = snd_soc_dai_get_drvdata(dai);
+	int ret;
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		ret = tegra30_ahub_allocate_tx_fifo(&i2s->playback_fifo_cif,
+					&i2s->playback_dma_data.addr,
+					&i2s->playback_dma_data.req_sel);
+		i2s->playback_dma_data.wrap = 4;
+		i2s->playback_dma_data.width = 32;
+		tegra30_ahub_set_rx_cif_source(i2s->playback_i2s_cif,
+					       i2s->playback_fifo_cif);
+	} else {
+		ret = tegra30_ahub_allocate_rx_fifo(&i2s->capture_fifo_cif,
+					&i2s->capture_dma_data.addr,
+					&i2s->capture_dma_data.req_sel);
+		i2s->capture_dma_data.wrap = 4;
+		i2s->capture_dma_data.width = 32;
+		tegra30_ahub_set_rx_cif_source(i2s->capture_fifo_cif,
+					       i2s->capture_i2s_cif);
+	}
+
+	return ret;
+}
+
+void tegra30_i2s_shutdown(struct snd_pcm_substream *substream,
+			struct snd_soc_dai *dai)
+{
+	struct tegra30_i2s *i2s = snd_soc_dai_get_drvdata(dai);
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		tegra30_ahub_unset_rx_cif_source(i2s->playback_i2s_cif);
+		tegra30_ahub_free_tx_fifo(i2s->playback_fifo_cif);
+	} else {
+		tegra30_ahub_unset_rx_cif_source(i2s->capture_fifo_cif);
+		tegra30_ahub_free_rx_fifo(i2s->capture_fifo_cif);
+	}
+}
+
+static int tegra30_i2s_set_fmt(struct snd_soc_dai *dai,
+				unsigned int fmt)
+{
+	struct tegra30_i2s *i2s = snd_soc_dai_get_drvdata(dai);
+
+	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+	case SND_SOC_DAIFMT_NB_NF:
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	i2s->reg_ctrl &= ~TEGRA30_I2S_CTRL_MASTER_ENABLE;
+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBS_CFS:
+		i2s->reg_ctrl |= TEGRA30_I2S_CTRL_MASTER_ENABLE;
+		break;
+	case SND_SOC_DAIFMT_CBM_CFM:
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	i2s->reg_ctrl &= ~(TEGRA30_I2S_CTRL_FRAME_FORMAT_MASK |
+			   TEGRA30_I2S_CTRL_LRCK_MASK);
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_DSP_A:
+		i2s->reg_ctrl |= TEGRA30_I2S_CTRL_FRAME_FORMAT_FSYNC;
+		i2s->reg_ctrl |= TEGRA30_I2S_CTRL_LRCK_L_LOW;
+		break;
+	case SND_SOC_DAIFMT_DSP_B:
+		i2s->reg_ctrl |= TEGRA30_I2S_CTRL_FRAME_FORMAT_FSYNC;
+		i2s->reg_ctrl |= TEGRA30_I2S_CTRL_LRCK_R_LOW;
+		break;
+	case SND_SOC_DAIFMT_I2S:
+		i2s->reg_ctrl |= TEGRA30_I2S_CTRL_FRAME_FORMAT_LRCK;
+		i2s->reg_ctrl |= TEGRA30_I2S_CTRL_LRCK_L_LOW;
+		break;
+	case SND_SOC_DAIFMT_RIGHT_J:
+		i2s->reg_ctrl |= TEGRA30_I2S_CTRL_FRAME_FORMAT_LRCK;
+		i2s->reg_ctrl |= TEGRA30_I2S_CTRL_LRCK_L_LOW;
+		break;
+	case SND_SOC_DAIFMT_LEFT_J:
+		i2s->reg_ctrl |= TEGRA30_I2S_CTRL_FRAME_FORMAT_LRCK;
+		i2s->reg_ctrl |= TEGRA30_I2S_CTRL_LRCK_L_LOW;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	return 0;
+}
+
+static int tegra30_i2s_hw_params(struct snd_pcm_substream *substream,
+				 struct snd_pcm_hw_params *params,
+				 struct snd_soc_dai *dai)
+{
+	struct device *dev = substream->pcm->card->dev;
+	struct tegra30_i2s *i2s = snd_soc_dai_get_drvdata(dai);
+	u32 val;
+	int ret, sample_size, srate, i2sclock, bitcnt;
+
+	if (params_channels(params) != 2)
+		return -EINVAL;
+
+	i2s->reg_ctrl &= ~TEGRA30_I2S_CTRL_BIT_SIZE_MASK;
+	switch (params_format(params)) {
+	case SNDRV_PCM_FORMAT_S16_LE:
+		i2s->reg_ctrl |= TEGRA30_I2S_CTRL_BIT_SIZE_16;
+		sample_size = 16;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	srate = params_rate(params);
+
+	/* Final "* 2" required by Tegra hardware */
+	i2sclock = srate * params_channels(params) * sample_size * 2;
+
+	bitcnt = (i2sclock / (2 * srate)) - 1;
+	if (bitcnt < 0 || bitcnt > TEGRA30_I2S_TIMING_CHANNEL_BIT_COUNT_MASK_US)
+		return -EINVAL;
+
+	ret = clk_set_rate(i2s->clk_i2s, i2sclock);
+	if (ret) {
+		dev_err(dev, "Can't set I2S clock rate: %d\n", ret);
+		return ret;
+	}
+
+	val = bitcnt << TEGRA30_I2S_TIMING_CHANNEL_BIT_COUNT_SHIFT;
+
+	if (i2sclock % (2 * srate))
+		val |= TEGRA30_I2S_TIMING_NON_SYM_ENABLE;
+
+	tegra30_i2s_write(i2s, TEGRA30_I2S_TIMING, val);
+
+	val = (0 << TEGRA30_AUDIOCIF_CTRL_FIFO_THRESHOLD_SHIFT) |
+	      (1 << TEGRA30_AUDIOCIF_CTRL_AUDIO_CHANNELS_SHIFT) |
+	      (1 << TEGRA30_AUDIOCIF_CTRL_CLIENT_CHANNELS_SHIFT) |
+	      TEGRA30_AUDIOCIF_CTRL_AUDIO_BITS_16 |
+	      TEGRA30_AUDIOCIF_CTRL_CLIENT_BITS_16;
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		val |= TEGRA30_AUDIOCIF_CTRL_DIRECTION_RX;
+		tegra30_i2s_write(i2s, TEGRA30_I2S_CIF_RX_CTRL, val);
+	} else {
+		val |= TEGRA30_AUDIOCIF_CTRL_DIRECTION_TX;
+		tegra30_i2s_write(i2s, TEGRA30_I2S_CIF_TX_CTRL, val);
+	}
+
+	val = (1 << TEGRA30_I2S_OFFSET_RX_DATA_OFFSET_SHIFT) |
+	      (1 << TEGRA30_I2S_OFFSET_TX_DATA_OFFSET_SHIFT);
+	tegra30_i2s_write(i2s, TEGRA30_I2S_OFFSET, val);
+
+	return 0;
+}
+
+static void tegra30_i2s_start_playback(struct tegra30_i2s *i2s)
+{
+	tegra30_ahub_enable_tx_fifo(i2s->playback_fifo_cif);
+	i2s->reg_ctrl |= TEGRA30_I2S_CTRL_XFER_EN_TX;
+	tegra30_i2s_write(i2s, TEGRA30_I2S_CTRL, i2s->reg_ctrl);
+}
+
+static void tegra30_i2s_stop_playback(struct tegra30_i2s *i2s)
+{
+	tegra30_ahub_disable_tx_fifo(i2s->playback_fifo_cif);
+	i2s->reg_ctrl &= ~TEGRA30_I2S_CTRL_XFER_EN_TX;
+	tegra30_i2s_write(i2s, TEGRA30_I2S_CTRL, i2s->reg_ctrl);
+}
+
+static void tegra30_i2s_start_capture(struct tegra30_i2s *i2s)
+{
+	tegra30_ahub_enable_rx_fifo(i2s->capture_fifo_cif);
+	i2s->reg_ctrl |= TEGRA30_I2S_CTRL_XFER_EN_RX;
+	tegra30_i2s_write(i2s, TEGRA30_I2S_CTRL, i2s->reg_ctrl);
+}
+
+static void tegra30_i2s_stop_capture(struct tegra30_i2s *i2s)
+{
+	tegra30_ahub_disable_rx_fifo(i2s->capture_fifo_cif);
+	i2s->reg_ctrl &= ~TEGRA30_I2S_CTRL_XFER_EN_RX;
+	tegra30_i2s_write(i2s, TEGRA30_I2S_CTRL, i2s->reg_ctrl);
+}
+
+static int tegra30_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
+				struct snd_soc_dai *dai)
+{
+	struct tegra30_i2s *i2s = snd_soc_dai_get_drvdata(dai);
+
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_START:
+	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+	case SNDRV_PCM_TRIGGER_RESUME:
+		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+			tegra30_i2s_start_playback(i2s);
+		else
+			tegra30_i2s_start_capture(i2s);
+		break;
+	case SNDRV_PCM_TRIGGER_STOP:
+	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+	case SNDRV_PCM_TRIGGER_SUSPEND:
+		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+			tegra30_i2s_stop_playback(i2s);
+		else
+			tegra30_i2s_stop_capture(i2s);
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	return 0;
+}
+
+static int tegra30_i2s_probe(struct snd_soc_dai *dai)
+{
+	struct tegra30_i2s *i2s = snd_soc_dai_get_drvdata(dai);
+
+	dai->capture_dma_data = &i2s->capture_dma_data;
+	dai->playback_dma_data = &i2s->playback_dma_data;
+
+	return 0;
+}
+
+static struct snd_soc_dai_ops tegra30_i2s_dai_ops = {
+	.startup	= tegra30_i2s_startup,
+	.shutdown	= tegra30_i2s_shutdown,
+	.set_fmt	= tegra30_i2s_set_fmt,
+	.hw_params	= tegra30_i2s_hw_params,
+	.trigger	= tegra30_i2s_trigger,
+};
+
+static const struct snd_soc_dai_driver tegra30_i2s_dai_template = {
+	.probe = tegra30_i2s_probe,
+	.playback = {
+		.channels_min = 2,
+		.channels_max = 2,
+		.rates = SNDRV_PCM_RATE_8000_96000,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE,
+	},
+	.capture = {
+		.channels_min = 2,
+		.channels_max = 2,
+		.rates = SNDRV_PCM_RATE_8000_96000,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE,
+	},
+	.ops = &tegra30_i2s_dai_ops,
+	.symmetric_rates = 1,
+};
+
+static bool tegra30_i2s_wr_rd_reg(struct device *dev, unsigned int reg)
+{
+	switch (reg) {
+	case TEGRA30_I2S_CTRL:
+	case TEGRA30_I2S_TIMING:
+	case TEGRA30_I2S_OFFSET:
+	case TEGRA30_I2S_CH_CTRL:
+	case TEGRA30_I2S_SLOT_CTRL:
+	case TEGRA30_I2S_CIF_RX_CTRL:
+	case TEGRA30_I2S_CIF_TX_CTRL:
+	case TEGRA30_I2S_FLOWCTL:
+	case TEGRA30_I2S_TX_STEP:
+	case TEGRA30_I2S_FLOW_STATUS:
+	case TEGRA30_I2S_FLOW_TOTAL:
+	case TEGRA30_I2S_FLOW_OVER:
+	case TEGRA30_I2S_FLOW_UNDER:
+	case TEGRA30_I2S_LCOEF_1_4_0:
+	case TEGRA30_I2S_LCOEF_1_4_1:
+	case TEGRA30_I2S_LCOEF_1_4_2:
+	case TEGRA30_I2S_LCOEF_1_4_3:
+	case TEGRA30_I2S_LCOEF_1_4_4:
+	case TEGRA30_I2S_LCOEF_1_4_5:
+	case TEGRA30_I2S_LCOEF_2_4_0:
+	case TEGRA30_I2S_LCOEF_2_4_1:
+	case TEGRA30_I2S_LCOEF_2_4_2:
+		return true;
+	default:
+		return false;
+	};
+}
+
+static bool tegra30_i2s_volatile_reg(struct device *dev, unsigned int reg)
+{
+	switch (reg) {
+	case TEGRA30_I2S_FLOW_STATUS:
+	case TEGRA30_I2S_FLOW_TOTAL:
+	case TEGRA30_I2S_FLOW_OVER:
+	case TEGRA30_I2S_FLOW_UNDER:
+		return true;
+	default:
+		return false;
+	};
+}
+
+static const struct regmap_config tegra30_i2s_regmap_config = {
+	.reg_bits = 32,
+	.reg_stride = 4,
+	.val_bits = 32,
+	.max_register = TEGRA30_I2S_LCOEF_2_4_2,
+	.writeable_reg = tegra30_i2s_wr_rd_reg,
+	.readable_reg = tegra30_i2s_wr_rd_reg,
+	.volatile_reg = tegra30_i2s_volatile_reg,
+	.cache_type = REGCACHE_RBTREE,
+};
+
+static __devinit int tegra30_i2s_platform_probe(struct platform_device *pdev)
+{
+	struct tegra30_i2s *i2s;
+	u32 cif_ids[2];
+	struct resource *mem, *memregion;
+	void __iomem *regs;
+	int ret;
+
+	i2s = devm_kzalloc(&pdev->dev, sizeof(struct tegra30_i2s), GFP_KERNEL);
+	if (!i2s) {
+		dev_err(&pdev->dev, "Can't allocate tegra30_i2s\n");
+		ret = -ENOMEM;
+		goto err;
+	}
+	dev_set_drvdata(&pdev->dev, i2s);
+
+	i2s->dai = tegra30_i2s_dai_template;
+	i2s->dai.name = dev_name(&pdev->dev);
+
+	ret = of_property_read_u32_array(pdev->dev.of_node,
+					 "nvidia,ahub-cif-ids", cif_ids,
+					 ARRAY_SIZE(cif_ids));
+	if (ret < 0)
+		goto err;
+
+	i2s->playback_i2s_cif = cif_ids[0];
+	i2s->capture_i2s_cif = cif_ids[1];
+
+	i2s->clk_i2s = clk_get(&pdev->dev, NULL);
+	if (IS_ERR(i2s->clk_i2s)) {
+		dev_err(&pdev->dev, "Can't retrieve i2s clock\n");
+		ret = PTR_ERR(i2s->clk_i2s);
+		goto err;
+	}
+
+	mem = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+	if (!mem) {
+		dev_err(&pdev->dev, "No memory resource\n");
+		ret = -ENODEV;
+		goto err_clk_put;
+	}
+
+	memregion = devm_request_mem_region(&pdev->dev, mem->start,
+					    resource_size(mem), DRV_NAME);
+	if (!memregion) {
+		dev_err(&pdev->dev, "Memory region already claimed\n");
+		ret = -EBUSY;
+		goto err_clk_put;
+	}
+
+	regs = devm_ioremap(&pdev->dev, mem->start, resource_size(mem));
+	if (!regs) {
+		dev_err(&pdev->dev, "ioremap failed\n");
+		ret = -ENOMEM;
+		goto err_clk_put;
+	}
+
+	i2s->regmap = devm_regmap_init_mmio(&pdev->dev, regs,
+					    &tegra30_i2s_regmap_config);
+	if (IS_ERR(i2s->regmap)) {
+		dev_err(&pdev->dev, "regmap init failed\n");
+		ret = PTR_ERR(i2s->regmap);
+		goto err_clk_put;
+	}
+	regcache_cache_only(i2s->regmap, true);
+
+	pm_runtime_enable(&pdev->dev);
+	if (!pm_runtime_enabled(&pdev->dev)) {
+		ret = tegra30_i2s_runtime_resume(&pdev->dev);
+		if (ret)
+			goto err_pm_disable;
+	}
+
+	ret = snd_soc_register_dai(&pdev->dev, &i2s->dai);
+	if (ret) {
+		dev_err(&pdev->dev, "Could not register DAI: %d\n", ret);
+		ret = -ENOMEM;
+		goto err_suspend;
+	}
+
+	ret = tegra_pcm_platform_register(&pdev->dev);
+	if (ret) {
+		dev_err(&pdev->dev, "Could not register PCM: %d\n", ret);
+		goto err_unregister_dai;
+	}
+
+	return 0;
+
+err_unregister_dai:
+	snd_soc_unregister_dai(&pdev->dev);
+err_suspend:
+	if (!pm_runtime_status_suspended(&pdev->dev))
+		tegra30_i2s_runtime_suspend(&pdev->dev);
+err_pm_disable:
+	pm_runtime_disable(&pdev->dev);
+err_clk_put:
+	clk_put(i2s->clk_i2s);
+err:
+	return ret;
+}
+
+static int __devexit tegra30_i2s_platform_remove(struct platform_device *pdev)
+{
+	struct tegra30_i2s *i2s = dev_get_drvdata(&pdev->dev);
+
+	pm_runtime_disable(&pdev->dev);
+	if (!pm_runtime_status_suspended(&pdev->dev))
+		tegra30_i2s_runtime_suspend(&pdev->dev);
+
+	tegra_pcm_platform_unregister(&pdev->dev);
+	snd_soc_unregister_dai(&pdev->dev);
+
+	clk_put(i2s->clk_i2s);
+
+	return 0;
+}
+
+static const struct of_device_id tegra30_i2s_of_match[] __devinitconst = {
+	{ .compatible = "nvidia,tegra30-i2s", },
+	{},
+};
+
+static const struct dev_pm_ops tegra30_i2s_pm_ops __devinitconst = {
+	SET_RUNTIME_PM_OPS(tegra30_i2s_runtime_suspend,
+			   tegra30_i2s_runtime_resume, NULL)
+};
+
+static struct platform_driver tegra30_i2s_driver = {
+	.driver = {
+		.name = DRV_NAME,
+		.owner = THIS_MODULE,
+		.of_match_table = tegra30_i2s_of_match,
+		.pm = &tegra30_i2s_pm_ops,
+	},
+	.probe = tegra30_i2s_platform_probe,
+	.remove = __devexit_p(tegra30_i2s_platform_remove),
+};
+module_platform_driver(tegra30_i2s_driver);
+
+MODULE_AUTHOR("Stephen Warren <swarren@nvidia.com>");
+MODULE_DESCRIPTION("Tegra30 I2S ASoC driver");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:" DRV_NAME);
+MODULE_DEVICE_TABLE(of, tegra30_i2s_of_match);
diff --git a/sound/soc/tegra/tegra30_i2s.h b/sound/soc/tegra/tegra30_i2s.h
new file mode 100644
index 000000000000..91adf29c7a87
--- /dev/null
+++ b/sound/soc/tegra/tegra30_i2s.h
@@ -0,0 +1,242 @@
+/*
+ * tegra30_i2s.h - Definitions for Tegra30 I2S driver
+ *
+ * Copyright (c) 2011,2012, NVIDIA CORPORATION.  All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms and conditions of the GNU General Public License,
+ * version 2, as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope 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, see <http://www.gnu.org/licenses/>.
+ */
+
+#ifndef __TEGRA30_I2S_H__
+#define __TEGRA30_I2S_H__
+
+#include "tegra_pcm.h"
+
+/* Register offsets from TEGRA30_I2S*_BASE */
+
+#define TEGRA30_I2S_CTRL				0x0
+#define TEGRA30_I2S_TIMING				0x4
+#define TEGRA30_I2S_OFFSET				0x08
+#define TEGRA30_I2S_CH_CTRL				0x0c
+#define TEGRA30_I2S_SLOT_CTRL				0x10
+#define TEGRA30_I2S_CIF_RX_CTRL				0x14
+#define TEGRA30_I2S_CIF_TX_CTRL				0x18
+#define TEGRA30_I2S_FLOWCTL				0x1c
+#define TEGRA30_I2S_TX_STEP				0x20
+#define TEGRA30_I2S_FLOW_STATUS				0x24
+#define TEGRA30_I2S_FLOW_TOTAL				0x28
+#define TEGRA30_I2S_FLOW_OVER				0x2c
+#define TEGRA30_I2S_FLOW_UNDER				0x30
+#define TEGRA30_I2S_LCOEF_1_4_0				0x34
+#define TEGRA30_I2S_LCOEF_1_4_1				0x38
+#define TEGRA30_I2S_LCOEF_1_4_2				0x3c
+#define TEGRA30_I2S_LCOEF_1_4_3				0x40
+#define TEGRA30_I2S_LCOEF_1_4_4				0x44
+#define TEGRA30_I2S_LCOEF_1_4_5				0x48
+#define TEGRA30_I2S_LCOEF_2_4_0				0x4c
+#define TEGRA30_I2S_LCOEF_2_4_1				0x50
+#define TEGRA30_I2S_LCOEF_2_4_2				0x54
+
+/* Fields in TEGRA30_I2S_CTRL */
+
+#define TEGRA30_I2S_CTRL_XFER_EN_TX			(1 << 31)
+#define TEGRA30_I2S_CTRL_XFER_EN_RX			(1 << 30)
+#define TEGRA30_I2S_CTRL_CG_EN				(1 << 29)
+#define TEGRA30_I2S_CTRL_SOFT_RESET			(1 << 28)
+#define TEGRA30_I2S_CTRL_TX_FLOWCTL_EN			(1 << 27)
+
+#define TEGRA30_I2S_CTRL_OBS_SEL_SHIFT			24
+#define TEGRA30_I2S_CTRL_OBS_SEL_MASK			(7 << TEGRA30_I2S_CTRL_OBS_SEL_SHIFT)
+
+#define TEGRA30_I2S_FRAME_FORMAT_LRCK			0
+#define TEGRA30_I2S_FRAME_FORMAT_FSYNC			1
+
+#define TEGRA30_I2S_CTRL_FRAME_FORMAT_SHIFT		12
+#define TEGRA30_I2S_CTRL_FRAME_FORMAT_MASK		(7                              << TEGRA30_I2S_CTRL_FRAME_FORMAT_SHIFT)
+#define TEGRA30_I2S_CTRL_FRAME_FORMAT_LRCK		(TEGRA30_I2S_FRAME_FORMAT_LRCK  << TEGRA30_I2S_CTRL_FRAME_FORMAT_SHIFT)
+#define TEGRA30_I2S_CTRL_FRAME_FORMAT_FSYNC		(TEGRA30_I2S_FRAME_FORMAT_FSYNC << TEGRA30_I2S_CTRL_FRAME_FORMAT_SHIFT)
+
+#define TEGRA30_I2S_CTRL_MASTER_ENABLE			(1 << 10)
+
+#define TEGRA30_I2S_LRCK_LEFT_LOW			0
+#define TEGRA30_I2S_LRCK_RIGHT_LOW			1
+
+#define TEGRA30_I2S_CTRL_LRCK_SHIFT			9
+#define TEGRA30_I2S_CTRL_LRCK_MASK			(1                          << TEGRA30_I2S_CTRL_LRCK_SHIFT)
+#define TEGRA30_I2S_CTRL_LRCK_L_LOW			(TEGRA30_I2S_LRCK_LEFT_LOW  << TEGRA30_I2S_CTRL_LRCK_SHIFT)
+#define TEGRA30_I2S_CTRL_LRCK_R_LOW			(TEGRA30_I2S_LRCK_RIGHT_LOW << TEGRA30_I2S_CTRL_LRCK_SHIFT)
+
+#define TEGRA30_I2S_CTRL_LPBK_ENABLE			(1 << 8)
+
+#define TEGRA30_I2S_BIT_CODE_LINEAR			0
+#define TEGRA30_I2S_BIT_CODE_ULAW			1
+#define TEGRA30_I2S_BIT_CODE_ALAW			2
+
+#define TEGRA30_I2S_CTRL_BIT_CODE_SHIFT			4
+#define TEGRA30_I2S_CTRL_BIT_CODE_MASK			(3                           << TEGRA30_I2S_CTRL_BIT_CODE_SHIFT)
+#define TEGRA30_I2S_CTRL_BIT_CODE_LINEAR		(TEGRA30_I2S_BIT_CODE_LINEAR << TEGRA30_I2S_CTRL_BIT_CODE_SHIFT)
+#define TEGRA30_I2S_CTRL_BIT_CODE_ULAW			(TEGRA30_I2S_BIT_CODE_ULAW   << TEGRA30_I2S_CTRL_BIT_CODE_SHIFT)
+#define TEGRA30_I2S_CTRL_BIT_CODE_ALAW			(TEGRA30_I2S_BIT_CODE_ALAW   << TEGRA30_I2S_CTRL_BIT_CODE_SHIFT)
+
+#define TEGRA30_I2S_BITS_8				1
+#define TEGRA30_I2S_BITS_12				2
+#define TEGRA30_I2S_BITS_16				3
+#define TEGRA30_I2S_BITS_20				4
+#define TEGRA30_I2S_BITS_24				5
+#define TEGRA30_I2S_BITS_28				6
+#define TEGRA30_I2S_BITS_32				7
+
+/* Sample container size; see {RX,TX}_MASK field in CH_CTRL below */
+#define TEGRA30_I2S_CTRL_BIT_SIZE_SHIFT			0
+#define TEGRA30_I2S_CTRL_BIT_SIZE_MASK			(7                   << TEGRA30_I2S_CTRL_BIT_SIZE_SHIFT)
+#define TEGRA30_I2S_CTRL_BIT_SIZE_8			(TEGRA30_I2S_BITS_8  << TEGRA30_I2S_CTRL_BIT_SIZE_SHIFT)
+#define TEGRA30_I2S_CTRL_BIT_SIZE_12			(TEGRA30_I2S_BITS_12 << TEGRA30_I2S_CTRL_BIT_SIZE_SHIFT)
+#define TEGRA30_I2S_CTRL_BIT_SIZE_16			(TEGRA30_I2S_BITS_16 << TEGRA30_I2S_CTRL_BIT_SIZE_SHIFT)
+#define TEGRA30_I2S_CTRL_BIT_SIZE_20			(TEGRA30_I2S_BITS_20 << TEGRA30_I2S_CTRL_BIT_SIZE_SHIFT)
+#define TEGRA30_I2S_CTRL_BIT_SIZE_24			(TEGRA30_I2S_BITS_24 << TEGRA30_I2S_CTRL_BIT_SIZE_SHIFT)
+#define TEGRA30_I2S_CTRL_BIT_SIZE_28			(TEGRA30_I2S_BITS_28 << TEGRA30_I2S_CTRL_BIT_SIZE_SHIFT)
+#define TEGRA30_I2S_CTRL_BIT_SIZE_32			(TEGRA30_I2S_BITS_32 << TEGRA30_I2S_CTRL_BIT_SIZE_SHIFT)
+
+/* Fields in TEGRA30_I2S_TIMING */
+
+#define TEGRA30_I2S_TIMING_NON_SYM_ENABLE		(1 << 12)
+#define TEGRA30_I2S_TIMING_CHANNEL_BIT_COUNT_SHIFT	0
+#define TEGRA30_I2S_TIMING_CHANNEL_BIT_COUNT_MASK_US	0x7fff
+#define TEGRA30_I2S_TIMING_CHANNEL_BIT_COUNT_MASK	(TEGRA30_I2S_TIMING_CHANNEL_BIT_COUNT_MASK_US << TEGRA30_I2S_TIMING_CHANNEL_BIT_COUNT_SHIFT)
+
+/* Fields in TEGRA30_I2S_OFFSET */
+
+#define TEGRA30_I2S_OFFSET_RX_DATA_OFFSET_SHIFT		16
+#define TEGRA30_I2S_OFFSET_RX_DATA_OFFSET_MASK_US	0x7ff
+#define TEGRA30_I2S_OFFSET_RX_DATA_OFFSET_MASK		(TEGRA30_I2S_OFFSET_RX_DATA_OFFSET_MASK_US << TEGRA30_I2S_OFFSET_RX_DATA_OFFSET_SHIFT)
+#define TEGRA30_I2S_OFFSET_TX_DATA_OFFSET_SHIFT		0
+#define TEGRA30_I2S_OFFSET_TX_DATA_OFFSET_MASK_US	0x7ff
+#define TEGRA30_I2S_OFFSET_TX_DATA_OFFSET_MASK		(TEGRA30_I2S_OFFSET_TX_DATA_OFFSET_MASK_US << TEGRA30_I2S_OFFSET_TX_DATA_OFFSET_SHIFT)
+
+/* Fields in TEGRA30_I2S_CH_CTRL */
+
+/* (FSYNC width - 1) in bit clocks */
+#define TEGRA30_I2S_CH_CTRL_FSYNC_WIDTH_SHIFT		24
+#define TEGRA30_I2S_CH_CTRL_FSYNC_WIDTH_MASK_US		0xff
+#define TEGRA30_I2S_CH_CTRL_FSYNC_WIDTH_MASK		(TEGRA30_I2S_CH_CTRL_FSYNC_WIDTH_MASK_US << TEGRA30_I2S_CH_CTRL_FSYNC_WIDTH_SHIFT)
+
+#define TEGRA30_I2S_HIGHZ_NO				0
+#define TEGRA30_I2S_HIGHZ_YES				1
+#define TEGRA30_I2S_HIGHZ_ON_HALF_BIT_CLK		2
+
+#define TEGRA30_I2S_CH_CTRL_HIGHZ_CTRL_SHIFT		12
+#define TEGRA30_I2S_CH_CTRL_HIGHZ_CTRL_MASK		(3                                 << TEGRA30_I2S_CH_CTRL_HIGHZ_CTRL_SHIFT)
+#define TEGRA30_I2S_CH_CTRL_HIGHZ_CTRL_NO		(TEGRA30_I2S_HIGHZ_NO              << TEGRA30_I2S_CH_CTRL_HIGHZ_CTRL_SHIFT)
+#define TEGRA30_I2S_CH_CTRL_HIGHZ_CTRL_YES		(TEGRA30_I2S_HIGHZ_YES             << TEGRA30_I2S_CH_CTRL_HIGHZ_CTRL_SHIFT)
+#define TEGRA30_I2S_CH_CTRL_HIGHZ_CTRL_ON_HALF_BIT_CLK	(TEGRA30_I2S_HIGHZ_ON_HALF_BIT_CLK << TEGRA30_I2S_CH_CTRL_HIGHZ_CTRL_SHIFT)
+
+#define TEGRA30_I2S_MSB_FIRST				0
+#define TEGRA30_I2S_LSB_FIRST				1
+
+#define TEGRA30_I2S_CH_CTRL_RX_BIT_ORDER_SHIFT		10
+#define TEGRA30_I2S_CH_CTRL_RX_BIT_ORDER_MASK		(1                     << TEGRA30_I2S_CH_CTRL_RX_BIT_ORDER_SHIFT)
+#define TEGRA30_I2S_CH_CTRL_RX_BIT_ORDER_MSB_FIRST	(TEGRA30_I2S_MSB_FIRST << TEGRA30_I2S_CH_CTRL_RX_BIT_ORDER_SHIFT)
+#define TEGRA30_I2S_CH_CTRL_RX_BIT_ORDER_LSB_FIRST	(TEGRA30_I2S_LSB_FIRST << TEGRA30_I2S_CH_CTRL_RX_BIT_ORDER_SHIFT)
+#define TEGRA30_I2S_CH_CTRL_TX_BIT_ORDER_SHIFT		9
+#define TEGRA30_I2S_CH_CTRL_TX_BIT_ORDER_MASK		(1                     << TEGRA30_I2S_CH_CTRL_TX_BIT_ORDER_SHIFT)
+#define TEGRA30_I2S_CH_CTRL_TX_BIT_ORDER_MSB_FIRST	(TEGRA30_I2S_MSB_FIRST << TEGRA30_I2S_CH_CTRL_TX_BIT_ORDER_SHIFT)
+#define TEGRA30_I2S_CH_CTRL_TX_BIT_ORDER_LSB_FIRST	(TEGRA30_I2S_LSB_FIRST << TEGRA30_I2S_CH_CTRL_TX_BIT_ORDER_SHIFT)
+
+#define TEGRA30_I2S_POS_EDGE				0
+#define TEGRA30_I2S_NEG_EDGE				1
+
+#define TEGRA30_I2S_CH_CTRL_EGDE_CTRL_SHIFT		8
+#define TEGRA30_I2S_CH_CTRL_EGDE_CTRL_MASK		(1                    << TEGRA30_I2S_CH_CTRL_EGDE_CTRL_SHIFT)
+#define TEGRA30_I2S_CH_CTRL_EGDE_CTRL_POS_EDGE		(TEGRA30_I2S_POS_EDGE << TEGRA30_I2S_CH_CTRL_EGDE_CTRL_SHIFT)
+#define TEGRA30_I2S_CH_CTRL_EGDE_CTRL_NEG_EDGE		(TEGRA30_I2S_NEG_EDGE << TEGRA30_I2S_CH_CTRL_EGDE_CTRL_SHIFT)
+
+/* Sample size is # bits from BIT_SIZE minus this field */
+#define TEGRA30_I2S_CH_CTRL_RX_MASK_BITS_SHIFT		4
+#define TEGRA30_I2S_CH_CTRL_RX_MASK_BITS_MASK_US	7
+#define TEGRA30_I2S_CH_CTRL_RX_MASK_BITS_MASK		(TEGRA30_I2S_CH_CTRL_RX_MASK_BITS_MASK_US << TEGRA30_I2S_CH_CTRL_RX_MASK_BITS_SHIFT)
+
+#define TEGRA30_I2S_CH_CTRL_TX_MASK_BITS_SHIFT		0
+#define TEGRA30_I2S_CH_CTRL_TX_MASK_BITS_MASK_US	7
+#define TEGRA30_I2S_CH_CTRL_TX_MASK_BITS_MASK		(TEGRA30_I2S_CH_CTRL_TX_MASK_BITS_MASK_US << TEGRA30_I2S_CH_CTRL_TX_MASK_BITS_SHIFT)
+
+/* Fields in TEGRA30_I2S_SLOT_CTRL */
+
+/* Number of slots in frame, minus 1 */
+#define TEGRA30_I2S_SLOT_CTRL_TOTAL_SLOTS_SHIFT		16
+#define TEGRA30_I2S_SLOT_CTRL_TOTAL_SLOTS_MASK_US	7
+#define TEGRA30_I2S_SLOT_CTRL_TOTAL_SLOTS_MASK		(TEGRA30_I2S_SLOT_CTRL_TOTAL_SLOT_MASK_US << TEGRA30_I2S_SLOT_CTRL_TOTAL_SLOT_SHIFT)
+
+/* TDM mode slot enable bitmask */
+#define TEGRA30_I2S_SLOT_CTRL_RX_SLOT_ENABLES_SHIFT	8
+#define TEGRA30_I2S_SLOT_CTRL_RX_SLOT_ENABLES_MASK	(0xff << TEGRA30_I2S_SLOT_CTRL_RX_SLOT_ENABLES_SHIFT)
+
+#define TEGRA30_I2S_SLOT_CTRL_TX_SLOT_ENABLES_SHIFT	0
+#define TEGRA30_I2S_SLOT_CTRL_TX_SLOT_ENABLES_MASK	(0xff << TEGRA30_I2S_SLOT_CTRL_TX_SLOT_ENABLES_SHIFT)
+
+/* Fields in TEGRA30_I2S_CIF_RX_CTRL */
+/* Uses field from TEGRA30_AUDIOCIF_CTRL_* in tegra30_ahub.h */
+
+/* Fields in TEGRA30_I2S_CIF_TX_CTRL */
+/* Uses field from TEGRA30_AUDIOCIF_CTRL_* in tegra30_ahub.h */
+
+/* Fields in TEGRA30_I2S_FLOWCTL */
+
+#define TEGRA30_I2S_FILTER_LINEAR			0
+#define TEGRA30_I2S_FILTER_QUAD				1
+
+#define TEGRA30_I2S_FLOWCTL_FILTER_SHIFT		31
+#define TEGRA30_I2S_FLOWCTL_FILTER_MASK			(1                         << TEGRA30_I2S_FLOWCTL_FILTER_SHIFT)
+#define TEGRA30_I2S_FLOWCTL_FILTER_LINEAR		(TEGRA30_I2S_FILTER_LINEAR << TEGRA30_I2S_FLOWCTL_FILTER_SHIFT)
+#define TEGRA30_I2S_FLOWCTL_FILTER_QUAD			(TEGRA30_I2S_FILTER_QUAD   << TEGRA30_I2S_FLOWCTL_FILTER_SHIFT)
+
+/* Fields in TEGRA30_I2S_TX_STEP */
+
+#define TEGRA30_I2S_TX_STEP_SHIFT			0
+#define TEGRA30_I2S_TX_STEP_MASK_US			0xffff
+#define TEGRA30_I2S_TX_STEP_MASK			(TEGRA30_I2S_TX_STEP_MASK_US << TEGRA30_I2S_TX_STEP_SHIFT)
+
+/* Fields in TEGRA30_I2S_FLOW_STATUS */
+
+#define TEGRA30_I2S_FLOW_STATUS_UNDERFLOW		(1 << 31)
+#define TEGRA30_I2S_FLOW_STATUS_OVERFLOW		(1 << 30)
+#define TEGRA30_I2S_FLOW_STATUS_MONITOR_INT_EN		(1 << 4)
+#define TEGRA30_I2S_FLOW_STATUS_COUNTER_CLR		(1 << 3)
+#define TEGRA30_I2S_FLOW_STATUS_MONITOR_CLR		(1 << 2)
+#define TEGRA30_I2S_FLOW_STATUS_COUNTER_EN		(1 << 1)
+#define TEGRA30_I2S_FLOW_STATUS_MONITOR_EN		(1 << 0)
+
+/*
+ * There are no fields in TEGRA30_I2S_FLOW_TOTAL, TEGRA30_I2S_FLOW_OVER,
+ * TEGRA30_I2S_FLOW_UNDER; they are counters taking the whole register.
+ */
+
+/* Fields in TEGRA30_I2S_LCOEF_* */
+
+#define TEGRA30_I2S_LCOEF_COEF_SHIFT			0
+#define TEGRA30_I2S_LCOEF_COEF_MASK_US			0xffff
+#define TEGRA30_I2S_LCOEF_COEF_MASK			(TEGRA30_I2S_LCOEF_COEF_MASK_US << TEGRA30_I2S_LCOEF_COEF_SHIFT)
+
+struct tegra30_i2s {
+	struct snd_soc_dai_driver dai;
+	int cif_id;
+	struct clk *clk_i2s;
+	enum tegra30_ahub_txcif capture_i2s_cif;
+	enum tegra30_ahub_rxcif capture_fifo_cif;
+	struct tegra_pcm_dma_params capture_dma_data;
+	enum tegra30_ahub_rxcif playback_i2s_cif;
+	enum tegra30_ahub_txcif playback_fifo_cif;
+	struct tegra_pcm_dma_params playback_dma_data;
+	struct regmap *regmap;
+	u32 reg_ctrl;
+};
+
+#endif
diff --git a/sound/soc/tegra/tegra_alc5632.c b/sound/soc/tegra/tegra_alc5632.c
index e45ccd851f6a..32de7006daf0 100644
--- a/sound/soc/tegra/tegra_alc5632.c
+++ b/sound/soc/tegra/tegra_alc5632.c
@@ -1,16 +1,17 @@
 /*
-* tegra_alc5632.c  --  Toshiba AC100(PAZ00) machine ASoC driver
-*
-* Copyright (C) 2011 The AC100 Kernel Team <ac100@lists.lauchpad.net>
-*
-* Authors:  Leon Romanovsky <leon@leon.nu>
-*           Andrey Danin <danindrey@mail.ru>
-*           Marc Dietrich <marvin24@gmx.de>
-*
-* 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.
-*/
+ * tegra_alc5632.c  --  Toshiba AC100(PAZ00) machine ASoC driver
+ *
+ * Copyright (C) 2011 The AC100 Kernel Team <ac100@lists.lauchpad.net>
+ * Copyright (C) 2012 - NVIDIA, Inc.
+ *
+ * Authors:  Leon Romanovsky <leon@leon.nu>
+ *           Andrey Danin <danindrey@mail.ru>
+ *           Marc Dietrich <marvin24@gmx.de>
+ *
+ * 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 <asm/mach-types.h>
 
@@ -28,9 +29,6 @@
 
 #include "../codecs/alc5632.h"
 
-#include "tegra_das.h"
-#include "tegra_i2s.h"
-#include "tegra_pcm.h"
 #include "tegra_asoc_utils.h"
 
 #define DRV_NAME "tegra-alc5632"
@@ -39,7 +37,6 @@
 
 struct tegra_alc5632 {
 	struct tegra_asoc_utils_data util_data;
-	struct platform_device *pcm_dev;
 	int gpio_requested;
 	int gpio_hp_det;
 };
@@ -140,7 +137,6 @@ static int tegra_alc5632_asoc_init(struct snd_soc_pcm_runtime *rtd)
 static struct snd_soc_dai_link tegra_alc5632_dai = {
 	.name = "ALC5632",
 	.stream_name = "ALC5632 PCM",
-	.platform_name = "tegra-pcm-audio",
 	.codec_dai_name = "alc5632-hifi",
 	.init = tegra_alc5632_asoc_init,
 	.ops = &tegra_alc5632_asoc_ops,
@@ -179,8 +175,6 @@ static __devinit int tegra_alc5632_probe(struct platform_device *pdev)
 	platform_set_drvdata(pdev, card);
 	snd_soc_card_set_drvdata(card, alc5632);
 
-	alc5632->pcm_dev = ERR_PTR(-EINVAL);
-
 	if (!(pdev->dev.of_node)) {
 		dev_err(&pdev->dev, "Must be instantiated using device tree\n");
 		ret = -EINVAL;
@@ -214,18 +208,11 @@ static __devinit int tegra_alc5632_probe(struct platform_device *pdev)
 		goto err;
 	}
 
-	alc5632->pcm_dev = platform_device_register_simple(
-		"tegra-pcm-audio", -1, NULL, 0);
-	if (IS_ERR(alc5632->pcm_dev)) {
-		dev_err(&pdev->dev,
-			"Can't instantiate tegra-pcm-audio\n");
-		ret = PTR_ERR(alc5632->pcm_dev);
-		goto err;
-	}
+	tegra_alc5632_dai.platform_of_node = tegra_alc5632_dai.cpu_dai_of_node;
 
 	ret = tegra_asoc_utils_init(&alc5632->util_data, &pdev->dev);
 	if (ret)
-		goto err_unregister;
+		goto err;
 
 	ret = snd_soc_register_card(card);
 	if (ret) {
@@ -238,9 +225,6 @@ static __devinit int tegra_alc5632_probe(struct platform_device *pdev)
 
 err_fini_utils:
 	tegra_asoc_utils_fini(&alc5632->util_data);
-err_unregister:
-	if (!IS_ERR(alc5632->pcm_dev))
-		platform_device_unregister(alc5632->pcm_dev);
 err:
 	return ret;
 }
@@ -259,8 +243,6 @@ static int __devexit tegra_alc5632_remove(struct platform_device *pdev)
 	snd_soc_unregister_card(card);
 
 	tegra_asoc_utils_fini(&machine->util_data);
-	if (!IS_ERR(machine->pcm_dev))
-		platform_device_unregister(machine->pcm_dev);
 
 	return 0;
 }
diff --git a/sound/soc/tegra/tegra_asoc_utils.c b/sound/soc/tegra/tegra_asoc_utils.c
index f8428e410e05..9515ce58ea02 100644
--- a/sound/soc/tegra/tegra_asoc_utils.c
+++ b/sound/soc/tegra/tegra_asoc_utils.c
@@ -2,7 +2,7 @@
  * tegra_asoc_utils.c - Harmony machine ASoC driver
  *
  * Author: Stephen Warren <swarren@nvidia.com>
- * Copyright (C) 2010 - NVIDIA, Inc.
+ * Copyright (C) 2010,2012 - NVIDIA, Inc.
  *
  * This program is free software; you can redistribute it and/or
  * modify it under the terms of the GNU General Public License
@@ -25,6 +25,7 @@
 #include <linux/err.h>
 #include <linux/kernel.h>
 #include <linux/module.h>
+#include <linux/of.h>
 
 #include "tegra_asoc_utils.h"
 
@@ -40,7 +41,10 @@ int tegra_asoc_utils_set_rate(struct tegra_asoc_utils_data *data, int srate,
 	case 22050:
 	case 44100:
 	case 88200:
-		new_baseclock = 56448000;
+		if (data->soc == TEGRA_ASOC_UTILS_SOC_TEGRA20)
+			new_baseclock = 56448000;
+		else
+			new_baseclock = 564480000;
 		break;
 	case 8000:
 	case 16000:
@@ -48,7 +52,10 @@ int tegra_asoc_utils_set_rate(struct tegra_asoc_utils_data *data, int srate,
 	case 48000:
 	case 64000:
 	case 96000:
-		new_baseclock = 73728000;
+		if (data->soc == TEGRA_ASOC_UTILS_SOC_TEGRA20)
+			new_baseclock = 73728000;
+		else
+			new_baseclock = 552960000;
 		break;
 	default:
 		return -EINVAL;
@@ -78,7 +85,7 @@ int tegra_asoc_utils_set_rate(struct tegra_asoc_utils_data *data, int srate,
 		return err;
 	}
 
-	/* Don't set cdev1 rate; its locked to pll_a_out0 */
+	/* Don't set cdev1/extern1 rate; it's locked to pll_a_out0 */
 
 	err = clk_enable(data->clk_pll_a);
 	if (err) {
@@ -112,6 +119,17 @@ int tegra_asoc_utils_init(struct tegra_asoc_utils_data *data,
 
 	data->dev = dev;
 
+	if (of_machine_is_compatible("nvidia,tegra20"))
+		data->soc = TEGRA_ASOC_UTILS_SOC_TEGRA20;
+	else if (of_machine_is_compatible("nvidia,tegra30"))
+		data->soc = TEGRA_ASOC_UTILS_SOC_TEGRA30;
+	else if (!dev->of_node)
+		/* non-DT is always Tegra20 */
+		data->soc = TEGRA_ASOC_UTILS_SOC_TEGRA20;
+	else
+		/* DT boot, but unknown SoC */
+		return -EINVAL;
+
 	data->clk_pll_a = clk_get_sys(NULL, "pll_a");
 	if (IS_ERR(data->clk_pll_a)) {
 		dev_err(data->dev, "Can't retrieve clk pll_a\n");
@@ -126,15 +144,24 @@ int tegra_asoc_utils_init(struct tegra_asoc_utils_data *data,
 		goto err_put_pll_a;
 	}
 
-	data->clk_cdev1 = clk_get_sys(NULL, "cdev1");
+	if (data->soc == TEGRA_ASOC_UTILS_SOC_TEGRA20)
+		data->clk_cdev1 = clk_get_sys(NULL, "cdev1");
+	else
+		data->clk_cdev1 = clk_get_sys("extern1", NULL);
 	if (IS_ERR(data->clk_cdev1)) {
 		dev_err(data->dev, "Can't retrieve clk cdev1\n");
 		ret = PTR_ERR(data->clk_cdev1);
 		goto err_put_pll_a_out0;
 	}
 
+	ret = tegra_asoc_utils_set_rate(data, 44100, 256 * 44100);
+	if (ret)
+		goto err_put_cdev1;
+
 	return 0;
 
+err_put_cdev1:
+	clk_put(data->clk_cdev1);
 err_put_pll_a_out0:
 	clk_put(data->clk_pll_a_out0);
 err_put_pll_a:
diff --git a/sound/soc/tegra/tegra_asoc_utils.h b/sound/soc/tegra/tegra_asoc_utils.h
index 4818195da25c..44db1dbb8f21 100644
--- a/sound/soc/tegra/tegra_asoc_utils.h
+++ b/sound/soc/tegra/tegra_asoc_utils.h
@@ -2,7 +2,7 @@
  * tegra_asoc_utils.h - Definitions for Tegra DAS driver
  *
  * Author: Stephen Warren <swarren@nvidia.com>
- * Copyright (C) 2010 - NVIDIA, Inc.
+ * Copyright (C) 2010,2012 - NVIDIA, Inc.
  *
  * This program is free software; you can redistribute it and/or
  * modify it under the terms of the GNU General Public License
@@ -26,8 +26,14 @@
 struct clk;
 struct device;
 
+enum tegra_asoc_utils_soc {
+	TEGRA_ASOC_UTILS_SOC_TEGRA20,
+	TEGRA_ASOC_UTILS_SOC_TEGRA30,
+};
+
 struct tegra_asoc_utils_data {
 	struct device *dev;
+	enum tegra_asoc_utils_soc soc;
 	struct clk *clk_pll_a;
 	struct clk *clk_pll_a_out0;
 	struct clk *clk_cdev1;
@@ -42,4 +48,3 @@ int tegra_asoc_utils_init(struct tegra_asoc_utils_data *data,
 void tegra_asoc_utils_fini(struct tegra_asoc_utils_data *data);
 
 #endif
-
diff --git a/sound/soc/tegra/tegra_das.c b/sound/soc/tegra/tegra_das.c
deleted file mode 100644
index 3b3c1ba4d235..000000000000
--- a/sound/soc/tegra/tegra_das.c
+++ /dev/null
@@ -1,261 +0,0 @@
-/*
- * tegra_das.c - Tegra DAS driver
- *
- * Author: Stephen Warren <swarren@nvidia.com>
- * Copyright (C) 2010 - NVIDIA, Inc.
- *
- * 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
- *
- */
-
-#include <linux/module.h>
-#include <linux/debugfs.h>
-#include <linux/device.h>
-#include <linux/platform_device.h>
-#include <linux/seq_file.h>
-#include <linux/slab.h>
-#include <linux/io.h>
-#include <mach/iomap.h>
-#include <sound/soc.h>
-#include "tegra_das.h"
-
-#define DRV_NAME "tegra-das"
-
-static struct tegra_das *das;
-
-static inline void tegra_das_write(u32 reg, u32 val)
-{
-	__raw_writel(val, das->regs + reg);
-}
-
-static inline u32 tegra_das_read(u32 reg)
-{
-	return __raw_readl(das->regs + reg);
-}
-
-int tegra_das_connect_dap_to_dac(int dap, int dac)
-{
-	u32 addr;
-	u32 reg;
-
-	if (!das)
-		return -ENODEV;
-
-	addr = TEGRA_DAS_DAP_CTRL_SEL +
-		(dap * TEGRA_DAS_DAP_CTRL_SEL_STRIDE);
-	reg = dac << TEGRA_DAS_DAP_CTRL_SEL_DAP_CTRL_SEL_P;
-
-	tegra_das_write(addr, reg);
-
-	return 0;
-}
-EXPORT_SYMBOL_GPL(tegra_das_connect_dap_to_dac);
-
-int tegra_das_connect_dap_to_dap(int dap, int otherdap, int master,
-					int sdata1rx, int sdata2rx)
-{
-	u32 addr;
-	u32 reg;
-
-	if (!das)
-		return -ENODEV;
-
-	addr = TEGRA_DAS_DAP_CTRL_SEL +
-		(dap * TEGRA_DAS_DAP_CTRL_SEL_STRIDE);
-	reg = otherdap << TEGRA_DAS_DAP_CTRL_SEL_DAP_CTRL_SEL_P |
-		!!sdata2rx << TEGRA_DAS_DAP_CTRL_SEL_DAP_SDATA2_TX_RX_P |
-		!!sdata1rx << TEGRA_DAS_DAP_CTRL_SEL_DAP_SDATA1_TX_RX_P |
-		!!master << TEGRA_DAS_DAP_CTRL_SEL_DAP_MS_SEL_P;
-
-	tegra_das_write(addr, reg);
-
-	return 0;
-}
-EXPORT_SYMBOL_GPL(tegra_das_connect_dap_to_dap);
-
-int tegra_das_connect_dac_to_dap(int dac, int dap)
-{
-	u32 addr;
-	u32 reg;
-
-	if (!das)
-		return -ENODEV;
-
-	addr = TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL +
-		(dac * TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_STRIDE);
-	reg = dap << TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_CLK_SEL_P |
-		dap << TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_SDATA1_SEL_P |
-		dap << TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_SDATA2_SEL_P;
-
-	tegra_das_write(addr, reg);
-
-	return 0;
-}
-EXPORT_SYMBOL_GPL(tegra_das_connect_dac_to_dap);
-
-#ifdef CONFIG_DEBUG_FS
-static int tegra_das_show(struct seq_file *s, void *unused)
-{
-	int i;
-	u32 addr;
-	u32 reg;
-
-	for (i = 0; i < TEGRA_DAS_DAP_CTRL_SEL_COUNT; i++) {
-		addr = TEGRA_DAS_DAP_CTRL_SEL +
-			(i * TEGRA_DAS_DAP_CTRL_SEL_STRIDE);
-		reg = tegra_das_read(addr);
-		seq_printf(s, "TEGRA_DAS_DAP_CTRL_SEL[%d] = %08x\n", i, reg);
-	}
-
-	for (i = 0; i < TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_COUNT; i++) {
-		addr = TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL +
-			(i * TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_STRIDE);
-		reg = tegra_das_read(addr);
-		seq_printf(s, "TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL[%d] = %08x\n",
-				 i, reg);
-	}
-
-	return 0;
-}
-
-static int tegra_das_debug_open(struct inode *inode, struct file *file)
-{
-	return single_open(file, tegra_das_show, inode->i_private);
-}
-
-static const struct file_operations tegra_das_debug_fops = {
-	.open    = tegra_das_debug_open,
-	.read    = seq_read,
-	.llseek  = seq_lseek,
-	.release = single_release,
-};
-
-static void tegra_das_debug_add(struct tegra_das *das)
-{
-	das->debug = debugfs_create_file(DRV_NAME, S_IRUGO,
-					 snd_soc_debugfs_root, das,
-					 &tegra_das_debug_fops);
-}
-
-static void tegra_das_debug_remove(struct tegra_das *das)
-{
-	if (das->debug)
-		debugfs_remove(das->debug);
-}
-#else
-static inline void tegra_das_debug_add(struct tegra_das *das)
-{
-}
-
-static inline void tegra_das_debug_remove(struct tegra_das *das)
-{
-}
-#endif
-
-static int __devinit tegra_das_probe(struct platform_device *pdev)
-{
-	struct resource *res, *region;
-	int ret = 0;
-
-	if (das)
-		return -ENODEV;
-
-	das = devm_kzalloc(&pdev->dev, sizeof(struct tegra_das), GFP_KERNEL);
-	if (!das) {
-		dev_err(&pdev->dev, "Can't allocate tegra_das\n");
-		ret = -ENOMEM;
-		goto err;
-	}
-	das->dev = &pdev->dev;
-
-	res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
-	if (!res) {
-		dev_err(&pdev->dev, "No memory resource\n");
-		ret = -ENODEV;
-		goto err;
-	}
-
-	region = devm_request_mem_region(&pdev->dev, res->start,
-					 resource_size(res), pdev->name);
-	if (!region) {
-		dev_err(&pdev->dev, "Memory region already claimed\n");
-		ret = -EBUSY;
-		goto err;
-	}
-
-	das->regs = devm_ioremap(&pdev->dev, res->start, resource_size(res));
-	if (!das->regs) {
-		dev_err(&pdev->dev, "ioremap failed\n");
-		ret = -ENOMEM;
-		goto err;
-	}
-
-	ret = tegra_das_connect_dap_to_dac(TEGRA_DAS_DAP_ID_1,
-					   TEGRA_DAS_DAP_SEL_DAC1);
-	if (ret) {
-		dev_err(&pdev->dev, "Can't set up DAS DAP connection\n");
-		goto err;
-	}
-	ret = tegra_das_connect_dac_to_dap(TEGRA_DAS_DAC_ID_1,
-					   TEGRA_DAS_DAC_SEL_DAP1);
-	if (ret) {
-		dev_err(&pdev->dev, "Can't set up DAS DAC connection\n");
-		goto err;
-	}
-
-	tegra_das_debug_add(das);
-
-	platform_set_drvdata(pdev, das);
-
-	return 0;
-
-err:
-	das = NULL;
-	return ret;
-}
-
-static int __devexit tegra_das_remove(struct platform_device *pdev)
-{
-	if (!das)
-		return -ENODEV;
-
-	tegra_das_debug_remove(das);
-
-	das = NULL;
-
-	return 0;
-}
-
-static const struct of_device_id tegra_das_of_match[] __devinitconst = {
-	{ .compatible = "nvidia,tegra20-das", },
-	{},
-};
-
-static struct platform_driver tegra_das_driver = {
-	.probe = tegra_das_probe,
-	.remove = __devexit_p(tegra_das_remove),
-	.driver = {
-		.name = DRV_NAME,
-		.owner = THIS_MODULE,
-		.of_match_table = tegra_das_of_match,
-	},
-};
-module_platform_driver(tegra_das_driver);
-
-MODULE_AUTHOR("Stephen Warren <swarren@nvidia.com>");
-MODULE_DESCRIPTION("Tegra DAS driver");
-MODULE_LICENSE("GPL");
-MODULE_ALIAS("platform:" DRV_NAME);
-MODULE_DEVICE_TABLE(of, tegra_das_of_match);
diff --git a/sound/soc/tegra/tegra_das.h b/sound/soc/tegra/tegra_das.h
deleted file mode 100644
index 2c96c7b3c459..000000000000
--- a/sound/soc/tegra/tegra_das.h
+++ /dev/null
@@ -1,135 +0,0 @@
-/*
- * tegra_das.h - Definitions for Tegra DAS driver
- *
- * Author: Stephen Warren <swarren@nvidia.com>
- * Copyright (C) 2010 - NVIDIA, Inc.
- *
- * 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 __TEGRA_DAS_H__
-#define __TEGRA_DAS_H__
-
-/* Register TEGRA_DAS_DAP_CTRL_SEL */
-#define TEGRA_DAS_DAP_CTRL_SEL				0x00
-#define TEGRA_DAS_DAP_CTRL_SEL_COUNT			5
-#define TEGRA_DAS_DAP_CTRL_SEL_STRIDE			4
-#define TEGRA_DAS_DAP_CTRL_SEL_DAP_MS_SEL_P		31
-#define TEGRA_DAS_DAP_CTRL_SEL_DAP_MS_SEL_S		1
-#define TEGRA_DAS_DAP_CTRL_SEL_DAP_SDATA1_TX_RX_P	30
-#define TEGRA_DAS_DAP_CTRL_SEL_DAP_SDATA1_TX_RX_S	1
-#define TEGRA_DAS_DAP_CTRL_SEL_DAP_SDATA2_TX_RX_P	29
-#define TEGRA_DAS_DAP_CTRL_SEL_DAP_SDATA2_TX_RX_S	1
-#define TEGRA_DAS_DAP_CTRL_SEL_DAP_CTRL_SEL_P		0
-#define TEGRA_DAS_DAP_CTRL_SEL_DAP_CTRL_SEL_S		5
-
-/* Values for field TEGRA_DAS_DAP_CTRL_SEL_DAP_CTRL_SEL */
-#define TEGRA_DAS_DAP_SEL_DAC1	0
-#define TEGRA_DAS_DAP_SEL_DAC2	1
-#define TEGRA_DAS_DAP_SEL_DAC3	2
-#define TEGRA_DAS_DAP_SEL_DAP1	16
-#define TEGRA_DAS_DAP_SEL_DAP2	17
-#define TEGRA_DAS_DAP_SEL_DAP3	18
-#define TEGRA_DAS_DAP_SEL_DAP4	19
-#define TEGRA_DAS_DAP_SEL_DAP5	20
-
-/* Register TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL */
-#define TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL			0x40
-#define TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_COUNT			3
-#define TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_STRIDE			4
-#define TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_SDATA2_SEL_P	28
-#define TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_SDATA2_SEL_S	4
-#define TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_SDATA1_SEL_P	24
-#define TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_SDATA1_SEL_S	4
-#define TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_CLK_SEL_P		0
-#define TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_CLK_SEL_S		4
-
-/*
- * Values for:
- * TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_SDATA2_SEL
- * TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_SDATA1_SEL
- * TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_CLK_SEL
- */
-#define TEGRA_DAS_DAC_SEL_DAP1	0
-#define TEGRA_DAS_DAC_SEL_DAP2	1
-#define TEGRA_DAS_DAC_SEL_DAP3	2
-#define TEGRA_DAS_DAC_SEL_DAP4	3
-#define TEGRA_DAS_DAC_SEL_DAP5	4
-
-/*
- * Names/IDs of the DACs/DAPs.
- */
-
-#define TEGRA_DAS_DAP_ID_1 0
-#define TEGRA_DAS_DAP_ID_2 1
-#define TEGRA_DAS_DAP_ID_3 2
-#define TEGRA_DAS_DAP_ID_4 3
-#define TEGRA_DAS_DAP_ID_5 4
-
-#define TEGRA_DAS_DAC_ID_1 0
-#define TEGRA_DAS_DAC_ID_2 1
-#define TEGRA_DAS_DAC_ID_3 2
-
-struct tegra_das {
-	struct device *dev;
-	void __iomem *regs;
-	struct dentry *debug;
-};
-
-/*
- * Terminology:
- * DAS: Digital audio switch (HW module controlled by this driver)
- * DAP: Digital audio port (port/pins on Tegra device)
- * DAC: Digital audio controller (e.g. I2S or AC97 controller elsewhere)
- * 
- * The Tegra DAS is a mux/cross-bar which can connect each DAP to a specific
- * DAC, or another DAP. When DAPs are connected, one must be the master and
- * one the slave. Each DAC allows selection of a specific DAP for input, to
- * cater for the case where N DAPs are connected to 1 DAC for broadcast
- * output.
- *
- * This driver is dumb; no attempt is made to ensure that a valid routing
- * configuration is programmed.
- */
-
-/*
- * Connect a DAP to to a DAC
- * dap_id: DAP to connect: TEGRA_DAS_DAP_ID_*
- * dac_sel: DAC to connect to: TEGRA_DAS_DAP_SEL_DAC*
- */
-extern int tegra_das_connect_dap_to_dac(int dap_id, int dac_sel);
-
-/*
- * Connect a DAP to to another DAP
- * dap_id: DAP to connect: TEGRA_DAS_DAP_ID_*
- * other_dap_sel: DAP to connect to: TEGRA_DAS_DAP_SEL_DAP*
- * master: Is this DAP the master (1) or slave (0)
- * sdata1rx: Is this DAP's SDATA1 pin RX (1) or TX (0)
- * sdata2rx: Is this DAP's SDATA2 pin RX (1) or TX (0)
- */
-extern int tegra_das_connect_dap_to_dap(int dap_id, int other_dap_sel,
-					int master, int sdata1rx,
-					int sdata2rx);
-
-/*
- * Connect a DAC's input to a DAP
- * (DAC outputs are selected by the DAP)
- * dac_id: DAC ID to connect: TEGRA_DAS_DAC_ID_*
- * dap_sel: DAP to receive input from: TEGRA_DAS_DAC_SEL_DAP*
- */
-extern int tegra_das_connect_dac_to_dap(int dac_id, int dap_sel);
-
-#endif
diff --git a/sound/soc/tegra/tegra_i2s.c b/sound/soc/tegra/tegra_i2s.c
deleted file mode 100644
index e53349912b2e..000000000000
--- a/sound/soc/tegra/tegra_i2s.c
+++ /dev/null
@@ -1,459 +0,0 @@
-/*
- * tegra_i2s.c - Tegra I2S driver
- *
- * Author: Stephen Warren <swarren@nvidia.com>
- * Copyright (C) 2010 - NVIDIA, Inc.
- *
- * Based on code copyright/by:
- *
- * Copyright (c) 2009-2010, NVIDIA Corporation.
- * Scott Peterson <speterson@nvidia.com>
- *
- * Copyright (C) 2010 Google, Inc.
- * Iliyan Malchev <malchev@google.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
- *
- */
-
-#include <linux/clk.h>
-#include <linux/module.h>
-#include <linux/debugfs.h>
-#include <linux/device.h>
-#include <linux/platform_device.h>
-#include <linux/seq_file.h>
-#include <linux/slab.h>
-#include <linux/io.h>
-#include <linux/of.h>
-#include <mach/iomap.h>
-#include <sound/core.h>
-#include <sound/pcm.h>
-#include <sound/pcm_params.h>
-#include <sound/soc.h>
-
-#include "tegra_i2s.h"
-
-#define DRV_NAME "tegra-i2s"
-
-static inline void tegra_i2s_write(struct tegra_i2s *i2s, u32 reg, u32 val)
-{
-	__raw_writel(val, i2s->regs + reg);
-}
-
-static inline u32 tegra_i2s_read(struct tegra_i2s *i2s, u32 reg)
-{
-	return __raw_readl(i2s->regs + reg);
-}
-
-#ifdef CONFIG_DEBUG_FS
-static int tegra_i2s_show(struct seq_file *s, void *unused)
-{
-#define REG(r) { r, #r }
-	static const struct {
-		int offset;
-		const char *name;
-	} regs[] = {
-		REG(TEGRA_I2S_CTRL),
-		REG(TEGRA_I2S_STATUS),
-		REG(TEGRA_I2S_TIMING),
-		REG(TEGRA_I2S_FIFO_SCR),
-		REG(TEGRA_I2S_PCM_CTRL),
-		REG(TEGRA_I2S_NW_CTRL),
-		REG(TEGRA_I2S_TDM_CTRL),
-		REG(TEGRA_I2S_TDM_TX_RX_CTRL),
-	};
-#undef REG
-
-	struct tegra_i2s *i2s = s->private;
-	int i;
-
-	clk_enable(i2s->clk_i2s);
-
-	for (i = 0; i < ARRAY_SIZE(regs); i++) {
-		u32 val = tegra_i2s_read(i2s, regs[i].offset);
-		seq_printf(s, "%s = %08x\n", regs[i].name, val);
-	}
-
-	clk_disable(i2s->clk_i2s);
-
-	return 0;
-}
-
-static int tegra_i2s_debug_open(struct inode *inode, struct file *file)
-{
-	return single_open(file, tegra_i2s_show, inode->i_private);
-}
-
-static const struct file_operations tegra_i2s_debug_fops = {
-	.open    = tegra_i2s_debug_open,
-	.read    = seq_read,
-	.llseek  = seq_lseek,
-	.release = single_release,
-};
-
-static void tegra_i2s_debug_add(struct tegra_i2s *i2s)
-{
-	i2s->debug = debugfs_create_file(i2s->dai.name, S_IRUGO,
-					 snd_soc_debugfs_root, i2s,
-					 &tegra_i2s_debug_fops);
-}
-
-static void tegra_i2s_debug_remove(struct tegra_i2s *i2s)
-{
-	if (i2s->debug)
-		debugfs_remove(i2s->debug);
-}
-#else
-static inline void tegra_i2s_debug_add(struct tegra_i2s *i2s)
-{
-}
-
-static inline void tegra_i2s_debug_remove(struct tegra_i2s *i2s)
-{
-}
-#endif
-
-static int tegra_i2s_set_fmt(struct snd_soc_dai *dai,
-				unsigned int fmt)
-{
-	struct tegra_i2s *i2s = snd_soc_dai_get_drvdata(dai);
-
-	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
-	case SND_SOC_DAIFMT_NB_NF:
-		break;
-	default:
-		return -EINVAL;
-	}
-
-	i2s->reg_ctrl &= ~TEGRA_I2S_CTRL_MASTER_ENABLE;
-	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
-	case SND_SOC_DAIFMT_CBS_CFS:
-		i2s->reg_ctrl |= TEGRA_I2S_CTRL_MASTER_ENABLE;
-		break;
-	case SND_SOC_DAIFMT_CBM_CFM:
-		break;
-	default:
-		return -EINVAL;
-	}
-
-	i2s->reg_ctrl &= ~(TEGRA_I2S_CTRL_BIT_FORMAT_MASK | 
-				TEGRA_I2S_CTRL_LRCK_MASK);
-	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
-	case SND_SOC_DAIFMT_DSP_A:
-		i2s->reg_ctrl |= TEGRA_I2S_CTRL_BIT_FORMAT_DSP;
-		i2s->reg_ctrl |= TEGRA_I2S_CTRL_LRCK_L_LOW;
-		break;
-	case SND_SOC_DAIFMT_DSP_B:
-		i2s->reg_ctrl |= TEGRA_I2S_CTRL_BIT_FORMAT_DSP;
-		i2s->reg_ctrl |= TEGRA_I2S_CTRL_LRCK_R_LOW;
-		break;
-	case SND_SOC_DAIFMT_I2S:
-		i2s->reg_ctrl |= TEGRA_I2S_CTRL_BIT_FORMAT_I2S;
-		i2s->reg_ctrl |= TEGRA_I2S_CTRL_LRCK_L_LOW;
-		break;
-	case SND_SOC_DAIFMT_RIGHT_J:
-		i2s->reg_ctrl |= TEGRA_I2S_CTRL_BIT_FORMAT_RJM;
-		i2s->reg_ctrl |= TEGRA_I2S_CTRL_LRCK_L_LOW;
-		break;
-	case SND_SOC_DAIFMT_LEFT_J:
-		i2s->reg_ctrl |= TEGRA_I2S_CTRL_BIT_FORMAT_LJM;
-		i2s->reg_ctrl |= TEGRA_I2S_CTRL_LRCK_L_LOW;
-		break;
-	default:
-		return -EINVAL;
-	}
-
-	return 0;
-}
-
-static int tegra_i2s_hw_params(struct snd_pcm_substream *substream,
-				struct snd_pcm_hw_params *params,
-				struct snd_soc_dai *dai)
-{
-        struct device *dev = substream->pcm->card->dev;
-	struct tegra_i2s *i2s = snd_soc_dai_get_drvdata(dai);
-	u32 reg;
-	int ret, sample_size, srate, i2sclock, bitcnt;
-
-	i2s->reg_ctrl &= ~TEGRA_I2S_CTRL_BIT_SIZE_MASK;
-	switch (params_format(params)) {
-	case SNDRV_PCM_FORMAT_S16_LE:
-		i2s->reg_ctrl |= TEGRA_I2S_CTRL_BIT_SIZE_16;
-		sample_size = 16;
-		break;
-	case SNDRV_PCM_FORMAT_S24_LE:
-		i2s->reg_ctrl |= TEGRA_I2S_CTRL_BIT_SIZE_24;
-		sample_size = 24;
-		break;
-	case SNDRV_PCM_FORMAT_S32_LE:
-		i2s->reg_ctrl |= TEGRA_I2S_CTRL_BIT_SIZE_32;
-		sample_size = 32;
-		break;
-	default:
-		return -EINVAL;
-	}
-
-	srate = params_rate(params);
-
-	/* Final "* 2" required by Tegra hardware */
-	i2sclock = srate * params_channels(params) * sample_size * 2;
-
-	ret = clk_set_rate(i2s->clk_i2s, i2sclock);
-	if (ret) {
-		dev_err(dev, "Can't set I2S clock rate: %d\n", ret);
-		return ret;
-	}
-
-	bitcnt = (i2sclock / (2 * srate)) - 1;
-	if (bitcnt < 0 || bitcnt > TEGRA_I2S_TIMING_CHANNEL_BIT_COUNT_MASK_US)
-		return -EINVAL;
-	reg = bitcnt << TEGRA_I2S_TIMING_CHANNEL_BIT_COUNT_SHIFT;
-
-	if (i2sclock % (2 * srate))
-		reg |= TEGRA_I2S_TIMING_NON_SYM_ENABLE;
-
-	if (!i2s->clk_refs)
-		clk_enable(i2s->clk_i2s);
-
-	tegra_i2s_write(i2s, TEGRA_I2S_TIMING, reg);
-
-	tegra_i2s_write(i2s, TEGRA_I2S_FIFO_SCR,
-		TEGRA_I2S_FIFO_SCR_FIFO2_ATN_LVL_FOUR_SLOTS |
-		TEGRA_I2S_FIFO_SCR_FIFO1_ATN_LVL_FOUR_SLOTS);
-
-	if (!i2s->clk_refs)
-		clk_disable(i2s->clk_i2s);
-
-	return 0;
-}
-
-static void tegra_i2s_start_playback(struct tegra_i2s *i2s)
-{
-	i2s->reg_ctrl |= TEGRA_I2S_CTRL_FIFO1_ENABLE;
-	tegra_i2s_write(i2s, TEGRA_I2S_CTRL, i2s->reg_ctrl);
-}
-
-static void tegra_i2s_stop_playback(struct tegra_i2s *i2s)
-{
-	i2s->reg_ctrl &= ~TEGRA_I2S_CTRL_FIFO1_ENABLE;
-	tegra_i2s_write(i2s, TEGRA_I2S_CTRL, i2s->reg_ctrl);
-}
-
-static void tegra_i2s_start_capture(struct tegra_i2s *i2s)
-{
-	i2s->reg_ctrl |= TEGRA_I2S_CTRL_FIFO2_ENABLE;
-	tegra_i2s_write(i2s, TEGRA_I2S_CTRL, i2s->reg_ctrl);
-}
-
-static void tegra_i2s_stop_capture(struct tegra_i2s *i2s)
-{
-	i2s->reg_ctrl &= ~TEGRA_I2S_CTRL_FIFO2_ENABLE;
-	tegra_i2s_write(i2s, TEGRA_I2S_CTRL, i2s->reg_ctrl);
-}
-
-static int tegra_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
-				struct snd_soc_dai *dai)
-{
-	struct tegra_i2s *i2s = snd_soc_dai_get_drvdata(dai);
-
-	switch (cmd) {
-	case SNDRV_PCM_TRIGGER_START:
-	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
-	case SNDRV_PCM_TRIGGER_RESUME:
-		if (!i2s->clk_refs)
-			clk_enable(i2s->clk_i2s);
-		i2s->clk_refs++;
-		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
-			tegra_i2s_start_playback(i2s);
-		else
-			tegra_i2s_start_capture(i2s);
-		break;
-	case SNDRV_PCM_TRIGGER_STOP:
-	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
-	case SNDRV_PCM_TRIGGER_SUSPEND:
-		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
-			tegra_i2s_stop_playback(i2s);
-		else
-			tegra_i2s_stop_capture(i2s);
-		i2s->clk_refs--;
-		if (!i2s->clk_refs)
-			clk_disable(i2s->clk_i2s);
-		break;
-	default:
-		return -EINVAL;
-	}
-
-	return 0;
-}
-
-static int tegra_i2s_probe(struct snd_soc_dai *dai)
-{
-	struct tegra_i2s * i2s = snd_soc_dai_get_drvdata(dai);
-
-	dai->capture_dma_data = &i2s->capture_dma_data;
-	dai->playback_dma_data = &i2s->playback_dma_data;
-
-	return 0;
-}
-
-static const struct snd_soc_dai_ops tegra_i2s_dai_ops = {
-	.set_fmt	= tegra_i2s_set_fmt,
-	.hw_params	= tegra_i2s_hw_params,
-	.trigger	= tegra_i2s_trigger,
-};
-
-static const struct snd_soc_dai_driver tegra_i2s_dai_template = {
-	.probe = tegra_i2s_probe,
-	.playback = {
-		.channels_min = 2,
-		.channels_max = 2,
-		.rates = SNDRV_PCM_RATE_8000_96000,
-		.formats = SNDRV_PCM_FMTBIT_S16_LE,
-	},
-	.capture = {
-		.channels_min = 2,
-		.channels_max = 2,
-		.rates = SNDRV_PCM_RATE_8000_96000,
-		.formats = SNDRV_PCM_FMTBIT_S16_LE,
-	},
-	.ops = &tegra_i2s_dai_ops,
-	.symmetric_rates = 1,
-};
-
-static __devinit int tegra_i2s_platform_probe(struct platform_device *pdev)
-{
-	struct tegra_i2s * i2s;
-	struct resource *mem, *memregion, *dmareq;
-	u32 of_dma[2];
-	u32 dma_ch;
-	int ret;
-
-	i2s = devm_kzalloc(&pdev->dev, sizeof(struct tegra_i2s), GFP_KERNEL);
-	if (!i2s) {
-		dev_err(&pdev->dev, "Can't allocate tegra_i2s\n");
-		ret = -ENOMEM;
-		goto err;
-	}
-	dev_set_drvdata(&pdev->dev, i2s);
-
-	i2s->dai = tegra_i2s_dai_template;
-	i2s->dai.name = dev_name(&pdev->dev);
-
-	i2s->clk_i2s = clk_get(&pdev->dev, NULL);
-	if (IS_ERR(i2s->clk_i2s)) {
-		dev_err(&pdev->dev, "Can't retrieve i2s clock\n");
-		ret = PTR_ERR(i2s->clk_i2s);
-		goto err;
-	}
-
-	mem = platform_get_resource(pdev, IORESOURCE_MEM, 0);
-	if (!mem) {
-		dev_err(&pdev->dev, "No memory resource\n");
-		ret = -ENODEV;
-		goto err_clk_put;
-	}
-
-	dmareq = platform_get_resource(pdev, IORESOURCE_DMA, 0);
-	if (!dmareq) {
-		if (of_property_read_u32_array(pdev->dev.of_node,
-					"nvidia,dma-request-selector",
-					of_dma, 2) < 0) {
-			dev_err(&pdev->dev, "No DMA resource\n");
-			ret = -ENODEV;
-			goto err_clk_put;
-		}
-		dma_ch = of_dma[1];
-	} else {
-		dma_ch = dmareq->start;
-	}
-
-	memregion = devm_request_mem_region(&pdev->dev, mem->start,
-					    resource_size(mem), DRV_NAME);
-	if (!memregion) {
-		dev_err(&pdev->dev, "Memory region already claimed\n");
-		ret = -EBUSY;
-		goto err_clk_put;
-	}
-
-	i2s->regs = devm_ioremap(&pdev->dev, mem->start, resource_size(mem));
-	if (!i2s->regs) {
-		dev_err(&pdev->dev, "ioremap failed\n");
-		ret = -ENOMEM;
-		goto err_clk_put;
-	}
-
-	i2s->capture_dma_data.addr = mem->start + TEGRA_I2S_FIFO2;
-	i2s->capture_dma_data.wrap = 4;
-	i2s->capture_dma_data.width = 32;
-	i2s->capture_dma_data.req_sel = dma_ch;
-
-	i2s->playback_dma_data.addr = mem->start + TEGRA_I2S_FIFO1;
-	i2s->playback_dma_data.wrap = 4;
-	i2s->playback_dma_data.width = 32;
-	i2s->playback_dma_data.req_sel = dma_ch;
-
-	i2s->reg_ctrl = TEGRA_I2S_CTRL_FIFO_FORMAT_PACKED;
-
-	ret = snd_soc_register_dai(&pdev->dev, &i2s->dai);
-	if (ret) {
-		dev_err(&pdev->dev, "Could not register DAI: %d\n", ret);
-		ret = -ENOMEM;
-		goto err_clk_put;
-	}
-
-	tegra_i2s_debug_add(i2s);
-
-	return 0;
-
-err_clk_put:
-	clk_put(i2s->clk_i2s);
-err:
-	return ret;
-}
-
-static int __devexit tegra_i2s_platform_remove(struct platform_device *pdev)
-{
-	struct tegra_i2s *i2s = dev_get_drvdata(&pdev->dev);
-
-	snd_soc_unregister_dai(&pdev->dev);
-
-	tegra_i2s_debug_remove(i2s);
-
-	clk_put(i2s->clk_i2s);
-
-	return 0;
-}
-
-static const struct of_device_id tegra_i2s_of_match[] __devinitconst = {
-	{ .compatible = "nvidia,tegra20-i2s", },
-	{},
-};
-
-static struct platform_driver tegra_i2s_driver = {
-	.driver = {
-		.name = DRV_NAME,
-		.owner = THIS_MODULE,
-		.of_match_table = tegra_i2s_of_match,
-	},
-	.probe = tegra_i2s_platform_probe,
-	.remove = __devexit_p(tegra_i2s_platform_remove),
-};
-module_platform_driver(tegra_i2s_driver);
-
-MODULE_AUTHOR("Stephen Warren <swarren@nvidia.com>");
-MODULE_DESCRIPTION("Tegra I2S ASoC driver");
-MODULE_LICENSE("GPL");
-MODULE_ALIAS("platform:" DRV_NAME);
-MODULE_DEVICE_TABLE(of, tegra_i2s_of_match);
diff --git a/sound/soc/tegra/tegra_i2s.h b/sound/soc/tegra/tegra_i2s.h
deleted file mode 100644
index 15ce1e2e8bde..000000000000
--- a/sound/soc/tegra/tegra_i2s.h
+++ /dev/null
@@ -1,166 +0,0 @@
-/*
- * tegra_i2s.h - Definitions for Tegra I2S driver
- *
- * Author: Stephen Warren <swarren@nvidia.com>
- * Copyright (C) 2010 - NVIDIA, Inc.
- *
- * Based on code copyright/by:
- *
- * Copyright (c) 2009-2010, NVIDIA Corporation.
- * Scott Peterson <speterson@nvidia.com>
- *
- * Copyright (C) 2010 Google, Inc.
- * Iliyan Malchev <malchev@google.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 __TEGRA_I2S_H__
-#define __TEGRA_I2S_H__
-
-#include "tegra_pcm.h"
-
-/* Register offsets from TEGRA_I2S1_BASE and TEGRA_I2S2_BASE */
-
-#define TEGRA_I2S_CTRL					0x00
-#define TEGRA_I2S_STATUS				0x04
-#define TEGRA_I2S_TIMING				0x08
-#define TEGRA_I2S_FIFO_SCR				0x0c
-#define TEGRA_I2S_PCM_CTRL				0x10
-#define TEGRA_I2S_NW_CTRL				0x14
-#define TEGRA_I2S_TDM_CTRL				0x20
-#define TEGRA_I2S_TDM_TX_RX_CTRL			0x24
-#define TEGRA_I2S_FIFO1					0x40
-#define TEGRA_I2S_FIFO2					0x80
-
-/* Fields in TEGRA_I2S_CTRL */
-
-#define TEGRA_I2S_CTRL_FIFO2_TX_ENABLE			(1 << 30)
-#define TEGRA_I2S_CTRL_FIFO1_ENABLE			(1 << 29)
-#define TEGRA_I2S_CTRL_FIFO2_ENABLE			(1 << 28)
-#define TEGRA_I2S_CTRL_FIFO1_RX_ENABLE			(1 << 27)
-#define TEGRA_I2S_CTRL_FIFO_LPBK_ENABLE			(1 << 26)
-#define TEGRA_I2S_CTRL_MASTER_ENABLE			(1 << 25)
-
-#define TEGRA_I2S_LRCK_LEFT_LOW				0
-#define TEGRA_I2S_LRCK_RIGHT_LOW			1
-
-#define TEGRA_I2S_CTRL_LRCK_SHIFT			24
-#define TEGRA_I2S_CTRL_LRCK_MASK			(1                        << TEGRA_I2S_CTRL_LRCK_SHIFT)
-#define TEGRA_I2S_CTRL_LRCK_L_LOW			(TEGRA_I2S_LRCK_LEFT_LOW  << TEGRA_I2S_CTRL_LRCK_SHIFT)
-#define TEGRA_I2S_CTRL_LRCK_R_LOW			(TEGRA_I2S_LRCK_RIGHT_LOW << TEGRA_I2S_CTRL_LRCK_SHIFT)
-
-#define TEGRA_I2S_BIT_FORMAT_I2S			0
-#define TEGRA_I2S_BIT_FORMAT_RJM			1
-#define TEGRA_I2S_BIT_FORMAT_LJM			2
-#define TEGRA_I2S_BIT_FORMAT_DSP			3
-
-#define TEGRA_I2S_CTRL_BIT_FORMAT_SHIFT			10
-#define TEGRA_I2S_CTRL_BIT_FORMAT_MASK			(3                        << TEGRA_I2S_CTRL_BIT_FORMAT_SHIFT)
-#define TEGRA_I2S_CTRL_BIT_FORMAT_I2S			(TEGRA_I2S_BIT_FORMAT_I2S << TEGRA_I2S_CTRL_BIT_FORMAT_SHIFT)
-#define TEGRA_I2S_CTRL_BIT_FORMAT_RJM			(TEGRA_I2S_BIT_FORMAT_RJM << TEGRA_I2S_CTRL_BIT_FORMAT_SHIFT)
-#define TEGRA_I2S_CTRL_BIT_FORMAT_LJM			(TEGRA_I2S_BIT_FORMAT_LJM << TEGRA_I2S_CTRL_BIT_FORMAT_SHIFT)
-#define TEGRA_I2S_CTRL_BIT_FORMAT_DSP			(TEGRA_I2S_BIT_FORMAT_DSP << TEGRA_I2S_CTRL_BIT_FORMAT_SHIFT)
-
-#define TEGRA_I2S_BIT_SIZE_16				0
-#define TEGRA_I2S_BIT_SIZE_20				1
-#define TEGRA_I2S_BIT_SIZE_24				2
-#define TEGRA_I2S_BIT_SIZE_32				3
-
-#define TEGRA_I2S_CTRL_BIT_SIZE_SHIFT			8
-#define TEGRA_I2S_CTRL_BIT_SIZE_MASK			(3                     << TEGRA_I2S_CTRL_BIT_SIZE_SHIFT)
-#define TEGRA_I2S_CTRL_BIT_SIZE_16			(TEGRA_I2S_BIT_SIZE_16 << TEGRA_I2S_CTRL_BIT_SIZE_SHIFT)
-#define TEGRA_I2S_CTRL_BIT_SIZE_20			(TEGRA_I2S_BIT_SIZE_20 << TEGRA_I2S_CTRL_BIT_SIZE_SHIFT)
-#define TEGRA_I2S_CTRL_BIT_SIZE_24			(TEGRA_I2S_BIT_SIZE_24 << TEGRA_I2S_CTRL_BIT_SIZE_SHIFT)
-#define TEGRA_I2S_CTRL_BIT_SIZE_32			(TEGRA_I2S_BIT_SIZE_32 << TEGRA_I2S_CTRL_BIT_SIZE_SHIFT)
-
-#define TEGRA_I2S_FIFO_16_LSB				0
-#define TEGRA_I2S_FIFO_20_LSB				1
-#define TEGRA_I2S_FIFO_24_LSB				2
-#define TEGRA_I2S_FIFO_32				3
-#define TEGRA_I2S_FIFO_PACKED				7
-
-#define TEGRA_I2S_CTRL_FIFO_FORMAT_SHIFT		4
-#define TEGRA_I2S_CTRL_FIFO_FORMAT_MASK			(7                     << TEGRA_I2S_CTRL_FIFO_FORMAT_SHIFT)
-#define TEGRA_I2S_CTRL_FIFO_FORMAT_16_LSB		(TEGRA_I2S_FIFO_16_LSB << TEGRA_I2S_CTRL_FIFO_FORMAT_SHIFT)
-#define TEGRA_I2S_CTRL_FIFO_FORMAT_20_LSB		(TEGRA_I2S_FIFO_20_LSB << TEGRA_I2S_CTRL_FIFO_FORMAT_SHIFT)
-#define TEGRA_I2S_CTRL_FIFO_FORMAT_24_LSB		(TEGRA_I2S_FIFO_24_LSB << TEGRA_I2S_CTRL_FIFO_FORMAT_SHIFT)
-#define TEGRA_I2S_CTRL_FIFO_FORMAT_32			(TEGRA_I2S_FIFO_32     << TEGRA_I2S_CTRL_FIFO_FORMAT_SHIFT)
-#define TEGRA_I2S_CTRL_FIFO_FORMAT_PACKED		(TEGRA_I2S_FIFO_PACKED << TEGRA_I2S_CTRL_FIFO_FORMAT_SHIFT)
-
-#define TEGRA_I2S_CTRL_IE_FIFO1_ERR			(1 << 3)
-#define TEGRA_I2S_CTRL_IE_FIFO2_ERR			(1 << 2)
-#define TEGRA_I2S_CTRL_QE_FIFO1				(1 << 1)
-#define TEGRA_I2S_CTRL_QE_FIFO2				(1 << 0)
-
-/* Fields in TEGRA_I2S_STATUS */
-
-#define TEGRA_I2S_STATUS_FIFO1_RDY			(1 << 31)
-#define TEGRA_I2S_STATUS_FIFO2_RDY			(1 << 30)
-#define TEGRA_I2S_STATUS_FIFO1_BSY			(1 << 29)
-#define TEGRA_I2S_STATUS_FIFO2_BSY			(1 << 28)
-#define TEGRA_I2S_STATUS_FIFO1_ERR			(1 << 3)
-#define TEGRA_I2S_STATUS_FIFO2_ERR			(1 << 2)
-#define TEGRA_I2S_STATUS_QS_FIFO1			(1 << 1)
-#define TEGRA_I2S_STATUS_QS_FIFO2			(1 << 0)
-
-/* Fields in TEGRA_I2S_TIMING */
-
-#define TEGRA_I2S_TIMING_NON_SYM_ENABLE			(1 << 12)
-#define TEGRA_I2S_TIMING_CHANNEL_BIT_COUNT_SHIFT	0
-#define TEGRA_I2S_TIMING_CHANNEL_BIT_COUNT_MASK_US	0x7fff
-#define TEGRA_I2S_TIMING_CHANNEL_BIT_COUNT_MASK		(TEGRA_I2S_TIMING_CHANNEL_BIT_COUNT_MASK_US << TEGRA_I2S_TIMING_CHANNEL_BIT_COUNT_SHIFT)
-
-/* Fields in TEGRA_I2S_FIFO_SCR */
-
-#define TEGRA_I2S_FIFO_SCR_FIFO2_FULL_EMPTY_COUNT_SHIFT	24
-#define TEGRA_I2S_FIFO_SCR_FIFO1_FULL_EMPTY_COUNT_SHIFT	16
-#define TEGRA_I2S_FIFO_SCR_FIFO_FULL_EMPTY_COUNT_MASK	0x3f
-
-#define TEGRA_I2S_FIFO_SCR_FIFO2_CLR			(1 << 12)
-#define TEGRA_I2S_FIFO_SCR_FIFO1_CLR			(1 << 8)
-
-#define TEGRA_I2S_FIFO_ATN_LVL_ONE_SLOT			0
-#define TEGRA_I2S_FIFO_ATN_LVL_FOUR_SLOTS		1
-#define TEGRA_I2S_FIFO_ATN_LVL_EIGHT_SLOTS		2
-#define TEGRA_I2S_FIFO_ATN_LVL_TWELVE_SLOTS		3
-
-#define TEGRA_I2S_FIFO_SCR_FIFO2_ATN_LVL_SHIFT		4
-#define TEGRA_I2S_FIFO_SCR_FIFO2_ATN_LVL_MASK		(3 << TEGRA_I2S_FIFO_SCR_FIFO2_ATN_LVL_SHIFT)
-#define TEGRA_I2S_FIFO_SCR_FIFO2_ATN_LVL_ONE_SLOT	(TEGRA_I2S_FIFO_ATN_LVL_ONE_SLOT     << TEGRA_I2S_FIFO_SCR_FIFO2_ATN_LVL_SHIFT)
-#define TEGRA_I2S_FIFO_SCR_FIFO2_ATN_LVL_FOUR_SLOTS	(TEGRA_I2S_FIFO_ATN_LVL_FOUR_SLOTS   << TEGRA_I2S_FIFO_SCR_FIFO2_ATN_LVL_SHIFT)
-#define TEGRA_I2S_FIFO_SCR_FIFO2_ATN_LVL_EIGHT_SLOTS	(TEGRA_I2S_FIFO_ATN_LVL_EIGHT_SLOTS  << TEGRA_I2S_FIFO_SCR_FIFO2_ATN_LVL_SHIFT)
-#define TEGRA_I2S_FIFO_SCR_FIFO2_ATN_LVL_TWELVE_SLOTS	(TEGRA_I2S_FIFO_ATN_LVL_TWELVE_SLOTS << TEGRA_I2S_FIFO_SCR_FIFO2_ATN_LVL_SHIFT)
-
-#define TEGRA_I2S_FIFO_SCR_FIFO1_ATN_LVL_SHIFT		0
-#define TEGRA_I2S_FIFO_SCR_FIFO1_ATN_LVL_MASK		(3 << TEGRA_I2S_FIFO_SCR_FIFO1_ATN_LVL_SHIFT)
-#define TEGRA_I2S_FIFO_SCR_FIFO1_ATN_LVL_ONE_SLOT	(TEGRA_I2S_FIFO_ATN_LVL_ONE_SLOT     << TEGRA_I2S_FIFO_SCR_FIFO1_ATN_LVL_SHIFT)
-#define TEGRA_I2S_FIFO_SCR_FIFO1_ATN_LVL_FOUR_SLOTS	(TEGRA_I2S_FIFO_ATN_LVL_FOUR_SLOTS   << TEGRA_I2S_FIFO_SCR_FIFO1_ATN_LVL_SHIFT)
-#define TEGRA_I2S_FIFO_SCR_FIFO1_ATN_LVL_EIGHT_SLOTS	(TEGRA_I2S_FIFO_ATN_LVL_EIGHT_SLOTS  << TEGRA_I2S_FIFO_SCR_FIFO1_ATN_LVL_SHIFT)
-#define TEGRA_I2S_FIFO_SCR_FIFO1_ATN_LVL_TWELVE_SLOTS	(TEGRA_I2S_FIFO_ATN_LVL_TWELVE_SLOTS << TEGRA_I2S_FIFO_SCR_FIFO1_ATN_LVL_SHIFT)
-
-struct tegra_i2s {
-	struct snd_soc_dai_driver dai;
-	struct clk *clk_i2s;
-	int clk_refs;
-	struct tegra_pcm_dma_params capture_dma_data;
-	struct tegra_pcm_dma_params playback_dma_data;
-	void __iomem *regs;
-	struct dentry *debug;
-	u32 reg_ctrl;
-};
-
-#endif
diff --git a/sound/soc/tegra/tegra_pcm.c b/sound/soc/tegra/tegra_pcm.c
index 8b4457137c7c..127348dc09b1 100644
--- a/sound/soc/tegra/tegra_pcm.c
+++ b/sound/soc/tegra/tegra_pcm.c
@@ -2,7 +2,7 @@
  * tegra_pcm.c - Tegra PCM driver
  *
  * Author: Stephen Warren <swarren@nvidia.com>
- * Copyright (C) 2010 - NVIDIA, Inc.
+ * Copyright (C) 2010,2012 - NVIDIA, Inc.
  *
  * Based on code copyright/by:
  *
@@ -29,8 +29,8 @@
  *
  */
 
-#include <linux/module.h>
 #include <linux/dma-mapping.h>
+#include <linux/module.h>
 #include <linux/slab.h>
 #include <sound/core.h>
 #include <sound/pcm.h>
@@ -39,8 +39,6 @@
 
 #include "tegra_pcm.h"
 
-#define DRV_NAME "tegra-pcm-audio"
-
 static const struct snd_pcm_hardware tegra_pcm_hardware = {
 	.info			= SNDRV_PCM_INFO_MMAP |
 				  SNDRV_PCM_INFO_MMAP_VALID |
@@ -372,28 +370,18 @@ static struct snd_soc_platform_driver tegra_pcm_platform = {
 	.pcm_free	= tegra_pcm_free,
 };
 
-static int __devinit tegra_pcm_platform_probe(struct platform_device *pdev)
+int __devinit tegra_pcm_platform_register(struct device *dev)
 {
-	return snd_soc_register_platform(&pdev->dev, &tegra_pcm_platform);
+	return snd_soc_register_platform(dev, &tegra_pcm_platform);
 }
+EXPORT_SYMBOL_GPL(tegra_pcm_platform_register);
 
-static int __devexit tegra_pcm_platform_remove(struct platform_device *pdev)
+void __devexit tegra_pcm_platform_unregister(struct device *dev)
 {
-	snd_soc_unregister_platform(&pdev->dev);
-	return 0;
+	snd_soc_unregister_platform(dev);
 }
-
-static struct platform_driver tegra_pcm_driver = {
-	.driver = {
-		.name = DRV_NAME,
-		.owner = THIS_MODULE,
-	},
-	.probe = tegra_pcm_platform_probe,
-	.remove = __devexit_p(tegra_pcm_platform_remove),
-};
-module_platform_driver(tegra_pcm_driver);
+EXPORT_SYMBOL_GPL(tegra_pcm_platform_unregister);
 
 MODULE_AUTHOR("Stephen Warren <swarren@nvidia.com>");
 MODULE_DESCRIPTION("Tegra PCM ASoC driver");
 MODULE_LICENSE("GPL");
-MODULE_ALIAS("platform:" DRV_NAME);
diff --git a/sound/soc/tegra/tegra_pcm.h b/sound/soc/tegra/tegra_pcm.h
index dbb90339fe0d..985d418a35e7 100644
--- a/sound/soc/tegra/tegra_pcm.h
+++ b/sound/soc/tegra/tegra_pcm.h
@@ -2,7 +2,7 @@
  * tegra_pcm.h - Definitions for Tegra PCM driver
  *
  * Author: Stephen Warren <swarren@nvidia.com>
- * Copyright (C) 2010 - NVIDIA, Inc.
+ * Copyright (C) 2010,2012 - NVIDIA, Inc.
  *
  * Based on code copyright/by:
  *
@@ -52,4 +52,7 @@ struct tegra_runtime_data {
 	struct tegra_dma_channel *dma_chan;
 };
 
+int tegra_pcm_platform_register(struct device *dev);
+void tegra_pcm_platform_unregister(struct device *dev);
+
 #endif
diff --git a/sound/soc/tegra/tegra_spdif.c b/sound/soc/tegra/tegra_spdif.c
deleted file mode 100644
index 9ff2c601445f..000000000000
--- a/sound/soc/tegra/tegra_spdif.c
+++ /dev/null
@@ -1,364 +0,0 @@
-/*
- * tegra_spdif.c - Tegra SPDIF driver
- *
- * Author: Stephen Warren <swarren@nvidia.com>
- * Copyright (C) 2011 - NVIDIA, Inc.
- *
- * 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
- *
- */
-
-#include <linux/clk.h>
-#include <linux/module.h>
-#include <linux/debugfs.h>
-#include <linux/device.h>
-#include <linux/platform_device.h>
-#include <linux/seq_file.h>
-#include <linux/slab.h>
-#include <linux/io.h>
-#include <mach/iomap.h>
-#include <sound/core.h>
-#include <sound/pcm.h>
-#include <sound/pcm_params.h>
-#include <sound/soc.h>
-
-#include "tegra_spdif.h"
-
-#define DRV_NAME "tegra-spdif"
-
-static inline void tegra_spdif_write(struct tegra_spdif *spdif, u32 reg,
-					u32 val)
-{
-	__raw_writel(val, spdif->regs + reg);
-}
-
-static inline u32 tegra_spdif_read(struct tegra_spdif *spdif, u32 reg)
-{
-	return __raw_readl(spdif->regs + reg);
-}
-
-#ifdef CONFIG_DEBUG_FS
-static int tegra_spdif_show(struct seq_file *s, void *unused)
-{
-#define REG(r) { r, #r }
-	static const struct {
-		int offset;
-		const char *name;
-	} regs[] = {
-		REG(TEGRA_SPDIF_CTRL),
-		REG(TEGRA_SPDIF_STATUS),
-		REG(TEGRA_SPDIF_STROBE_CTRL),
-		REG(TEGRA_SPDIF_DATA_FIFO_CSR),
-		REG(TEGRA_SPDIF_CH_STA_RX_A),
-		REG(TEGRA_SPDIF_CH_STA_RX_B),
-		REG(TEGRA_SPDIF_CH_STA_RX_C),
-		REG(TEGRA_SPDIF_CH_STA_RX_D),
-		REG(TEGRA_SPDIF_CH_STA_RX_E),
-		REG(TEGRA_SPDIF_CH_STA_RX_F),
-		REG(TEGRA_SPDIF_CH_STA_TX_A),
-		REG(TEGRA_SPDIF_CH_STA_TX_B),
-		REG(TEGRA_SPDIF_CH_STA_TX_C),
-		REG(TEGRA_SPDIF_CH_STA_TX_D),
-		REG(TEGRA_SPDIF_CH_STA_TX_E),
-		REG(TEGRA_SPDIF_CH_STA_TX_F),
-	};
-#undef REG
-
-	struct tegra_spdif *spdif = s->private;
-	int i;
-
-	clk_enable(spdif->clk_spdif_out);
-
-	for (i = 0; i < ARRAY_SIZE(regs); i++) {
-		u32 val = tegra_spdif_read(spdif, regs[i].offset);
-		seq_printf(s, "%s = %08x\n", regs[i].name, val);
-	}
-
-	clk_disable(spdif->clk_spdif_out);
-
-	return 0;
-}
-
-static int tegra_spdif_debug_open(struct inode *inode, struct file *file)
-{
-	return single_open(file, tegra_spdif_show, inode->i_private);
-}
-
-static const struct file_operations tegra_spdif_debug_fops = {
-	.open    = tegra_spdif_debug_open,
-	.read    = seq_read,
-	.llseek  = seq_lseek,
-	.release = single_release,
-};
-
-static void tegra_spdif_debug_add(struct tegra_spdif *spdif)
-{
-	spdif->debug = debugfs_create_file(DRV_NAME, S_IRUGO,
-						snd_soc_debugfs_root, spdif,
-						&tegra_spdif_debug_fops);
-}
-
-static void tegra_spdif_debug_remove(struct tegra_spdif *spdif)
-{
-	if (spdif->debug)
-		debugfs_remove(spdif->debug);
-}
-#else
-static inline void tegra_spdif_debug_add(struct tegra_spdif *spdif)
-{
-}
-
-static inline void tegra_spdif_debug_remove(struct tegra_spdif *spdif)
-{
-}
-#endif
-
-static int tegra_spdif_hw_params(struct snd_pcm_substream *substream,
-				struct snd_pcm_hw_params *params,
-				struct snd_soc_dai *dai)
-{
-	struct device *dev = substream->pcm->card->dev;
-	struct tegra_spdif *spdif = snd_soc_dai_get_drvdata(dai);
-	int ret, spdifclock;
-
-	spdif->reg_ctrl &= ~TEGRA_SPDIF_CTRL_PACK;
-	spdif->reg_ctrl &= ~TEGRA_SPDIF_CTRL_BIT_MODE_MASK;
-	switch (params_format(params)) {
-	case SNDRV_PCM_FORMAT_S16_LE:
-		spdif->reg_ctrl |= TEGRA_SPDIF_CTRL_PACK;
-		spdif->reg_ctrl |= TEGRA_SPDIF_CTRL_BIT_MODE_16BIT;
-		break;
-	default:
-		return -EINVAL;
-	}
-
-	switch (params_rate(params)) {
-	case 32000:
-		spdifclock = 4096000;
-		break;
-	case 44100:
-		spdifclock = 5644800;
-		break;
-	case 48000:
-		spdifclock = 6144000;
-		break;
-	case 88200:
-		spdifclock = 11289600;
-		break;
-	case 96000:
-		spdifclock = 12288000;
-		break;
-	case 176400:
-		spdifclock = 22579200;
-		break;
-	case 192000:
-		spdifclock = 24576000;
-		break;
-	default:
-		return -EINVAL;
-	}
-
-	ret = clk_set_rate(spdif->clk_spdif_out, spdifclock);
-	if (ret) {
-		dev_err(dev, "Can't set SPDIF clock rate: %d\n", ret);
-		return ret;
-	}
-
-	return 0;
-}
-
-static void tegra_spdif_start_playback(struct tegra_spdif *spdif)
-{
-	spdif->reg_ctrl |= TEGRA_SPDIF_CTRL_TX_EN;
-	tegra_spdif_write(spdif, TEGRA_SPDIF_CTRL, spdif->reg_ctrl);
-}
-
-static void tegra_spdif_stop_playback(struct tegra_spdif *spdif)
-{
-	spdif->reg_ctrl &= ~TEGRA_SPDIF_CTRL_TX_EN;
-	tegra_spdif_write(spdif, TEGRA_SPDIF_CTRL, spdif->reg_ctrl);
-}
-
-static int tegra_spdif_trigger(struct snd_pcm_substream *substream, int cmd,
-				struct snd_soc_dai *dai)
-{
-	struct tegra_spdif *spdif = snd_soc_dai_get_drvdata(dai);
-
-	switch (cmd) {
-	case SNDRV_PCM_TRIGGER_START:
-	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
-	case SNDRV_PCM_TRIGGER_RESUME:
-		if (!spdif->clk_refs)
-			clk_enable(spdif->clk_spdif_out);
-		spdif->clk_refs++;
-		tegra_spdif_start_playback(spdif);
-		break;
-	case SNDRV_PCM_TRIGGER_STOP:
-	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
-	case SNDRV_PCM_TRIGGER_SUSPEND:
-		tegra_spdif_stop_playback(spdif);
-		spdif->clk_refs--;
-		if (!spdif->clk_refs)
-			clk_disable(spdif->clk_spdif_out);
-		break;
-	default:
-		return -EINVAL;
-	}
-
-	return 0;
-}
-
-static int tegra_spdif_probe(struct snd_soc_dai *dai)
-{
-	struct tegra_spdif *spdif = snd_soc_dai_get_drvdata(dai);
-
-	dai->capture_dma_data = NULL;
-	dai->playback_dma_data = &spdif->playback_dma_data;
-
-	return 0;
-}
-
-static const struct snd_soc_dai_ops tegra_spdif_dai_ops = {
-	.hw_params	= tegra_spdif_hw_params,
-	.trigger	= tegra_spdif_trigger,
-};
-
-static struct snd_soc_dai_driver tegra_spdif_dai = {
-	.name = DRV_NAME,
-	.probe = tegra_spdif_probe,
-	.playback = {
-		.channels_min = 2,
-		.channels_max = 2,
-		.rates = SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |
-				SNDRV_PCM_RATE_48000,
-		.formats = SNDRV_PCM_FMTBIT_S16_LE,
-	},
-	.ops = &tegra_spdif_dai_ops,
-};
-
-static __devinit int tegra_spdif_platform_probe(struct platform_device *pdev)
-{
-	struct tegra_spdif *spdif;
-	struct resource *mem, *memregion, *dmareq;
-	int ret;
-
-	spdif = kzalloc(sizeof(struct tegra_spdif), GFP_KERNEL);
-	if (!spdif) {
-		dev_err(&pdev->dev, "Can't allocate tegra_spdif\n");
-		ret = -ENOMEM;
-		goto exit;
-	}
-	dev_set_drvdata(&pdev->dev, spdif);
-
-	spdif->clk_spdif_out = clk_get(&pdev->dev, "spdif_out");
-	if (IS_ERR(spdif->clk_spdif_out)) {
-		pr_err("Can't retrieve spdif clock\n");
-		ret = PTR_ERR(spdif->clk_spdif_out);
-		goto err_free;
-	}
-
-	mem = platform_get_resource(pdev, IORESOURCE_MEM, 0);
-	if (!mem) {
-		dev_err(&pdev->dev, "No memory resource\n");
-		ret = -ENODEV;
-		goto err_clk_put;
-	}
-
-	dmareq = platform_get_resource(pdev, IORESOURCE_DMA, 0);
-	if (!dmareq) {
-		dev_err(&pdev->dev, "No DMA resource\n");
-		ret = -ENODEV;
-		goto err_clk_put;
-	}
-
-	memregion = request_mem_region(mem->start, resource_size(mem),
-					DRV_NAME);
-	if (!memregion) {
-		dev_err(&pdev->dev, "Memory region already claimed\n");
-		ret = -EBUSY;
-		goto err_clk_put;
-	}
-
-	spdif->regs = ioremap(mem->start, resource_size(mem));
-	if (!spdif->regs) {
-		dev_err(&pdev->dev, "ioremap failed\n");
-		ret = -ENOMEM;
-		goto err_release;
-	}
-
-	spdif->playback_dma_data.addr = mem->start + TEGRA_SPDIF_DATA_OUT;
-	spdif->playback_dma_data.wrap = 4;
-	spdif->playback_dma_data.width = 32;
-	spdif->playback_dma_data.req_sel = dmareq->start;
-
-	ret = snd_soc_register_dai(&pdev->dev, &tegra_spdif_dai);
-	if (ret) {
-		dev_err(&pdev->dev, "Could not register DAI: %d\n", ret);
-		ret = -ENOMEM;
-		goto err_unmap;
-	}
-
-	tegra_spdif_debug_add(spdif);
-
-	return 0;
-
-err_unmap:
-	iounmap(spdif->regs);
-err_release:
-	release_mem_region(mem->start, resource_size(mem));
-err_clk_put:
-	clk_put(spdif->clk_spdif_out);
-err_free:
-	kfree(spdif);
-exit:
-	return ret;
-}
-
-static int __devexit tegra_spdif_platform_remove(struct platform_device *pdev)
-{
-	struct tegra_spdif *spdif = dev_get_drvdata(&pdev->dev);
-	struct resource *res;
-
-	snd_soc_unregister_dai(&pdev->dev);
-
-	tegra_spdif_debug_remove(spdif);
-
-	iounmap(spdif->regs);
-
-	res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
-	release_mem_region(res->start, resource_size(res));
-
-	clk_put(spdif->clk_spdif_out);
-
-	kfree(spdif);
-
-	return 0;
-}
-
-static struct platform_driver tegra_spdif_driver = {
-	.driver = {
-		.name = DRV_NAME,
-		.owner = THIS_MODULE,
-	},
-	.probe = tegra_spdif_platform_probe,
-	.remove = __devexit_p(tegra_spdif_platform_remove),
-};
-
-module_platform_driver(tegra_spdif_driver);
-
-MODULE_AUTHOR("Stephen Warren <swarren@nvidia.com>");
-MODULE_DESCRIPTION("Tegra SPDIF ASoC driver");
-MODULE_LICENSE("GPL");
-MODULE_ALIAS("platform:" DRV_NAME);
diff --git a/sound/soc/tegra/tegra_spdif.h b/sound/soc/tegra/tegra_spdif.h
deleted file mode 100644
index 2e03db430279..000000000000
--- a/sound/soc/tegra/tegra_spdif.h
+++ /dev/null
@@ -1,473 +0,0 @@
-/*
- * tegra_spdif.h - Definitions for Tegra SPDIF driver
- *
- * Author: Stephen Warren <swarren@nvidia.com>
- * Copyright (C) 2011 - NVIDIA, Inc.
- *
- * Based on code copyright/by:
- * Copyright (c) 2008-2009, NVIDIA Corporation
- *
- * 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 __TEGRA_SPDIF_H__
-#define __TEGRA_SPDIF_H__
-
-#include "tegra_pcm.h"
-
-/* Offsets from TEGRA_SPDIF_BASE */
-
-#define TEGRA_SPDIF_CTRL					0x0
-#define TEGRA_SPDIF_STATUS					0x4
-#define TEGRA_SPDIF_STROBE_CTRL					0x8
-#define TEGRA_SPDIF_DATA_FIFO_CSR				0x0C
-#define TEGRA_SPDIF_DATA_OUT					0x40
-#define TEGRA_SPDIF_DATA_IN					0x80
-#define TEGRA_SPDIF_CH_STA_RX_A					0x100
-#define TEGRA_SPDIF_CH_STA_RX_B					0x104
-#define TEGRA_SPDIF_CH_STA_RX_C					0x108
-#define TEGRA_SPDIF_CH_STA_RX_D					0x10C
-#define TEGRA_SPDIF_CH_STA_RX_E					0x110
-#define TEGRA_SPDIF_CH_STA_RX_F					0x114
-#define TEGRA_SPDIF_CH_STA_TX_A					0x140
-#define TEGRA_SPDIF_CH_STA_TX_B					0x144
-#define TEGRA_SPDIF_CH_STA_TX_C					0x148
-#define TEGRA_SPDIF_CH_STA_TX_D					0x14C
-#define TEGRA_SPDIF_CH_STA_TX_E					0x150
-#define TEGRA_SPDIF_CH_STA_TX_F					0x154
-#define TEGRA_SPDIF_USR_STA_RX_A				0x180
-#define TEGRA_SPDIF_USR_DAT_TX_A				0x1C0
-
-/* Fields in TEGRA_SPDIF_CTRL */
-
-/* Start capturing from 0=right, 1=left channel */
-#define TEGRA_SPDIF_CTRL_CAP_LC					(1 << 30)
-
-/* SPDIF receiver(RX) enable */
-#define TEGRA_SPDIF_CTRL_RX_EN					(1 << 29)
-
-/* SPDIF Transmitter(TX) enable */
-#define TEGRA_SPDIF_CTRL_TX_EN					(1 << 28)
-
-/* Transmit Channel status */
-#define TEGRA_SPDIF_CTRL_TC_EN					(1 << 27)
-
-/* Transmit user Data */
-#define TEGRA_SPDIF_CTRL_TU_EN					(1 << 26)
-
-/* Interrupt on transmit error */
-#define TEGRA_SPDIF_CTRL_IE_TXE					(1 << 25)
-
-/* Interrupt on receive error */
-#define TEGRA_SPDIF_CTRL_IE_RXE					(1 << 24)
-
-/* Interrupt on invalid preamble */
-#define TEGRA_SPDIF_CTRL_IE_P					(1 << 23)
-
-/* Interrupt on "B" preamble */
-#define TEGRA_SPDIF_CTRL_IE_B					(1 << 22)
-
-/* Interrupt when block of channel status received */
-#define TEGRA_SPDIF_CTRL_IE_C					(1 << 21)
-
-/* Interrupt when a valid information unit (IU) is received */
-#define TEGRA_SPDIF_CTRL_IE_U					(1 << 20)
-
-/* Interrupt when RX user FIFO attention level is reached */
-#define TEGRA_SPDIF_CTRL_QE_RU					(1 << 19)
-
-/* Interrupt when TX user FIFO attention level is reached */
-#define TEGRA_SPDIF_CTRL_QE_TU					(1 << 18)
-
-/* Interrupt when RX data FIFO attention level is reached */
-#define TEGRA_SPDIF_CTRL_QE_RX					(1 << 17)
-
-/* Interrupt when TX data FIFO attention level is reached */
-#define TEGRA_SPDIF_CTRL_QE_TX					(1 << 16)
-
-/* Loopback test mode enable */
-#define TEGRA_SPDIF_CTRL_LBK_EN					(1 << 15)
-
-/*
- * Pack data mode:
- * 0 = Single data (16 bit needs to be  padded to match the
- *     interface data bit size).
- * 1 = Packeted left/right channel data into a single word.
- */
-#define TEGRA_SPDIF_CTRL_PACK					(1 << 14)
-
-/*
- * 00 = 16bit data
- * 01 = 20bit data
- * 10 = 24bit data
- * 11 = raw data
- */
-#define TEGRA_SPDIF_BIT_MODE_16BIT				0
-#define TEGRA_SPDIF_BIT_MODE_20BIT				1
-#define TEGRA_SPDIF_BIT_MODE_24BIT				2
-#define TEGRA_SPDIF_BIT_MODE_RAW				3
-
-#define TEGRA_SPDIF_CTRL_BIT_MODE_SHIFT				12
-#define TEGRA_SPDIF_CTRL_BIT_MODE_MASK				(3                          << TEGRA_SPDIF_CTRL_BIT_MODE_SHIFT)
-#define TEGRA_SPDIF_CTRL_BIT_MODE_16BIT				(TEGRA_SPDIF_BIT_MODE_16BIT << TEGRA_SPDIF_CTRL_BIT_MODE_SHIFT)
-#define TEGRA_SPDIF_CTRL_BIT_MODE_20BIT				(TEGRA_SPDIF_BIT_MODE_20BIT << TEGRA_SPDIF_CTRL_BIT_MODE_SHIFT)
-#define TEGRA_SPDIF_CTRL_BIT_MODE_24BIT				(TEGRA_SPDIF_BIT_MODE_24BIT << TEGRA_SPDIF_CTRL_BIT_MODE_SHIFT)
-#define TEGRA_SPDIF_CTRL_BIT_MODE_RAW				(TEGRA_SPDIF_BIT_MODE_RAW   << TEGRA_SPDIF_CTRL_BIT_MODE_SHIFT)
-
-/* Fields in TEGRA_SPDIF_STATUS */
-
-/*
- * Note: IS_P, IS_B, IS_C, and IS_U are sticky bits. Software must
- * write a 1 to the corresponding bit location to clear the status.
- */
-
-/*
- * Receiver(RX) shifter is busy receiving data.
- * This bit is asserted when the receiver first locked onto the
- * preamble of the data stream after RX_EN is asserted. This bit is
- * deasserted when either,
- * (a) the end of a frame is reached after RX_EN is deeasserted, or
- * (b) the SPDIF data stream becomes inactive.
- */
-#define TEGRA_SPDIF_STATUS_RX_BSY				(1 << 29)
-
-/*
- * Transmitter(TX) shifter is busy transmitting data.
- * This bit is asserted when TX_EN is asserted.
- * This bit is deasserted when the end of a frame is reached after
- * TX_EN is deasserted.
- */
-#define TEGRA_SPDIF_STATUS_TX_BSY				(1 << 28)
-
-/*
- * TX is busy shifting out channel status.
- * This bit is asserted when both TX_EN and TC_EN are asserted and
- * data from CH_STA_TX_A register is loaded into the internal shifter.
- * This bit is deasserted when either,
- * (a) the end of a frame is reached after TX_EN is deasserted, or
- * (b) CH_STA_TX_F register is loaded into the internal shifter.
- */
-#define TEGRA_SPDIF_STATUS_TC_BSY				(1 << 27)
-
-/*
- * TX User data FIFO busy.
- * This bit is asserted when TX_EN and TXU_EN are asserted and
- * there's data in the TX user FIFO.  This bit is deassert when either,
- * (a) the end of a frame is reached after TX_EN is deasserted, or
- * (b) there's no data left in the TX user FIFO.
- */
-#define TEGRA_SPDIF_STATUS_TU_BSY				(1 << 26)
-
-/* TX FIFO Underrun error status */
-#define TEGRA_SPDIF_STATUS_TX_ERR				(1 << 25)
-
-/* RX FIFO Overrun error status */
-#define TEGRA_SPDIF_STATUS_RX_ERR				(1 << 24)
-
-/* Preamble status: 0=Preamble OK, 1=bad/missing preamble */
-#define TEGRA_SPDIF_STATUS_IS_P					(1 << 23)
-
-/* B-preamble detection status: 0=not detected, 1=B-preamble detected */
-#define TEGRA_SPDIF_STATUS_IS_B					(1 << 22)
-
-/*
- * RX channel block data receive status:
- * 0=entire block not recieved yet.
- * 1=received entire block of channel status,
- */
-#define TEGRA_SPDIF_STATUS_IS_C					(1 << 21)
-
-/* RX User Data Valid flag:  1=valid IU detected, 0 = no IU detected. */
-#define TEGRA_SPDIF_STATUS_IS_U					(1 << 20)
-
-/*
- * RX User FIFO Status:
- * 1=attention level reached, 0=attention level not reached.
- */
-#define TEGRA_SPDIF_STATUS_QS_RU				(1 << 19)
-
-/*
- * TX User FIFO Status:
- * 1=attention level reached, 0=attention level not reached.
- */
-#define TEGRA_SPDIF_STATUS_QS_TU				(1 << 18)
-
-/*
- * RX Data FIFO Status:
- * 1=attention level reached, 0=attention level not reached.
- */
-#define TEGRA_SPDIF_STATUS_QS_RX				(1 << 17)
-
-/*
- * TX Data FIFO Status:
- * 1=attention level reached, 0=attention level not reached.
- */
-#define TEGRA_SPDIF_STATUS_QS_TX				(1 << 16)
-
-/* Fields in TEGRA_SPDIF_STROBE_CTRL */
-
-/*
- * Indicates the approximate number of detected SPDIFIN clocks within a
- * bi-phase period.
- */
-#define TEGRA_SPDIF_STROBE_CTRL_PERIOD_SHIFT			16
-#define TEGRA_SPDIF_STROBE_CTRL_PERIOD_MASK			(0xff << TEGRA_SPDIF_STROBE_CTRL_PERIOD_SHIFT)
-
-/* Data strobe mode: 0=Auto-locked 1=Manual locked */
-#define TEGRA_SPDIF_STROBE_CTRL_STROBE				(1 << 15)
-
-/*
- * Manual data strobe time within the bi-phase clock period (in terms of
- * the number of over-sampling clocks).
- */
-#define TEGRA_SPDIF_STROBE_CTRL_DATA_STROBES_SHIFT		8
-#define TEGRA_SPDIF_STROBE_CTRL_DATA_STROBES_MASK		(0x1f << TEGRA_SPDIF_STROBE_CTRL_DATA_STROBES_SHIFT)
-
-/*
- * Manual SPDIFIN bi-phase clock period (in terms of the number of
- * over-sampling clocks).
- */
-#define TEGRA_SPDIF_STROBE_CTRL_CLOCK_PERIOD_SHIFT		0
-#define TEGRA_SPDIF_STROBE_CTRL_CLOCK_PERIOD_MASK		(0x3f << TEGRA_SPDIF_STROBE_CTRL_CLOCK_PERIOD_SHIFT)
-
-/* Fields in SPDIF_DATA_FIFO_CSR */
-
-/* Clear Receiver User FIFO (RX USR.FIFO) */
-#define TEGRA_SPDIF_DATA_FIFO_CSR_RU_CLR			(1 << 31)
-
-#define TEGRA_SPDIF_FIFO_ATN_LVL_U_ONE_SLOT			0
-#define TEGRA_SPDIF_FIFO_ATN_LVL_U_TWO_SLOTS			1
-#define TEGRA_SPDIF_FIFO_ATN_LVL_U_THREE_SLOTS			2
-#define TEGRA_SPDIF_FIFO_ATN_LVL_U_FOUR_SLOTS			3
-
-/* RU FIFO attention level */
-#define TEGRA_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_SHIFT		29
-#define TEGRA_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_MASK		\
-		(0x3                                    << TEGRA_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_SHIFT)
-#define TEGRA_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_RU1_WORD_FULL	\
-		(TEGRA_SPDIF_FIFO_ATN_LVL_U_ONE_SLOT    << TEGRA_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_SHIFT)
-#define TEGRA_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_RU2_WORD_FULL	\
-		(TEGRA_SPDIF_FIFO_ATN_LVL_U_TWO_SLOTS   << TEGRA_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_SHIFT)
-#define TEGRA_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_RU3_WORD_FULL	\
-		(TEGRA_SPDIF_FIFO_ATN_LVL_U_THREE_SLOTS << TEGRA_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_SHIFT)
-#define TEGRA_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_RU4_WORD_FULL	\
-		(TEGRA_SPDIF_FIFO_ATN_LVL_U_FOUR_SLOTS  << TEGRA_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_SHIFT)
-
-/* Number of RX USR.FIFO levels with valid data. */
-#define TEGRA_SPDIF_DATA_FIFO_CSR_RU_FULL_COUNT_SHIFT		24
-#define TEGRA_SPDIF_DATA_FIFO_CSR_RU_FULL_COUNT_MASK		(0x1f << TEGRA_SPDIF_DATA_FIFO_CSR_RU_FULL_COUNT_SHIFT)
-
-/* Clear Transmitter User FIFO (TX USR.FIFO) */
-#define TEGRA_SPDIF_DATA_FIFO_CSR_TU_CLR			(1 << 23)
-
-/* TU FIFO attention level */
-#define TEGRA_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_SHIFT		21
-#define TEGRA_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_MASK		\
-		(0x3                                   << TEGRA_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_SHIFT)
-#define TEGRA_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_TU1_WORD_FULL	\
-		(TEGRA_SPDIF_FIFO_ATN_LVL_U_ONE_SLOT    << TEGRA_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_SHIFT)
-#define TEGRA_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_TU2_WORD_FULL	\
-		(TEGRA_SPDIF_FIFO_ATN_LVL_U_TWO_SLOTS   << TEGRA_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_SHIFT)
-#define TEGRA_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_TU3_WORD_FULL	\
-		(TEGRA_SPDIF_FIFO_ATN_LVL_U_THREE_SLOTS << TEGRA_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_SHIFT)
-#define TEGRA_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_TU4_WORD_FULL	\
-		(TEGRA_SPDIF_FIFO_ATN_LVL_U_FOUR_SLOTS  << TEGRA_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_SHIFT)
-
-/* Number of TX USR.FIFO levels that could be filled. */
-#define TEGRA_SPDIF_DATA_FIFO_CSR_TU_EMPTY_COUNT_SHIFT		16
-#define TEGRA_SPDIF_DATA_FIFO_CSR_TU_EMPTY_COUNT_MASK		(0x1f << SPDIF_DATA_FIFO_CSR_TU_EMPTY_COUNT_SHIFT)
-
-/* Clear Receiver Data FIFO (RX DATA.FIFO) */
-#define TEGRA_SPDIF_DATA_FIFO_CSR_RX_CLR			(1 << 15)
-
-#define TEGRA_SPDIF_FIFO_ATN_LVL_D_ONE_SLOT			0
-#define TEGRA_SPDIF_FIFO_ATN_LVL_D_FOUR_SLOTS			1
-#define TEGRA_SPDIF_FIFO_ATN_LVL_D_EIGHT_SLOTS			2
-#define TEGRA_SPDIF_FIFO_ATN_LVL_D_TWELVE_SLOTS			3
-
-/* RU FIFO attention level */
-#define TEGRA_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_SHIFT		13
-#define TEGRA_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_MASK		\
-		(0x3                                     << TEGRA_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_SHIFT)
-#define TEGRA_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_RU1_WORD_FULL	\
-		(TEGRA_SPDIF_FIFO_ATN_LVL_D_ONE_SLOT     << TEGRA_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_SHIFT)
-#define TEGRA_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_RU4_WORD_FULL	\
-		(TEGRA_SPDIF_FIFO_ATN_LVL_D_FOUR_SLOTS   << TEGRA_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_SHIFT)
-#define TEGRA_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_RU8_WORD_FULL	\
-		(TEGRA_SPDIF_FIFO_ATN_LVL_D_EIGHT_SLOTS  << TEGRA_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_SHIFT)
-#define TEGRA_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_RU12_WORD_FULL	\
-		(TEGRA_SPDIF_FIFO_ATN_LVL_D_TWELVE_SLOTS << TEGRA_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_SHIFT)
-
-/* Number of RX DATA.FIFO levels with valid data. */
-#define TEGRA_SPDIF_DATA_FIFO_CSR_RX_FULL_COUNT_SHIFT		8
-#define TEGRA_SPDIF_DATA_FIFO_CSR_RX_FULL_COUNT_MASK		(0x1f << TEGRA_SPDIF_DATA_FIFO_CSR_RX_FULL_COUNT_SHIFT)
-
-/* Clear Transmitter Data FIFO (TX DATA.FIFO) */
-#define TEGRA_SPDIF_DATA_FIFO_CSR_TX_CLR			(1 << 7)
-
-/* TU FIFO attention level */
-#define TEGRA_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_SHIFT		5
-#define TEGRA_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_MASK		\
-		(0x3                                     << TEGRA_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_SHIFT)
-#define TEGRA_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_TU1_WORD_FULL	\
-		(TEGRA_SPDIF_FIFO_ATN_LVL_D_ONE_SLOT     << TEGRA_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_SHIFT)
-#define TEGRA_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_TU4_WORD_FULL	\
-		(TEGRA_SPDIF_FIFO_ATN_LVL_D_FOUR_SLOTS   << TEGRA_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_SHIFT)
-#define TEGRA_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_TU8_WORD_FULL	\
-		(TEGRA_SPDIF_FIFO_ATN_LVL_D_EIGHT_SLOTS  << TEGRA_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_SHIFT)
-#define TEGRA_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_TU12_WORD_FULL	\
-		(TEGRA_SPDIF_FIFO_ATN_LVL_D_TWELVE_SLOTS << TEGRA_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_SHIFT)
-
-/* Number of TX DATA.FIFO levels that could be filled. */
-#define TEGRA_SPDIF_DATA_FIFO_CSR_TX_EMPTY_COUNT_SHIFT		0
-#define TEGRA_SPDIF_DATA_FIFO_CSR_TX_EMPTY_COUNT_MASK		(0x1f << SPDIF_DATA_FIFO_CSR_TX_EMPTY_COUNT_SHIFT)
-
-/* Fields in TEGRA_SPDIF_DATA_OUT */
-
-/*
- * This register has 5 different formats:
- * 16-bit        (BIT_MODE=00, PACK=0)
- * 20-bit        (BIT_MODE=01, PACK=0)
- * 24-bit        (BIT_MODE=10, PACK=0)
- * raw           (BIT_MODE=11, PACK=0)
- * 16-bit packed (BIT_MODE=00, PACK=1)
- */
-
-#define TEGRA_SPDIF_DATA_OUT_DATA_16_SHIFT			0
-#define TEGRA_SPDIF_DATA_OUT_DATA_16_MASK			(0xffff << TEGRA_SPDIF_DATA_OUT_DATA_16_SHIFT)
-
-#define TEGRA_SPDIF_DATA_OUT_DATA_20_SHIFT			0
-#define TEGRA_SPDIF_DATA_OUT_DATA_20_MASK			(0xfffff << TEGRA_SPDIF_DATA_OUT_DATA_20_SHIFT)
-
-#define TEGRA_SPDIF_DATA_OUT_DATA_24_SHIFT			0
-#define TEGRA_SPDIF_DATA_OUT_DATA_24_MASK			(0xffffff << TEGRA_SPDIF_DATA_OUT_DATA_24_SHIFT)
-
-#define TEGRA_SPDIF_DATA_OUT_DATA_RAW_P				(1 << 31)
-#define TEGRA_SPDIF_DATA_OUT_DATA_RAW_C				(1 << 30)
-#define TEGRA_SPDIF_DATA_OUT_DATA_RAW_U				(1 << 29)
-#define TEGRA_SPDIF_DATA_OUT_DATA_RAW_V				(1 << 28)
-
-#define TEGRA_SPDIF_DATA_OUT_DATA_RAW_DATA_SHIFT		8
-#define TEGRA_SPDIF_DATA_OUT_DATA_RAW_DATA_MASK			(0xfffff << TEGRA_SPDIF_DATA_OUT_DATA_RAW_DATA_SHIFT)
-
-#define TEGRA_SPDIF_DATA_OUT_DATA_RAW_AUX_SHIFT			4
-#define TEGRA_SPDIF_DATA_OUT_DATA_RAW_AUX_MASK			(0xf << TEGRA_SPDIF_DATA_OUT_DATA_RAW_AUX_SHIFT)
-
-#define TEGRA_SPDIF_DATA_OUT_DATA_RAW_PREAMBLE_SHIFT		0
-#define TEGRA_SPDIF_DATA_OUT_DATA_RAW_PREAMBLE_MASK		(0xf << TEGRA_SPDIF_DATA_OUT_DATA_RAW_PREAMBLE_SHIFT)
-
-#define TEGRA_SPDIF_DATA_OUT_DATA_16_PACKED_RIGHT_SHIFT		16
-#define TEGRA_SPDIF_DATA_OUT_DATA_16_PACKED_RIGHT_MASK		(0xffff << TEGRA_SPDIF_DATA_OUT_DATA_16_PACKED_RIGHT_SHIFT)
-
-#define TEGRA_SPDIF_DATA_OUT_DATA_16_PACKED_LEFT_SHIFT		0
-#define TEGRA_SPDIF_DATA_OUT_DATA_16_PACKED_LEFT_MASK		(0xffff << TEGRA_SPDIF_DATA_OUT_DATA_16_PACKED_LEFT_SHIFT)
-
-/* Fields in TEGRA_SPDIF_DATA_IN */
-
-/*
- * This register has 5 different formats:
- * 16-bit        (BIT_MODE=00, PACK=0)
- * 20-bit        (BIT_MODE=01, PACK=0)
- * 24-bit        (BIT_MODE=10, PACK=0)
- * raw           (BIT_MODE=11, PACK=0)
- * 16-bit packed (BIT_MODE=00, PACK=1)
- *
- * Bits 31:24 are common to all modes except 16-bit packed
- */
-
-#define TEGRA_SPDIF_DATA_IN_DATA_P				(1 << 31)
-#define TEGRA_SPDIF_DATA_IN_DATA_C				(1 << 30)
-#define TEGRA_SPDIF_DATA_IN_DATA_U				(1 << 29)
-#define TEGRA_SPDIF_DATA_IN_DATA_V				(1 << 28)
-
-#define TEGRA_SPDIF_DATA_IN_DATA_PREAMBLE_SHIFT			24
-#define TEGRA_SPDIF_DATA_IN_DATA_PREAMBLE_MASK			(0xf << TEGRA_SPDIF_DATA_IN_DATA_PREAMBLE_SHIFT)
-
-#define TEGRA_SPDIF_DATA_IN_DATA_16_SHIFT			0
-#define TEGRA_SPDIF_DATA_IN_DATA_16_MASK			(0xffff << TEGRA_SPDIF_DATA_IN_DATA_16_SHIFT)
-
-#define TEGRA_SPDIF_DATA_IN_DATA_20_SHIFT			0
-#define TEGRA_SPDIF_DATA_IN_DATA_20_MASK			(0xfffff << TEGRA_SPDIF_DATA_IN_DATA_20_SHIFT)
-
-#define TEGRA_SPDIF_DATA_IN_DATA_24_SHIFT			0
-#define TEGRA_SPDIF_DATA_IN_DATA_24_MASK			(0xffffff << TEGRA_SPDIF_DATA_IN_DATA_24_SHIFT)
-
-#define TEGRA_SPDIF_DATA_IN_DATA_RAW_DATA_SHIFT			8
-#define TEGRA_SPDIF_DATA_IN_DATA_RAW_DATA_MASK			(0xfffff << TEGRA_SPDIF_DATA_IN_DATA_RAW_DATA_SHIFT)
-
-#define TEGRA_SPDIF_DATA_IN_DATA_RAW_AUX_SHIFT			4
-#define TEGRA_SPDIF_DATA_IN_DATA_RAW_AUX_MASK			(0xf << TEGRA_SPDIF_DATA_IN_DATA_RAW_AUX_SHIFT)
-
-#define TEGRA_SPDIF_DATA_IN_DATA_RAW_PREAMBLE_SHIFT		0
-#define TEGRA_SPDIF_DATA_IN_DATA_RAW_PREAMBLE_MASK		(0xf << TEGRA_SPDIF_DATA_IN_DATA_RAW_PREAMBLE_SHIFT)
-
-#define TEGRA_SPDIF_DATA_IN_DATA_16_PACKED_RIGHT_SHIFT		16
-#define TEGRA_SPDIF_DATA_IN_DATA_16_PACKED_RIGHT_MASK		(0xffff << TEGRA_SPDIF_DATA_IN_DATA_16_PACKED_RIGHT_SHIFT)
-
-#define TEGRA_SPDIF_DATA_IN_DATA_16_PACKED_LEFT_SHIFT		0
-#define TEGRA_SPDIF_DATA_IN_DATA_16_PACKED_LEFT_MASK		(0xffff << TEGRA_SPDIF_DATA_IN_DATA_16_PACKED_LEFT_SHIFT)
-
-/* Fields in TEGRA_SPDIF_CH_STA_RX_A */
-/* Fields in TEGRA_SPDIF_CH_STA_RX_B */
-/* Fields in TEGRA_SPDIF_CH_STA_RX_C */
-/* Fields in TEGRA_SPDIF_CH_STA_RX_D */
-/* Fields in TEGRA_SPDIF_CH_STA_RX_E */
-/* Fields in TEGRA_SPDIF_CH_STA_RX_F */
-
-/*
- * The 6-word receive channel data page buffer holds a block (192 frames) of
- * channel status information. The order of receive is from LSB to MSB
- * bit, and from CH_STA_RX_A to CH_STA_RX_F then back to CH_STA_RX_A.
- */
-
-/* Fields in TEGRA_SPDIF_CH_STA_TX_A */
-/* Fields in TEGRA_SPDIF_CH_STA_TX_B */
-/* Fields in TEGRA_SPDIF_CH_STA_TX_C */
-/* Fields in TEGRA_SPDIF_CH_STA_TX_D */
-/* Fields in TEGRA_SPDIF_CH_STA_TX_E */
-/* Fields in TEGRA_SPDIF_CH_STA_TX_F */
-
-/*
- * The 6-word transmit channel data page buffer holds a block (192 frames) of
- * channel status information. The order of transmission is from LSB to MSB
- * bit, and from CH_STA_TX_A to CH_STA_TX_F then back to CH_STA_TX_A.
- */
-
-/* Fields in TEGRA_SPDIF_USR_STA_RX_A */
-
-/*
- * This 4-word deep FIFO receives user FIFO field information. The order of
- * receive is from LSB to MSB bit.
- */
-
-/* Fields in TEGRA_SPDIF_USR_DAT_TX_A */
-
-/*
- * This 4-word deep FIFO transmits user FIFO field information. The order of
- * transmission is from LSB to MSB bit.
- */
-
-struct tegra_spdif {
-	struct clk *clk_spdif_out;
-	int clk_refs;
-	struct tegra_pcm_dma_params capture_dma_data;
-	struct tegra_pcm_dma_params playback_dma_data;
-	void __iomem *regs;
-	struct dentry *debug;
-	u32 reg_ctrl;
-};
-
-#endif
diff --git a/sound/soc/tegra/tegra_wm8753.c b/sound/soc/tegra/tegra_wm8753.c
new file mode 100644
index 000000000000..4e77026807a2
--- /dev/null
+++ b/sound/soc/tegra/tegra_wm8753.c
@@ -0,0 +1,224 @@
+/*
+ * tegra_wm8753.c - Tegra machine ASoC driver for boards using WM8753 codec.
+ *
+ * Author: Stephen Warren <swarren@nvidia.com>
+ * Copyright (C) 2010-2012 - NVIDIA, Inc.
+ *
+ * Based on code copyright/by:
+ *
+ * (c) 2009, 2010 Nvidia Graphics Pvt. Ltd.
+ *
+ * Copyright 2007 Wolfson Microelectronics PLC.
+ * Author: Graeme Gregory
+ *         graeme.gregory@wolfsonmicro.com or linux@wolfsonmicro.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
+ *
+ */
+
+#include <asm/mach-types.h>
+
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <linux/slab.h>
+#include <linux/gpio.h>
+#include <linux/of_gpio.h>
+
+#include <sound/core.h>
+#include <sound/jack.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+
+#include "../codecs/wm8753.h"
+
+#include "tegra_asoc_utils.h"
+
+#define DRV_NAME "tegra-snd-wm8753"
+
+struct tegra_wm8753 {
+	struct tegra_asoc_utils_data util_data;
+};
+
+static int tegra_wm8753_hw_params(struct snd_pcm_substream *substream,
+					struct snd_pcm_hw_params *params)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_codec *codec = rtd->codec;
+	struct snd_soc_card *card = codec->card;
+	struct tegra_wm8753 *machine = snd_soc_card_get_drvdata(card);
+	int srate, mclk;
+	int err;
+
+	srate = params_rate(params);
+	switch (srate) {
+	case 11025:
+	case 22050:
+	case 44100:
+	case 88200:
+		mclk = 11289600;
+		break;
+	default:
+		mclk = 12288000;
+		break;
+	}
+
+	err = tegra_asoc_utils_set_rate(&machine->util_data, srate, mclk);
+	if (err < 0) {
+		dev_err(card->dev, "Can't configure clocks\n");
+		return err;
+	}
+
+	err = snd_soc_dai_set_sysclk(codec_dai, WM8753_MCLK, mclk,
+					SND_SOC_CLOCK_IN);
+	if (err < 0) {
+		dev_err(card->dev, "codec_dai clock not set\n");
+		return err;
+	}
+
+	return 0;
+}
+
+static struct snd_soc_ops tegra_wm8753_ops = {
+	.hw_params = tegra_wm8753_hw_params,
+};
+
+static const struct snd_soc_dapm_widget tegra_wm8753_dapm_widgets[] = {
+	SND_SOC_DAPM_HP("Headphone Jack", NULL),
+	SND_SOC_DAPM_MIC("Mic Jack", NULL),
+};
+
+static struct snd_soc_dai_link tegra_wm8753_dai = {
+	.name = "WM8753",
+	.stream_name = "WM8753 PCM",
+	.codec_dai_name = "wm8753-hifi",
+	.ops = &tegra_wm8753_ops,
+	.dai_fmt = SND_SOC_DAIFMT_I2S |
+			SND_SOC_DAIFMT_NB_NF |
+			SND_SOC_DAIFMT_CBS_CFS,
+};
+
+static struct snd_soc_card snd_soc_tegra_wm8753 = {
+	.name = "tegra-wm8753",
+	.owner = THIS_MODULE,
+	.dai_link = &tegra_wm8753_dai,
+	.num_links = 1,
+
+	.dapm_widgets = tegra_wm8753_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(tegra_wm8753_dapm_widgets),
+	.fully_routed = true,
+};
+
+static __devinit int tegra_wm8753_driver_probe(struct platform_device *pdev)
+{
+	struct snd_soc_card *card = &snd_soc_tegra_wm8753;
+	struct tegra_wm8753 *machine;
+	int ret;
+
+	machine = devm_kzalloc(&pdev->dev, sizeof(struct tegra_wm8753),
+			       GFP_KERNEL);
+	if (!machine) {
+		dev_err(&pdev->dev, "Can't allocate tegra_wm8753 struct\n");
+		ret = -ENOMEM;
+		goto err;
+	}
+
+	card->dev = &pdev->dev;
+	platform_set_drvdata(pdev, card);
+	snd_soc_card_set_drvdata(card, machine);
+
+	ret = snd_soc_of_parse_card_name(card, "nvidia,model");
+	if (ret)
+		goto err;
+
+	ret = snd_soc_of_parse_audio_routing(card, "nvidia,audio-routing");
+	if (ret)
+		goto err;
+
+	tegra_wm8753_dai.codec_of_node = of_parse_phandle(
+			pdev->dev.of_node, "nvidia,audio-codec", 0);
+	if (!tegra_wm8753_dai.codec_of_node) {
+		dev_err(&pdev->dev,
+			"Property 'nvidia,audio-codec' missing or invalid\n");
+		ret = -EINVAL;
+		goto err;
+	}
+
+	tegra_wm8753_dai.cpu_dai_of_node = of_parse_phandle(
+			pdev->dev.of_node, "nvidia,i2s-controller", 0);
+	if (!tegra_wm8753_dai.cpu_dai_of_node) {
+		dev_err(&pdev->dev,
+			"Property 'nvidia,i2s-controller' missing or invalid\n");
+		ret = -EINVAL;
+		goto err;
+	}
+
+	tegra_wm8753_dai.platform_of_node =
+				tegra_wm8753_dai.cpu_dai_of_node;
+
+	ret = tegra_asoc_utils_init(&machine->util_data, &pdev->dev);
+	if (ret)
+		goto err;
+
+	ret = snd_soc_register_card(card);
+	if (ret) {
+		dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n",
+			ret);
+		goto err_fini_utils;
+	}
+
+	return 0;
+
+err_fini_utils:
+	tegra_asoc_utils_fini(&machine->util_data);
+err:
+	return ret;
+}
+
+static int __devexit tegra_wm8753_driver_remove(struct platform_device *pdev)
+{
+	struct snd_soc_card *card = platform_get_drvdata(pdev);
+	struct tegra_wm8753 *machine = snd_soc_card_get_drvdata(card);
+
+	snd_soc_unregister_card(card);
+
+	tegra_asoc_utils_fini(&machine->util_data);
+
+	return 0;
+}
+
+static const struct of_device_id tegra_wm8753_of_match[] __devinitconst = {
+	{ .compatible = "nvidia,tegra-audio-wm8753", },
+	{},
+};
+
+static struct platform_driver tegra_wm8753_driver = {
+	.driver = {
+		.name = DRV_NAME,
+		.owner = THIS_MODULE,
+		.pm = &snd_soc_pm_ops,
+		.of_match_table = tegra_wm8753_of_match,
+	},
+	.probe = tegra_wm8753_driver_probe,
+	.remove = __devexit_p(tegra_wm8753_driver_remove),
+};
+module_platform_driver(tegra_wm8753_driver);
+
+MODULE_AUTHOR("Stephen Warren <swarren@nvidia.com>");
+MODULE_DESCRIPTION("Tegra+WM8753 machine ASoC driver");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:" DRV_NAME);
+MODULE_DEVICE_TABLE(of, tegra_wm8753_of_match);
diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c
index 566655e23b7d..0b0df49d9d33 100644
--- a/sound/soc/tegra/tegra_wm8903.c
+++ b/sound/soc/tegra/tegra_wm8903.c
@@ -2,7 +2,7 @@
  * tegra_wm8903.c - Tegra machine ASoC driver for boards using WM8903 codec.
  *
  * Author: Stephen Warren <swarren@nvidia.com>
- * Copyright (C) 2010-2011 - NVIDIA, Inc.
+ * Copyright (C) 2010-2012 - NVIDIA, Inc.
  *
  * Based on code copyright/by:
  *
@@ -46,9 +46,6 @@
 
 #include "../codecs/wm8903.h"
 
-#include "tegra_das.h"
-#include "tegra_i2s.h"
-#include "tegra_pcm.h"
 #include "tegra_asoc_utils.h"
 
 #define DRV_NAME "tegra-snd-wm8903"
@@ -61,7 +58,6 @@
 
 struct tegra_wm8903 {
 	struct tegra_wm8903_platform_data pdata;
-	struct platform_device *pcm_dev;
 	struct tegra_asoc_utils_data util_data;
 	int gpio_requested;
 };
@@ -354,8 +350,8 @@ static struct snd_soc_dai_link tegra_wm8903_dai = {
 	.name = "WM8903",
 	.stream_name = "WM8903 PCM",
 	.codec_name = "wm8903.0-001a",
-	.platform_name = "tegra-pcm-audio",
-	.cpu_dai_name = "tegra-i2s.0",
+	.platform_name = "tegra20-i2s.0",
+	.cpu_dai_name = "tegra20-i2s.0",
 	.codec_dai_name = "wm8903-hifi",
 	.init = tegra_wm8903_init,
 	.ops = &tegra_wm8903_ops,
@@ -392,7 +388,6 @@ static __devinit int tegra_wm8903_driver_probe(struct platform_device *pdev)
 		ret = -ENOMEM;
 		goto err;
 	}
-	machine->pcm_dev = ERR_PTR(-EINVAL);
 
 	card->dev = &pdev->dev;
 	platform_set_drvdata(pdev, card);
@@ -428,14 +423,9 @@ static __devinit int tegra_wm8903_driver_probe(struct platform_device *pdev)
 			goto err;
 		}
 
-		machine->pcm_dev = platform_device_register_simple(
-					"tegra-pcm-audio", -1, NULL, 0);
-		if (IS_ERR(machine->pcm_dev)) {
-			dev_err(&pdev->dev,
-				"Can't instantiate tegra-pcm-audio\n");
-			ret = PTR_ERR(machine->pcm_dev);
-			goto err;
-		}
+		tegra_wm8903_dai.platform_name = NULL;
+		tegra_wm8903_dai.platform_of_node =
+					tegra_wm8903_dai.cpu_dai_of_node;
 	} else {
 		if (machine_is_harmony()) {
 			card->dapm_routes = harmony_audio_map;
@@ -454,7 +444,7 @@ static __devinit int tegra_wm8903_driver_probe(struct platform_device *pdev)
 
 	ret = tegra_asoc_utils_init(&machine->util_data, &pdev->dev);
 	if (ret)
-		goto err_unregister;
+		goto err;
 
 	ret = snd_soc_register_card(card);
 	if (ret) {
@@ -467,9 +457,6 @@ static __devinit int tegra_wm8903_driver_probe(struct platform_device *pdev)
 
 err_fini_utils:
 	tegra_asoc_utils_fini(&machine->util_data);
-err_unregister:
-	if (!IS_ERR(machine->pcm_dev))
-		platform_device_unregister(machine->pcm_dev);
 err:
 	return ret;
 }
@@ -497,8 +484,6 @@ static int __devexit tegra_wm8903_driver_remove(struct platform_device *pdev)
 	snd_soc_unregister_card(card);
 
 	tegra_asoc_utils_fini(&machine->util_data);
-	if (!IS_ERR(machine->pcm_dev))
-		platform_device_unregister(machine->pcm_dev);
 
 	return 0;
 }
diff --git a/sound/soc/tegra/trimslice.c b/sound/soc/tegra/trimslice.c
index 2bdfc550cff8..4a8d5b672c9f 100644
--- a/sound/soc/tegra/trimslice.c
+++ b/sound/soc/tegra/trimslice.c
@@ -27,6 +27,7 @@
 #include <asm/mach-types.h>
 
 #include <linux/module.h>
+#include <linux/of.h>
 #include <linux/platform_device.h>
 #include <linux/slab.h>
 
@@ -38,9 +39,6 @@
 
 #include "../codecs/tlv320aic23.h"
 
-#include "tegra_das.h"
-#include "tegra_i2s.h"
-#include "tegra_pcm.h"
 #include "tegra_asoc_utils.h"
 
 #define DRV_NAME "tegra-snd-trimslice"
@@ -119,8 +117,8 @@ static struct snd_soc_dai_link trimslice_tlv320aic23_dai = {
 	.name = "TLV320AIC23",
 	.stream_name = "AIC23",
 	.codec_name = "tlv320aic23-codec.2-001a",
-	.platform_name = "tegra-pcm-audio",
-	.cpu_dai_name = "tegra-i2s.0",
+	.platform_name = "tegra20-i2s.0",
+	.cpu_dai_name = "tegra20-i2s.0",
 	.codec_dai_name = "tlv320aic23-hifi",
 	.ops = &trimslice_asoc_ops,
 };
@@ -152,6 +150,32 @@ static __devinit int tegra_snd_trimslice_probe(struct platform_device *pdev)
 		goto err;
 	}
 
+	if (pdev->dev.of_node) {
+		trimslice_tlv320aic23_dai.codec_name = NULL;
+		trimslice_tlv320aic23_dai.codec_of_node = of_parse_phandle(
+				pdev->dev.of_node, "nvidia,audio-codec", 0);
+		if (!trimslice_tlv320aic23_dai.codec_of_node) {
+			dev_err(&pdev->dev,
+				"Property 'nvidia,audio-codec' missing or invalid\n");
+			ret = -EINVAL;
+			goto err;
+		}
+
+		trimslice_tlv320aic23_dai.cpu_dai_name = NULL;
+		trimslice_tlv320aic23_dai.cpu_dai_of_node = of_parse_phandle(
+				pdev->dev.of_node, "nvidia,i2s-controller", 0);
+		if (!trimslice_tlv320aic23_dai.cpu_dai_of_node) {
+			dev_err(&pdev->dev,
+				"Property 'nvidia,i2s-controller' missing or invalid\n");
+			ret = -EINVAL;
+			goto err;
+		}
+
+		trimslice_tlv320aic23_dai.platform_name = NULL;
+		trimslice_tlv320aic23_dai.platform_of_node =
+				trimslice_tlv320aic23_dai.cpu_dai_of_node;
+	}
+
 	ret = tegra_asoc_utils_init(&trimslice->util_data, &pdev->dev);
 	if (ret)
 		goto err;
@@ -187,10 +211,17 @@ static int __devexit tegra_snd_trimslice_remove(struct platform_device *pdev)
 	return 0;
 }
 
+static const struct of_device_id trimslice_of_match[] __devinitconst = {
+	{ .compatible = "nvidia,tegra-audio-trimslice", },
+	{},
+};
+MODULE_DEVICE_TABLE(of, trimslice_of_match);
+
 static struct platform_driver tegra_snd_trimslice_driver = {
 	.driver = {
 		.name = DRV_NAME,
 		.owner = THIS_MODULE,
+		.of_match_table = trimslice_of_match,
 	},
 	.probe = tegra_snd_trimslice_probe,
 	.remove = __devexit_p(tegra_snd_trimslice_remove),
diff --git a/sound/soc/ux500/Kconfig b/sound/soc/ux500/Kconfig
new file mode 100644
index 000000000000..44cf43404cd9
--- /dev/null
+++ b/sound/soc/ux500/Kconfig
@@ -0,0 +1,14 @@
+#
+# Ux500 SoC audio configuration
+#
+menuconfig SND_SOC_UX500
+	tristate "SoC Audio support for Ux500 platform"
+	depends on SND_SOC
+	depends on MFD_DB8500_PRCMU
+	help
+		Say Y if you want to enable ASoC-support for
+		any of the Ux500 platforms (e.g. U8500).
+
+config SND_SOC_UX500_PLAT_MSP_I2S
+	tristate
+	depends on SND_SOC_UX500
diff --git a/sound/soc/ux500/Makefile b/sound/soc/ux500/Makefile
new file mode 100644
index 000000000000..19974c5a2ea1
--- /dev/null
+++ b/sound/soc/ux500/Makefile
@@ -0,0 +1,4 @@
+# Ux500 Platform Support
+
+snd-soc-ux500-plat-msp-i2s-objs := ux500_msp_dai.o ux500_msp_i2s.o
+obj-$(CONFIG_SND_SOC_UX500_PLAT_MSP_I2S) += snd-soc-ux500-plat-msp-i2s.o
diff --git a/sound/soc/ux500/ux500_msp_dai.c b/sound/soc/ux500/ux500_msp_dai.c
new file mode 100644
index 000000000000..93c6c40e724c
--- /dev/null
+++ b/sound/soc/ux500/ux500_msp_dai.c
@@ -0,0 +1,843 @@
+/*
+ * Copyright (C) ST-Ericsson SA 2012
+ *
+ * Author: Ola Lilja <ola.o.lilja@stericsson.com>,
+ *         Roger Nilsson <roger.xr.nilsson@stericsson.com>
+ *         for ST-Ericsson.
+ *
+ * License terms:
+ *
+ * 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/slab.h>
+#include <linux/bitops.h>
+#include <linux/platform_device.h>
+#include <linux/clk.h>
+#include <linux/regulator/consumer.h>
+#include <linux/mfd/dbx500-prcmu.h>
+
+#include <mach/hardware.h>
+#include <mach/board-mop500-msp.h>
+
+#include <sound/soc.h>
+#include <sound/soc-dai.h>
+
+#include "ux500_msp_i2s.h"
+#include "ux500_msp_dai.h"
+
+static int setup_pcm_multichan(struct snd_soc_dai *dai,
+			struct ux500_msp_config *msp_config)
+{
+	struct ux500_msp_i2s_drvdata *drvdata = dev_get_drvdata(dai->dev);
+	struct msp_multichannel_config *multi =
+					&msp_config->multichannel_config;
+
+	if (drvdata->slots > 1) {
+		msp_config->multichannel_configured = 1;
+
+		multi->tx_multichannel_enable = true;
+		multi->rx_multichannel_enable = true;
+		multi->rx_comparison_enable_mode = MSP_COMPARISON_DISABLED;
+
+		multi->tx_channel_0_enable = drvdata->tx_mask;
+		multi->tx_channel_1_enable = 0;
+		multi->tx_channel_2_enable = 0;
+		multi->tx_channel_3_enable = 0;
+
+		multi->rx_channel_0_enable = drvdata->rx_mask;
+		multi->rx_channel_1_enable = 0;
+		multi->rx_channel_2_enable = 0;
+		multi->rx_channel_3_enable = 0;
+
+		dev_dbg(dai->dev,
+			"%s: Multichannel enabled. Slots: %d, TX: %u, RX: %u\n",
+			__func__, drvdata->slots, multi->tx_channel_0_enable,
+			multi->rx_channel_0_enable);
+	}
+
+	return 0;
+}
+
+static int setup_frameper(struct snd_soc_dai *dai, unsigned int rate,
+			struct msp_protdesc *prot_desc)
+{
+	struct ux500_msp_i2s_drvdata *drvdata = dev_get_drvdata(dai->dev);
+
+	switch (drvdata->slots) {
+	case 1:
+		switch (rate) {
+		case 8000:
+			prot_desc->frame_period =
+				FRAME_PER_SINGLE_SLOT_8_KHZ;
+			break;
+
+		case 16000:
+			prot_desc->frame_period =
+				FRAME_PER_SINGLE_SLOT_16_KHZ;
+			break;
+
+		case 44100:
+			prot_desc->frame_period =
+				FRAME_PER_SINGLE_SLOT_44_1_KHZ;
+			break;
+
+		case 48000:
+			prot_desc->frame_period =
+				FRAME_PER_SINGLE_SLOT_48_KHZ;
+			break;
+
+		default:
+			dev_err(dai->dev,
+				"%s: Error: Unsupported sample-rate (freq = %d)!\n",
+				__func__, rate);
+			return -EINVAL;
+		}
+		break;
+
+	case 2:
+		prot_desc->frame_period = FRAME_PER_2_SLOTS;
+		break;
+
+	case 8:
+		prot_desc->frame_period = FRAME_PER_8_SLOTS;
+		break;
+
+	case 16:
+		prot_desc->frame_period = FRAME_PER_16_SLOTS;
+		break;
+	default:
+		dev_err(dai->dev,
+			"%s: Error: Unsupported slot-count (slots = %d)!\n",
+			__func__, drvdata->slots);
+		return -EINVAL;
+	}
+
+	prot_desc->clocks_per_frame =
+			prot_desc->frame_period+1;
+
+	dev_dbg(dai->dev, "%s: Clocks per frame: %u\n",
+		__func__,
+		prot_desc->clocks_per_frame);
+
+	return 0;
+}
+
+static int setup_pcm_framing(struct snd_soc_dai *dai, unsigned int rate,
+			struct msp_protdesc *prot_desc)
+{
+	struct ux500_msp_i2s_drvdata *drvdata = dev_get_drvdata(dai->dev);
+
+	u32 frame_length = MSP_FRAME_LEN_1;
+	prot_desc->frame_width = 0;
+
+	switch (drvdata->slots) {
+	case 1:
+		frame_length = MSP_FRAME_LEN_1;
+		break;
+
+	case 2:
+		frame_length = MSP_FRAME_LEN_2;
+		break;
+
+	case 8:
+		frame_length = MSP_FRAME_LEN_8;
+		break;
+
+	case 16:
+		frame_length = MSP_FRAME_LEN_16;
+		break;
+	default:
+		dev_err(dai->dev,
+			"%s: Error: Unsupported slot-count (slots = %d)!\n",
+			__func__, drvdata->slots);
+		return -EINVAL;
+	}
+
+	prot_desc->tx_frame_len_1 = frame_length;
+	prot_desc->rx_frame_len_1 = frame_length;
+	prot_desc->tx_frame_len_2 = frame_length;
+	prot_desc->rx_frame_len_2 = frame_length;
+
+	prot_desc->tx_elem_len_1 = MSP_ELEM_LEN_16;
+	prot_desc->rx_elem_len_1 = MSP_ELEM_LEN_16;
+	prot_desc->tx_elem_len_2 = MSP_ELEM_LEN_16;
+	prot_desc->rx_elem_len_2 = MSP_ELEM_LEN_16;
+
+	return setup_frameper(dai, rate, prot_desc);
+}
+
+static int setup_clocking(struct snd_soc_dai *dai,
+			unsigned int fmt,
+			struct ux500_msp_config *msp_config)
+{
+	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+	case SND_SOC_DAIFMT_NB_NF:
+		break;
+
+	case SND_SOC_DAIFMT_NB_IF:
+		msp_config->tx_fsync_pol ^= 1 << TFSPOL_SHIFT;
+		msp_config->rx_fsync_pol ^= 1 << RFSPOL_SHIFT;
+
+		break;
+
+	default:
+		dev_err(dai->dev,
+			"%s: Error: Unsopported inversion (fmt = 0x%x)!\n",
+			__func__, fmt);
+
+		return -EINVAL;
+	}
+
+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBM_CFM:
+		dev_dbg(dai->dev, "%s: Codec is master.\n", __func__);
+
+		msp_config->iodelay = 0x20;
+		msp_config->rx_fsync_sel = 0;
+		msp_config->tx_fsync_sel = 1 << TFSSEL_SHIFT;
+		msp_config->tx_clk_sel = 0;
+		msp_config->rx_clk_sel = 0;
+		msp_config->srg_clk_sel = 0x2 << SCKSEL_SHIFT;
+
+		break;
+
+	case SND_SOC_DAIFMT_CBS_CFS:
+		dev_dbg(dai->dev, "%s: Codec is slave.\n", __func__);
+
+		msp_config->tx_clk_sel = TX_CLK_SEL_SRG;
+		msp_config->tx_fsync_sel = TX_SYNC_SRG_PROG;
+		msp_config->rx_clk_sel = RX_CLK_SEL_SRG;
+		msp_config->rx_fsync_sel = RX_SYNC_SRG;
+		msp_config->srg_clk_sel = 1 << SCKSEL_SHIFT;
+
+		break;
+
+	default:
+		dev_err(dai->dev, "%s: Error: Unsopported master (fmt = 0x%x)!\n",
+			__func__, fmt);
+
+		return -EINVAL;
+	}
+
+	return 0;
+}
+
+static int setup_pcm_protdesc(struct snd_soc_dai *dai,
+				unsigned int fmt,
+				struct msp_protdesc *prot_desc)
+{
+	prot_desc->rx_phase_mode = MSP_SINGLE_PHASE;
+	prot_desc->tx_phase_mode = MSP_SINGLE_PHASE;
+	prot_desc->rx_phase2_start_mode = MSP_PHASE2_START_MODE_IMEDIATE;
+	prot_desc->tx_phase2_start_mode = MSP_PHASE2_START_MODE_IMEDIATE;
+	prot_desc->rx_byte_order = MSP_BTF_MS_BIT_FIRST;
+	prot_desc->tx_byte_order = MSP_BTF_MS_BIT_FIRST;
+	prot_desc->tx_fsync_pol = MSP_FSYNC_POL(MSP_FSYNC_POL_ACT_HI);
+	prot_desc->rx_fsync_pol = MSP_FSYNC_POL_ACT_HI << RFSPOL_SHIFT;
+
+	if ((fmt & SND_SOC_DAIFMT_FORMAT_MASK) == SND_SOC_DAIFMT_DSP_A) {
+		dev_dbg(dai->dev, "%s: DSP_A.\n", __func__);
+		prot_desc->rx_clk_pol = MSP_RISING_EDGE;
+		prot_desc->tx_clk_pol = MSP_FALLING_EDGE;
+
+		prot_desc->rx_data_delay = MSP_DELAY_1;
+		prot_desc->tx_data_delay = MSP_DELAY_1;
+	} else {
+		dev_dbg(dai->dev, "%s: DSP_B.\n", __func__);
+		prot_desc->rx_clk_pol = MSP_FALLING_EDGE;
+		prot_desc->tx_clk_pol = MSP_RISING_EDGE;
+
+		prot_desc->rx_data_delay = MSP_DELAY_0;
+		prot_desc->tx_data_delay = MSP_DELAY_0;
+	}
+
+	prot_desc->rx_half_word_swap = MSP_SWAP_NONE;
+	prot_desc->tx_half_word_swap = MSP_SWAP_NONE;
+	prot_desc->compression_mode = MSP_COMPRESS_MODE_LINEAR;
+	prot_desc->expansion_mode = MSP_EXPAND_MODE_LINEAR;
+	prot_desc->frame_sync_ignore = MSP_FSYNC_IGNORE;
+
+	return 0;
+}
+
+static int setup_i2s_protdesc(struct msp_protdesc *prot_desc)
+{
+	prot_desc->rx_phase_mode = MSP_DUAL_PHASE;
+	prot_desc->tx_phase_mode = MSP_DUAL_PHASE;
+	prot_desc->rx_phase2_start_mode = MSP_PHASE2_START_MODE_FSYNC;
+	prot_desc->tx_phase2_start_mode = MSP_PHASE2_START_MODE_FSYNC;
+	prot_desc->rx_byte_order = MSP_BTF_MS_BIT_FIRST;
+	prot_desc->tx_byte_order = MSP_BTF_MS_BIT_FIRST;
+	prot_desc->tx_fsync_pol = MSP_FSYNC_POL(MSP_FSYNC_POL_ACT_LO);
+	prot_desc->rx_fsync_pol = MSP_FSYNC_POL_ACT_LO << RFSPOL_SHIFT;
+
+	prot_desc->rx_frame_len_1 = MSP_FRAME_LEN_1;
+	prot_desc->rx_frame_len_2 = MSP_FRAME_LEN_1;
+	prot_desc->tx_frame_len_1 = MSP_FRAME_LEN_1;
+	prot_desc->tx_frame_len_2 = MSP_FRAME_LEN_1;
+	prot_desc->rx_elem_len_1 = MSP_ELEM_LEN_16;
+	prot_desc->rx_elem_len_2 = MSP_ELEM_LEN_16;
+	prot_desc->tx_elem_len_1 = MSP_ELEM_LEN_16;
+	prot_desc->tx_elem_len_2 = MSP_ELEM_LEN_16;
+
+	prot_desc->rx_clk_pol = MSP_RISING_EDGE;
+	prot_desc->tx_clk_pol = MSP_FALLING_EDGE;
+
+	prot_desc->rx_data_delay = MSP_DELAY_0;
+	prot_desc->tx_data_delay = MSP_DELAY_0;
+
+	prot_desc->tx_half_word_swap = MSP_SWAP_NONE;
+	prot_desc->rx_half_word_swap = MSP_SWAP_NONE;
+	prot_desc->compression_mode = MSP_COMPRESS_MODE_LINEAR;
+	prot_desc->expansion_mode = MSP_EXPAND_MODE_LINEAR;
+	prot_desc->frame_sync_ignore = MSP_FSYNC_IGNORE;
+
+	return 0;
+}
+
+static int setup_msp_config(struct snd_pcm_substream *substream,
+			struct snd_soc_dai *dai,
+			struct ux500_msp_config *msp_config)
+{
+	struct ux500_msp_i2s_drvdata *drvdata = dev_get_drvdata(dai->dev);
+	struct msp_protdesc *prot_desc = &msp_config->protdesc;
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	unsigned int fmt = drvdata->fmt;
+	int ret;
+
+	memset(msp_config, 0, sizeof(*msp_config));
+
+	msp_config->f_inputclk = drvdata->master_clk;
+
+	msp_config->tx_fifo_config = TX_FIFO_ENABLE;
+	msp_config->rx_fifo_config = RX_FIFO_ENABLE;
+	msp_config->def_elem_len = 1;
+	msp_config->direction = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ?
+				MSP_DIR_TX : MSP_DIR_RX;
+	msp_config->data_size = MSP_DATA_BITS_32;
+	msp_config->frame_freq = runtime->rate;
+
+	dev_dbg(dai->dev, "%s: f_inputclk = %u, frame_freq = %u.\n",
+	       __func__, msp_config->f_inputclk, msp_config->frame_freq);
+	/* To avoid division by zero */
+	prot_desc->clocks_per_frame = 1;
+
+	dev_dbg(dai->dev, "%s: rate: %u, channels: %d.\n", __func__,
+		runtime->rate, runtime->channels);
+	switch (fmt &
+		(SND_SOC_DAIFMT_FORMAT_MASK | SND_SOC_DAIFMT_MASTER_MASK)) {
+	case SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS:
+		dev_dbg(dai->dev, "%s: SND_SOC_DAIFMT_I2S.\n", __func__);
+
+		msp_config->default_protdesc = 1;
+		msp_config->protocol = MSP_I2S_PROTOCOL;
+		break;
+
+	case SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM:
+		dev_dbg(dai->dev, "%s: SND_SOC_DAIFMT_I2S.\n", __func__);
+
+		msp_config->data_size = MSP_DATA_BITS_16;
+		msp_config->protocol = MSP_I2S_PROTOCOL;
+
+		ret = setup_i2s_protdesc(prot_desc);
+		if (ret < 0)
+			return ret;
+
+		break;
+
+	case SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_CBS_CFS:
+	case SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_CBM_CFM:
+	case SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_CBS_CFS:
+	case SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_CBM_CFM:
+		dev_dbg(dai->dev, "%s: PCM format.\n", __func__);
+
+		msp_config->data_size = MSP_DATA_BITS_16;
+		msp_config->protocol = MSP_PCM_PROTOCOL;
+
+		ret = setup_pcm_protdesc(dai, fmt, prot_desc);
+		if (ret < 0)
+			return ret;
+
+		ret = setup_pcm_multichan(dai, msp_config);
+		if (ret < 0)
+			return ret;
+
+		ret = setup_pcm_framing(dai, runtime->rate, prot_desc);
+		if (ret < 0)
+			return ret;
+
+		break;
+
+	default:
+		dev_err(dai->dev, "%s: Error: Unsopported format (%d)!\n",
+			__func__, fmt);
+		return -EINVAL;
+	}
+
+	return setup_clocking(dai, fmt, msp_config);
+}
+
+static int ux500_msp_dai_startup(struct snd_pcm_substream *substream,
+				struct snd_soc_dai *dai)
+{
+	int ret = 0;
+	struct ux500_msp_i2s_drvdata *drvdata = dev_get_drvdata(dai->dev);
+
+	dev_dbg(dai->dev, "%s: MSP %d (%s): Enter.\n", __func__, dai->id,
+		snd_pcm_stream_str(substream));
+
+	/* Enable regulator */
+	ret = regulator_enable(drvdata->reg_vape);
+	if (ret != 0) {
+		dev_err(drvdata->msp->dev,
+			"%s: Failed to enable regulator!\n", __func__);
+		return ret;
+	}
+
+	/* Enable clock */
+	dev_dbg(dai->dev, "%s: Enabling MSP-clock.\n", __func__);
+	clk_enable(drvdata->clk);
+
+	return 0;
+}
+
+static void ux500_msp_dai_shutdown(struct snd_pcm_substream *substream,
+				struct snd_soc_dai *dai)
+{
+	int ret;
+	struct ux500_msp_i2s_drvdata *drvdata = dev_get_drvdata(dai->dev);
+	bool is_playback = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK);
+
+	dev_dbg(dai->dev, "%s: MSP %d (%s): Enter.\n", __func__, dai->id,
+		snd_pcm_stream_str(substream));
+
+	if (drvdata->vape_opp_constraint == 1) {
+		prcmu_qos_update_requirement(PRCMU_QOS_APE_OPP,
+					"ux500_msp_i2s", 50);
+		drvdata->vape_opp_constraint = 0;
+	}
+
+	if (ux500_msp_i2s_close(drvdata->msp,
+				is_playback ? MSP_DIR_TX : MSP_DIR_RX)) {
+		dev_err(dai->dev,
+			"%s: Error: MSP %d (%s): Unable to close i2s.\n",
+			__func__, dai->id, snd_pcm_stream_str(substream));
+	}
+
+	/* Disable clock */
+	clk_disable(drvdata->clk);
+
+	/* Disable regulator */
+	ret = regulator_disable(drvdata->reg_vape);
+	if (ret < 0)
+		dev_err(dai->dev,
+			"%s: ERROR: Failed to disable regulator (%d)!\n",
+			__func__, ret);
+}
+
+static int ux500_msp_dai_prepare(struct snd_pcm_substream *substream,
+				struct snd_soc_dai *dai)
+{
+	int ret = 0;
+	struct ux500_msp_i2s_drvdata *drvdata = dev_get_drvdata(dai->dev);
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct ux500_msp_config msp_config;
+
+	dev_dbg(dai->dev, "%s: MSP %d (%s): Enter (rate = %d).\n", __func__,
+		dai->id, snd_pcm_stream_str(substream), runtime->rate);
+
+	setup_msp_config(substream, dai, &msp_config);
+
+	ret = ux500_msp_i2s_open(drvdata->msp, &msp_config);
+	if (ret < 0) {
+		dev_err(dai->dev, "%s: Error: msp_setup failed (ret = %d)!\n",
+			__func__, ret);
+		return ret;
+	}
+
+	/* Set OPP-level */
+	if ((drvdata->fmt & SND_SOC_DAIFMT_MASTER_MASK) &&
+		(drvdata->msp->f_bitclk > 19200000)) {
+		/* If the bit-clock is higher than 19.2MHz, Vape should be
+		 * run in 100% OPP. Only when bit-clock is used (MSP master) */
+		prcmu_qos_update_requirement(PRCMU_QOS_APE_OPP,
+					"ux500-msp-i2s", 100);
+		drvdata->vape_opp_constraint = 1;
+	} else {
+		prcmu_qos_update_requirement(PRCMU_QOS_APE_OPP,
+					"ux500-msp-i2s", 50);
+		drvdata->vape_opp_constraint = 0;
+	}
+
+	return ret;
+}
+
+static int ux500_msp_dai_hw_params(struct snd_pcm_substream *substream,
+				struct snd_pcm_hw_params *params,
+				struct snd_soc_dai *dai)
+{
+	unsigned int mask, slots_active;
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct ux500_msp_i2s_drvdata *drvdata = dev_get_drvdata(dai->dev);
+
+	dev_dbg(dai->dev, "%s: MSP %d (%s): Enter.\n",
+			__func__, dai->id, snd_pcm_stream_str(substream));
+
+	switch (drvdata->fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+		snd_pcm_hw_constraint_minmax(runtime,
+				SNDRV_PCM_HW_PARAM_CHANNELS,
+				1, 2);
+		break;
+
+	case SND_SOC_DAIFMT_DSP_B:
+	case SND_SOC_DAIFMT_DSP_A:
+		mask = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ?
+			drvdata->tx_mask :
+			drvdata->rx_mask;
+
+		slots_active = hweight32(mask);
+		dev_dbg(dai->dev, "TDM-slots active: %d", slots_active);
+
+		snd_pcm_hw_constraint_minmax(runtime,
+				SNDRV_PCM_HW_PARAM_CHANNELS,
+				slots_active, slots_active);
+		break;
+
+	default:
+		dev_err(dai->dev,
+			"%s: Error: Unsupported protocol (fmt = 0x%x)!\n",
+			__func__, drvdata->fmt);
+		return -EINVAL;
+	}
+
+	return 0;
+}
+
+static int ux500_msp_dai_set_dai_fmt(struct snd_soc_dai *dai,
+				unsigned int fmt)
+{
+	struct ux500_msp_i2s_drvdata *drvdata = dev_get_drvdata(dai->dev);
+
+	dev_dbg(dai->dev, "%s: MSP %d: Enter.\n", __func__, dai->id);
+
+	switch (fmt & (SND_SOC_DAIFMT_FORMAT_MASK |
+		SND_SOC_DAIFMT_MASTER_MASK)) {
+	case SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS:
+	case SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM:
+	case SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_CBS_CFS:
+	case SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_CBM_CFM:
+	case SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_CBS_CFS:
+	case SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_CBM_CFM:
+		break;
+
+	default:
+		dev_err(dai->dev,
+			"%s: Error: Unsupported protocol/master (fmt = 0x%x)!\n",
+			__func__, drvdata->fmt);
+		return -EINVAL;
+	}
+
+	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+	case SND_SOC_DAIFMT_NB_NF:
+	case SND_SOC_DAIFMT_NB_IF:
+	case SND_SOC_DAIFMT_IB_IF:
+		break;
+
+	default:
+		dev_err(dai->dev,
+			"%s: Error: Unsupported inversion (fmt = 0x%x)!\n",
+			__func__, drvdata->fmt);
+		return -EINVAL;
+	}
+
+	drvdata->fmt = fmt;
+	return 0;
+}
+
+static int ux500_msp_dai_set_tdm_slot(struct snd_soc_dai *dai,
+				unsigned int tx_mask,
+				unsigned int rx_mask,
+				int slots, int slot_width)
+{
+	struct ux500_msp_i2s_drvdata *drvdata = dev_get_drvdata(dai->dev);
+	unsigned int cap;
+
+	switch (slots) {
+	case 1:
+		cap = 0x01;
+		break;
+	case 2:
+		cap = 0x03;
+		break;
+	case 8:
+		cap = 0xFF;
+		break;
+	case 16:
+		cap = 0xFFFF;
+		break;
+	default:
+		dev_err(dai->dev, "%s: Error: Unsupported slot-count (%d)!\n",
+			__func__, slots);
+		return -EINVAL;
+	}
+	drvdata->slots = slots;
+
+	if (!(slot_width == 16)) {
+		dev_err(dai->dev, "%s: Error: Unsupported slot-width (%d)!\n",
+			__func__, slot_width);
+		return -EINVAL;
+	}
+	drvdata->slot_width = slot_width;
+
+	drvdata->tx_mask = tx_mask & cap;
+	drvdata->rx_mask = rx_mask & cap;
+
+	return 0;
+}
+
+static int ux500_msp_dai_set_dai_sysclk(struct snd_soc_dai *dai,
+					int clk_id, unsigned int freq, int dir)
+{
+	struct ux500_msp_i2s_drvdata *drvdata = dev_get_drvdata(dai->dev);
+
+	dev_dbg(dai->dev, "%s: MSP %d: Enter. clk-id: %d, freq: %u.\n",
+		__func__, dai->id, clk_id, freq);
+
+	switch (clk_id) {
+	case UX500_MSP_MASTER_CLOCK:
+		drvdata->master_clk = freq;
+		break;
+
+	default:
+		dev_err(dai->dev, "%s: MSP %d: Invalid clk-id (%d)!\n",
+			__func__, dai->id, clk_id);
+		return -EINVAL;
+	}
+
+	return 0;
+}
+
+static int ux500_msp_dai_trigger(struct snd_pcm_substream *substream,
+				int cmd, struct snd_soc_dai *dai)
+{
+	int ret = 0;
+	struct ux500_msp_i2s_drvdata *drvdata = dev_get_drvdata(dai->dev);
+
+	dev_dbg(dai->dev, "%s: MSP %d (%s): Enter (msp->id = %d, cmd = %d).\n",
+		__func__, dai->id, snd_pcm_stream_str(substream),
+		(int)drvdata->msp->id, cmd);
+
+	ret = ux500_msp_i2s_trigger(drvdata->msp, cmd, substream->stream);
+
+	return ret;
+}
+
+static int ux500_msp_dai_probe(struct snd_soc_dai *dai)
+{
+	struct ux500_msp_i2s_drvdata *drvdata = dev_get_drvdata(dai->dev);
+
+	drvdata->playback_dma_data.dma_cfg = drvdata->msp->dma_cfg_tx;
+	drvdata->capture_dma_data.dma_cfg = drvdata->msp->dma_cfg_rx;
+
+	dai->playback_dma_data = &drvdata->playback_dma_data;
+	dai->capture_dma_data = &drvdata->capture_dma_data;
+
+	drvdata->playback_dma_data.data_size = drvdata->slot_width;
+	drvdata->capture_dma_data.data_size = drvdata->slot_width;
+
+	return 0;
+}
+
+static struct snd_soc_dai_ops ux500_msp_dai_ops[] = {
+	{
+		.set_sysclk = ux500_msp_dai_set_dai_sysclk,
+		.set_fmt = ux500_msp_dai_set_dai_fmt,
+		.set_tdm_slot = ux500_msp_dai_set_tdm_slot,
+		.startup = ux500_msp_dai_startup,
+		.shutdown = ux500_msp_dai_shutdown,
+		.prepare = ux500_msp_dai_prepare,
+		.trigger = ux500_msp_dai_trigger,
+		.hw_params = ux500_msp_dai_hw_params,
+	}
+};
+
+static struct snd_soc_dai_driver ux500_msp_dai_drv[UX500_NBR_OF_DAI] = {
+	{
+		.name = "ux500-msp-i2s.0",
+		.probe = ux500_msp_dai_probe,
+		.id = 0,
+		.suspend = NULL,
+		.resume = NULL,
+		.playback = {
+			.channels_min = UX500_MSP_MIN_CHANNELS,
+			.channels_max = UX500_MSP_MAX_CHANNELS,
+			.rates = UX500_I2S_RATES,
+			.formats = UX500_I2S_FORMATS,
+		},
+		.capture = {
+			.channels_min = UX500_MSP_MIN_CHANNELS,
+			.channels_max = UX500_MSP_MAX_CHANNELS,
+			.rates = UX500_I2S_RATES,
+			.formats = UX500_I2S_FORMATS,
+		},
+		.ops = ux500_msp_dai_ops,
+	},
+	{
+		.name = "ux500-msp-i2s.1",
+		.probe = ux500_msp_dai_probe,
+		.id = 1,
+		.suspend = NULL,
+		.resume = NULL,
+		.playback = {
+			.channels_min = UX500_MSP_MIN_CHANNELS,
+			.channels_max = UX500_MSP_MAX_CHANNELS,
+			.rates = UX500_I2S_RATES,
+			.formats = UX500_I2S_FORMATS,
+		},
+		.capture = {
+			.channels_min = UX500_MSP_MIN_CHANNELS,
+			.channels_max = UX500_MSP_MAX_CHANNELS,
+			.rates = UX500_I2S_RATES,
+			.formats = UX500_I2S_FORMATS,
+		},
+		.ops = ux500_msp_dai_ops,
+	},
+	{
+		.name = "ux500-msp-i2s.2",
+		.id = 2,
+		.probe = ux500_msp_dai_probe,
+		.suspend = NULL,
+		.resume = NULL,
+		.playback = {
+			.channels_min = UX500_MSP_MIN_CHANNELS,
+			.channels_max = UX500_MSP_MAX_CHANNELS,
+			.rates = UX500_I2S_RATES,
+			.formats = UX500_I2S_FORMATS,
+		},
+		.capture = {
+			.channels_min = UX500_MSP_MIN_CHANNELS,
+			.channels_max = UX500_MSP_MAX_CHANNELS,
+			.rates = UX500_I2S_RATES,
+			.formats = UX500_I2S_FORMATS,
+		},
+		.ops = ux500_msp_dai_ops,
+	},
+	{
+		.name = "ux500-msp-i2s.3",
+		.probe = ux500_msp_dai_probe,
+		.id = 3,
+		.suspend = NULL,
+		.resume = NULL,
+		.playback = {
+			.channels_min = UX500_MSP_MIN_CHANNELS,
+			.channels_max = UX500_MSP_MAX_CHANNELS,
+			.rates = UX500_I2S_RATES,
+			.formats = UX500_I2S_FORMATS,
+		},
+		.capture = {
+			.channels_min = UX500_MSP_MIN_CHANNELS,
+			.channels_max = UX500_MSP_MAX_CHANNELS,
+			.rates = UX500_I2S_RATES,
+			.formats = UX500_I2S_FORMATS,
+		},
+		.ops = ux500_msp_dai_ops,
+	},
+};
+
+static int __devinit ux500_msp_drv_probe(struct platform_device *pdev)
+{
+	struct ux500_msp_i2s_drvdata *drvdata;
+	int ret = 0;
+
+	dev_dbg(&pdev->dev, "%s: Enter (pdev->name = %s).\n", __func__,
+		pdev->name);
+
+	drvdata = devm_kzalloc(&pdev->dev,
+				sizeof(struct ux500_msp_i2s_drvdata),
+				GFP_KERNEL);
+	drvdata->fmt = 0;
+	drvdata->slots = 1;
+	drvdata->tx_mask = 0x01;
+	drvdata->rx_mask = 0x01;
+	drvdata->slot_width = 16;
+	drvdata->master_clk = MSP_INPUT_FREQ_APB;
+
+	drvdata->reg_vape = devm_regulator_get(&pdev->dev, "v-ape");
+	if (IS_ERR(drvdata->reg_vape)) {
+		ret = (int)PTR_ERR(drvdata->reg_vape);
+		dev_err(&pdev->dev,
+			"%s: ERROR: Failed to get Vape supply (%d)!\n",
+			__func__, ret);
+		return ret;
+	}
+	prcmu_qos_add_requirement(PRCMU_QOS_APE_OPP, (char *)pdev->name, 50);
+
+	drvdata->clk = clk_get(&pdev->dev, NULL);
+	if (IS_ERR(drvdata->clk)) {
+		ret = (int)PTR_ERR(drvdata->clk);
+		dev_err(&pdev->dev, "%s: ERROR: clk_get failed (%d)!\n",
+			__func__, ret);
+		goto err_clk;
+	}
+
+	ret = ux500_msp_i2s_init_msp(pdev, &drvdata->msp,
+				pdev->dev.platform_data);
+	if (!drvdata->msp) {
+		dev_err(&pdev->dev,
+			"%s: ERROR: Failed to init MSP-struct (%d)!",
+			__func__, ret);
+		goto err_init_msp;
+	}
+	dev_set_drvdata(&pdev->dev, drvdata);
+
+	ret = snd_soc_register_dai(&pdev->dev,
+				&ux500_msp_dai_drv[drvdata->msp->id]);
+	if (ret < 0) {
+		dev_err(&pdev->dev, "Error: %s: Failed to register MSP%d!\n",
+			__func__, drvdata->msp->id);
+		goto err_init_msp;
+	}
+
+	return 0;
+
+err_init_msp:
+	clk_put(drvdata->clk);
+
+err_clk:
+	devm_regulator_put(drvdata->reg_vape);
+
+	return ret;
+}
+
+static int __devexit ux500_msp_drv_remove(struct platform_device *pdev)
+{
+	struct ux500_msp_i2s_drvdata *drvdata = dev_get_drvdata(&pdev->dev);
+
+	snd_soc_unregister_dais(&pdev->dev, ARRAY_SIZE(ux500_msp_dai_drv));
+
+	devm_regulator_put(drvdata->reg_vape);
+	prcmu_qos_remove_requirement(PRCMU_QOS_APE_OPP, "ux500_msp_i2s");
+
+	clk_put(drvdata->clk);
+
+	ux500_msp_i2s_cleanup_msp(pdev, drvdata->msp);
+
+	return 0;
+}
+
+static struct platform_driver msp_i2s_driver = {
+	.driver = {
+		.name = "ux500-msp-i2s",
+		.owner = THIS_MODULE,
+	},
+	.probe = ux500_msp_drv_probe,
+	.remove = ux500_msp_drv_remove,
+};
+module_platform_driver(msp_i2s_driver);
+
+MODULE_LICENSE("GPLv2");
diff --git a/sound/soc/ux500/ux500_msp_dai.h b/sound/soc/ux500/ux500_msp_dai.h
new file mode 100644
index 000000000000..98202a34a5dd
--- /dev/null
+++ b/sound/soc/ux500/ux500_msp_dai.h
@@ -0,0 +1,79 @@
+/*
+ * Copyright (C) ST-Ericsson SA 2012
+ *
+ * Author: Ola Lilja <ola.o.lilja@stericsson.com>,
+ *         Roger Nilsson <roger.xr.nilsson@stericsson.com>
+ *         for ST-Ericsson.
+ *
+ * License terms:
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as published
+ * by the Free Software Foundation.
+ */
+
+#ifndef UX500_msp_dai_H
+#define UX500_msp_dai_H
+
+#include <linux/types.h>
+#include <linux/spinlock.h>
+
+#include "ux500_msp_i2s.h"
+
+#define UX500_NBR_OF_DAI	4
+
+#define UX500_I2S_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |	\
+			SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000)
+
+#define UX500_I2S_FORMATS (SNDRV_PCM_FMTBIT_S16_LE)
+
+#define FRAME_PER_SINGLE_SLOT_8_KHZ		31
+#define FRAME_PER_SINGLE_SLOT_16_KHZ	124
+#define FRAME_PER_SINGLE_SLOT_44_1_KHZ	63
+#define FRAME_PER_SINGLE_SLOT_48_KHZ	49
+#define FRAME_PER_2_SLOTS				31
+#define FRAME_PER_8_SLOTS				138
+#define FRAME_PER_16_SLOTS				277
+
+#ifndef CONFIG_SND_SOC_UX500_AB5500
+#define UX500_MSP_INTERNAL_CLOCK_FREQ  40000000
+#define UX500_MSP1_INTERNAL_CLOCK_FREQ UX500_MSP_INTERNAL_CLOCK_FREQ
+#else
+#define UX500_MSP_INTERNAL_CLOCK_FREQ 13000000
+#define UX500_MSP1_INTERNAL_CLOCK_FREQ (UX500_MSP_INTERNAL_CLOCK_FREQ * 2)
+#endif
+
+#define UX500_MSP_MIN_CHANNELS		1
+#define UX500_MSP_MAX_CHANNELS		8
+
+#define PLAYBACK_CONFIGURED		1
+#define CAPTURE_CONFIGURED		2
+
+enum ux500_msp_clock_id {
+	UX500_MSP_MASTER_CLOCK,
+};
+
+struct ux500_msp_i2s_drvdata {
+	struct ux500_msp *msp;
+	struct regulator *reg_vape;
+	struct ux500_msp_dma_params playback_dma_data;
+	struct ux500_msp_dma_params capture_dma_data;
+	unsigned int fmt;
+	unsigned int tx_mask;
+	unsigned int rx_mask;
+	int slots;
+	int slot_width;
+	u8 configured;
+	int data_delay;
+
+	/* Clocks */
+	unsigned int master_clk;
+	struct clk *clk;
+
+	/* Regulators */
+	int vape_opp_constraint;
+};
+
+int ux500_msp_dai_set_data_delay(struct snd_soc_dai *dai, int delay);
+
+#endif
diff --git a/sound/soc/ux500/ux500_msp_i2s.c b/sound/soc/ux500/ux500_msp_i2s.c
new file mode 100644
index 000000000000..496dec10c96e
--- /dev/null
+++ b/sound/soc/ux500/ux500_msp_i2s.c
@@ -0,0 +1,742 @@
+/*
+ * Copyright (C) ST-Ericsson SA 2012
+ *
+ * Author: Ola Lilja <ola.o.lilja@stericsson.com>,
+ *         Roger Nilsson <roger.xr.nilsson@stericsson.com>,
+ *         Sandeep Kaushik <sandeep.kaushik@st.com>
+ *         for ST-Ericsson.
+ *
+ * License terms:
+ *
+ * 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/platform_device.h>
+#include <linux/delay.h>
+#include <linux/slab.h>
+
+#include <mach/hardware.h>
+#include <mach/board-mop500-msp.h>
+
+#include <sound/soc.h>
+
+#include "ux500_msp_i2s.h"
+
+ /* Protocol desciptors */
+static const struct msp_protdesc prot_descs[] = {
+	{ /* I2S */
+		MSP_SINGLE_PHASE,
+		MSP_SINGLE_PHASE,
+		MSP_PHASE2_START_MODE_IMEDIATE,
+		MSP_PHASE2_START_MODE_IMEDIATE,
+		MSP_BTF_MS_BIT_FIRST,
+		MSP_BTF_MS_BIT_FIRST,
+		MSP_FRAME_LEN_1,
+		MSP_FRAME_LEN_1,
+		MSP_FRAME_LEN_1,
+		MSP_FRAME_LEN_1,
+		MSP_ELEM_LEN_32,
+		MSP_ELEM_LEN_32,
+		MSP_ELEM_LEN_32,
+		MSP_ELEM_LEN_32,
+		MSP_DELAY_1,
+		MSP_DELAY_1,
+		MSP_RISING_EDGE,
+		MSP_FALLING_EDGE,
+		MSP_FSYNC_POL_ACT_LO,
+		MSP_FSYNC_POL_ACT_LO,
+		MSP_SWAP_NONE,
+		MSP_SWAP_NONE,
+		MSP_COMPRESS_MODE_LINEAR,
+		MSP_EXPAND_MODE_LINEAR,
+		MSP_FSYNC_IGNORE,
+		31,
+		15,
+		32,
+	}, { /* PCM */
+		MSP_DUAL_PHASE,
+		MSP_DUAL_PHASE,
+		MSP_PHASE2_START_MODE_FSYNC,
+		MSP_PHASE2_START_MODE_FSYNC,
+		MSP_BTF_MS_BIT_FIRST,
+		MSP_BTF_MS_BIT_FIRST,
+		MSP_FRAME_LEN_1,
+		MSP_FRAME_LEN_1,
+		MSP_FRAME_LEN_1,
+		MSP_FRAME_LEN_1,
+		MSP_ELEM_LEN_16,
+		MSP_ELEM_LEN_16,
+		MSP_ELEM_LEN_16,
+		MSP_ELEM_LEN_16,
+		MSP_DELAY_0,
+		MSP_DELAY_0,
+		MSP_RISING_EDGE,
+		MSP_FALLING_EDGE,
+		MSP_FSYNC_POL_ACT_HI,
+		MSP_FSYNC_POL_ACT_HI,
+		MSP_SWAP_NONE,
+		MSP_SWAP_NONE,
+		MSP_COMPRESS_MODE_LINEAR,
+		MSP_EXPAND_MODE_LINEAR,
+		MSP_FSYNC_IGNORE,
+		255,
+		0,
+		256,
+	}, { /* Companded PCM */
+		MSP_SINGLE_PHASE,
+		MSP_SINGLE_PHASE,
+		MSP_PHASE2_START_MODE_FSYNC,
+		MSP_PHASE2_START_MODE_FSYNC,
+		MSP_BTF_MS_BIT_FIRST,
+		MSP_BTF_MS_BIT_FIRST,
+		MSP_FRAME_LEN_1,
+		MSP_FRAME_LEN_1,
+		MSP_FRAME_LEN_1,
+		MSP_FRAME_LEN_1,
+		MSP_ELEM_LEN_8,
+		MSP_ELEM_LEN_8,
+		MSP_ELEM_LEN_8,
+		MSP_ELEM_LEN_8,
+		MSP_DELAY_0,
+		MSP_DELAY_0,
+		MSP_RISING_EDGE,
+		MSP_RISING_EDGE,
+		MSP_FSYNC_POL_ACT_HI,
+		MSP_FSYNC_POL_ACT_HI,
+		MSP_SWAP_NONE,
+		MSP_SWAP_NONE,
+		MSP_COMPRESS_MODE_LINEAR,
+		MSP_EXPAND_MODE_LINEAR,
+		MSP_FSYNC_IGNORE,
+		255,
+		0,
+		256,
+	},
+};
+
+static void set_prot_desc_tx(struct ux500_msp *msp,
+			struct msp_protdesc *protdesc,
+			enum msp_data_size data_size)
+{
+	u32 temp_reg = 0;
+
+	temp_reg |= MSP_P2_ENABLE_BIT(protdesc->tx_phase_mode);
+	temp_reg |= MSP_P2_START_MODE_BIT(protdesc->tx_phase2_start_mode);
+	temp_reg |= MSP_P1_FRAME_LEN_BITS(protdesc->tx_frame_len_1);
+	temp_reg |= MSP_P2_FRAME_LEN_BITS(protdesc->tx_frame_len_2);
+	if (msp->def_elem_len) {
+		temp_reg |= MSP_P1_ELEM_LEN_BITS(protdesc->tx_elem_len_1);
+		temp_reg |= MSP_P2_ELEM_LEN_BITS(protdesc->tx_elem_len_2);
+	} else {
+		temp_reg |= MSP_P1_ELEM_LEN_BITS(data_size);
+		temp_reg |= MSP_P2_ELEM_LEN_BITS(data_size);
+	}
+	temp_reg |= MSP_DATA_DELAY_BITS(protdesc->tx_data_delay);
+	temp_reg |= MSP_SET_ENDIANNES_BIT(protdesc->tx_byte_order);
+	temp_reg |= MSP_FSYNC_POL(protdesc->tx_fsync_pol);
+	temp_reg |= MSP_DATA_WORD_SWAP(protdesc->tx_half_word_swap);
+	temp_reg |= MSP_SET_COMPANDING_MODE(protdesc->compression_mode);
+	temp_reg |= MSP_SET_FSYNC_IGNORE(protdesc->frame_sync_ignore);
+
+	writel(temp_reg, msp->registers + MSP_TCF);
+}
+
+static void set_prot_desc_rx(struct ux500_msp *msp,
+			struct msp_protdesc *protdesc,
+			enum msp_data_size data_size)
+{
+	u32 temp_reg = 0;
+
+	temp_reg |= MSP_P2_ENABLE_BIT(protdesc->rx_phase_mode);
+	temp_reg |= MSP_P2_START_MODE_BIT(protdesc->rx_phase2_start_mode);
+	temp_reg |= MSP_P1_FRAME_LEN_BITS(protdesc->rx_frame_len_1);
+	temp_reg |= MSP_P2_FRAME_LEN_BITS(protdesc->rx_frame_len_2);
+	if (msp->def_elem_len) {
+		temp_reg |= MSP_P1_ELEM_LEN_BITS(protdesc->rx_elem_len_1);
+		temp_reg |= MSP_P2_ELEM_LEN_BITS(protdesc->rx_elem_len_2);
+	} else {
+		temp_reg |= MSP_P1_ELEM_LEN_BITS(data_size);
+		temp_reg |= MSP_P2_ELEM_LEN_BITS(data_size);
+	}
+
+	temp_reg |= MSP_DATA_DELAY_BITS(protdesc->rx_data_delay);
+	temp_reg |= MSP_SET_ENDIANNES_BIT(protdesc->rx_byte_order);
+	temp_reg |= MSP_FSYNC_POL(protdesc->rx_fsync_pol);
+	temp_reg |= MSP_DATA_WORD_SWAP(protdesc->rx_half_word_swap);
+	temp_reg |= MSP_SET_COMPANDING_MODE(protdesc->expansion_mode);
+	temp_reg |= MSP_SET_FSYNC_IGNORE(protdesc->frame_sync_ignore);
+
+	writel(temp_reg, msp->registers + MSP_RCF);
+}
+
+static int configure_protocol(struct ux500_msp *msp,
+			struct ux500_msp_config *config)
+{
+	struct msp_protdesc *protdesc;
+	enum msp_data_size data_size;
+	u32 temp_reg = 0;
+
+	data_size = config->data_size;
+	msp->def_elem_len = config->def_elem_len;
+	if (config->default_protdesc == 1) {
+		if (config->protocol >= MSP_INVALID_PROTOCOL) {
+			dev_err(msp->dev, "%s: ERROR: Invalid protocol!\n",
+				__func__);
+			return -EINVAL;
+		}
+		protdesc =
+		    (struct msp_protdesc *)&prot_descs[config->protocol];
+	} else {
+		protdesc = (struct msp_protdesc *)&config->protdesc;
+	}
+
+	if (data_size < MSP_DATA_BITS_DEFAULT || data_size > MSP_DATA_BITS_32) {
+		dev_err(msp->dev,
+			"%s: ERROR: Invalid data-size requested (data_size = %d)!\n",
+			__func__, data_size);
+		return -EINVAL;
+	}
+
+	if (config->direction & MSP_DIR_TX)
+		set_prot_desc_tx(msp, protdesc, data_size);
+	if (config->direction & MSP_DIR_RX)
+		set_prot_desc_rx(msp, protdesc, data_size);
+
+	/* The code below should not be separated. */
+	temp_reg = readl(msp->registers + MSP_GCR) & ~TX_CLK_POL_RISING;
+	temp_reg |= MSP_TX_CLKPOL_BIT(~protdesc->tx_clk_pol);
+	writel(temp_reg, msp->registers + MSP_GCR);
+	temp_reg = readl(msp->registers + MSP_GCR) & ~RX_CLK_POL_RISING;
+	temp_reg |= MSP_RX_CLKPOL_BIT(protdesc->rx_clk_pol);
+	writel(temp_reg, msp->registers + MSP_GCR);
+
+	return 0;
+}
+
+static int setup_bitclk(struct ux500_msp *msp, struct ux500_msp_config *config)
+{
+	u32 reg_val_GCR;
+	u32 frame_per = 0;
+	u32 sck_div = 0;
+	u32 frame_width = 0;
+	u32 temp_reg = 0;
+	struct msp_protdesc *protdesc = NULL;
+
+	reg_val_GCR = readl(msp->registers + MSP_GCR);
+	writel(reg_val_GCR & ~SRG_ENABLE, msp->registers + MSP_GCR);
+
+	if (config->default_protdesc)
+		protdesc =
+			(struct msp_protdesc *)&prot_descs[config->protocol];
+	else
+		protdesc = (struct msp_protdesc *)&config->protdesc;
+
+	switch (config->protocol) {
+	case MSP_PCM_PROTOCOL:
+	case MSP_PCM_COMPAND_PROTOCOL:
+		frame_width = protdesc->frame_width;
+		sck_div = config->f_inputclk / (config->frame_freq *
+			(protdesc->clocks_per_frame));
+		frame_per = protdesc->frame_period;
+		break;
+	case MSP_I2S_PROTOCOL:
+		frame_width = protdesc->frame_width;
+		sck_div = config->f_inputclk / (config->frame_freq *
+			(protdesc->clocks_per_frame));
+		frame_per = protdesc->frame_period;
+		break;
+	default:
+		dev_err(msp->dev, "%s: ERROR: Unknown protocol (%d)!\n",
+			__func__,
+			config->protocol);
+		return -EINVAL;
+	}
+
+	temp_reg = (sck_div - 1) & SCK_DIV_MASK;
+	temp_reg |= FRAME_WIDTH_BITS(frame_width);
+	temp_reg |= FRAME_PERIOD_BITS(frame_per);
+	writel(temp_reg, msp->registers + MSP_SRG);
+
+	msp->f_bitclk = (config->f_inputclk)/(sck_div + 1);
+
+	/* Enable bit-clock */
+	udelay(100);
+	reg_val_GCR = readl(msp->registers + MSP_GCR);
+	writel(reg_val_GCR | SRG_ENABLE, msp->registers + MSP_GCR);
+	udelay(100);
+
+	return 0;
+}
+
+static int configure_multichannel(struct ux500_msp *msp,
+				struct ux500_msp_config *config)
+{
+	struct msp_protdesc *protdesc;
+	struct msp_multichannel_config *mcfg;
+	u32 reg_val_MCR;
+
+	if (config->default_protdesc == 1) {
+		if (config->protocol >= MSP_INVALID_PROTOCOL) {
+			dev_err(msp->dev,
+				"%s: ERROR: Invalid protocol (%d)!\n",
+				__func__, config->protocol);
+			return -EINVAL;
+		}
+		protdesc = (struct msp_protdesc *)
+				&prot_descs[config->protocol];
+	} else {
+		protdesc = (struct msp_protdesc *)&config->protdesc;
+	}
+
+	mcfg = &config->multichannel_config;
+	if (mcfg->tx_multichannel_enable) {
+		if (protdesc->tx_phase_mode == MSP_SINGLE_PHASE) {
+			reg_val_MCR = readl(msp->registers + MSP_MCR);
+			writel(reg_val_MCR | (mcfg->tx_multichannel_enable ?
+						1 << TMCEN_BIT : 0),
+				msp->registers + MSP_MCR);
+			writel(mcfg->tx_channel_0_enable,
+				msp->registers + MSP_TCE0);
+			writel(mcfg->tx_channel_1_enable,
+				msp->registers + MSP_TCE1);
+			writel(mcfg->tx_channel_2_enable,
+				msp->registers + MSP_TCE2);
+			writel(mcfg->tx_channel_3_enable,
+				msp->registers + MSP_TCE3);
+		} else {
+			dev_err(msp->dev,
+				"%s: ERROR: Only single-phase supported (TX-mode: %d)!\n",
+				__func__, protdesc->tx_phase_mode);
+			return -EINVAL;
+		}
+	}
+	if (mcfg->rx_multichannel_enable) {
+		if (protdesc->rx_phase_mode == MSP_SINGLE_PHASE) {
+			reg_val_MCR = readl(msp->registers + MSP_MCR);
+			writel(reg_val_MCR | (mcfg->rx_multichannel_enable ?
+						1 << RMCEN_BIT : 0),
+				msp->registers + MSP_MCR);
+			writel(mcfg->rx_channel_0_enable,
+					msp->registers + MSP_RCE0);
+			writel(mcfg->rx_channel_1_enable,
+					msp->registers + MSP_RCE1);
+			writel(mcfg->rx_channel_2_enable,
+					msp->registers + MSP_RCE2);
+			writel(mcfg->rx_channel_3_enable,
+					msp->registers + MSP_RCE3);
+		} else {
+			dev_err(msp->dev,
+				"%s: ERROR: Only single-phase supported (RX-mode: %d)!\n",
+				__func__, protdesc->rx_phase_mode);
+			return -EINVAL;
+		}
+		if (mcfg->rx_comparison_enable_mode) {
+			reg_val_MCR = readl(msp->registers + MSP_MCR);
+			writel(reg_val_MCR |
+				(mcfg->rx_comparison_enable_mode << RCMPM_BIT),
+				msp->registers + MSP_MCR);
+
+			writel(mcfg->comparison_mask,
+					msp->registers + MSP_RCM);
+			writel(mcfg->comparison_value,
+					msp->registers + MSP_RCV);
+
+		}
+	}
+
+	return 0;
+}
+
+static int enable_msp(struct ux500_msp *msp, struct ux500_msp_config *config)
+{
+	int status = 0;
+	u32 reg_val_DMACR, reg_val_GCR;
+
+	/* Check msp state whether in RUN or CONFIGURED Mode */
+	if ((msp->msp_state == MSP_STATE_IDLE) && (msp->plat_init)) {
+		status = msp->plat_init();
+		if (status) {
+			dev_err(msp->dev, "%s: ERROR: Failed to init MSP (%d)!\n",
+				__func__, status);
+			return status;
+		}
+	}
+
+	/* Configure msp with protocol dependent settings */
+	configure_protocol(msp, config);
+	setup_bitclk(msp, config);
+	if (config->multichannel_configured == 1) {
+		status = configure_multichannel(msp, config);
+		if (status)
+			dev_warn(msp->dev,
+				"%s: WARN: configure_multichannel failed (%d)!\n",
+				__func__, status);
+	}
+
+	/* Make sure the correct DMA-directions are configured */
+	if ((config->direction & MSP_DIR_RX) && (!msp->dma_cfg_rx)) {
+		dev_err(msp->dev, "%s: ERROR: MSP RX-mode is not configured!",
+			__func__);
+		return -EINVAL;
+	}
+	if ((config->direction == MSP_DIR_TX) && (!msp->dma_cfg_tx)) {
+		dev_err(msp->dev, "%s: ERROR: MSP TX-mode is not configured!",
+			__func__);
+		return -EINVAL;
+	}
+
+	reg_val_DMACR = readl(msp->registers + MSP_DMACR);
+	if (config->direction & MSP_DIR_RX)
+		reg_val_DMACR |= RX_DMA_ENABLE;
+	if (config->direction & MSP_DIR_TX)
+		reg_val_DMACR |= TX_DMA_ENABLE;
+	writel(reg_val_DMACR, msp->registers + MSP_DMACR);
+
+	writel(config->iodelay, msp->registers + MSP_IODLY);
+
+	/* Enable frame generation logic */
+	reg_val_GCR = readl(msp->registers + MSP_GCR);
+	writel(reg_val_GCR | FRAME_GEN_ENABLE, msp->registers + MSP_GCR);
+
+	return status;
+}
+
+static void flush_fifo_rx(struct ux500_msp *msp)
+{
+	u32 reg_val_DR, reg_val_GCR, reg_val_FLR;
+	u32 limit = 32;
+
+	reg_val_GCR = readl(msp->registers + MSP_GCR);
+	writel(reg_val_GCR | RX_ENABLE, msp->registers + MSP_GCR);
+
+	reg_val_FLR = readl(msp->registers + MSP_FLR);
+	while (!(reg_val_FLR & RX_FIFO_EMPTY) && limit--) {
+		reg_val_DR = readl(msp->registers + MSP_DR);
+		reg_val_FLR = readl(msp->registers + MSP_FLR);
+	}
+
+	writel(reg_val_GCR, msp->registers + MSP_GCR);
+}
+
+static void flush_fifo_tx(struct ux500_msp *msp)
+{
+	u32 reg_val_TSTDR, reg_val_GCR, reg_val_FLR;
+	u32 limit = 32;
+
+	reg_val_GCR = readl(msp->registers + MSP_GCR);
+	writel(reg_val_GCR | TX_ENABLE, msp->registers + MSP_GCR);
+	writel(MSP_ITCR_ITEN | MSP_ITCR_TESTFIFO, msp->registers + MSP_ITCR);
+
+	reg_val_FLR = readl(msp->registers + MSP_FLR);
+	while (!(reg_val_FLR & TX_FIFO_EMPTY) && limit--) {
+		reg_val_TSTDR = readl(msp->registers + MSP_TSTDR);
+		reg_val_FLR = readl(msp->registers + MSP_FLR);
+	}
+	writel(0x0, msp->registers + MSP_ITCR);
+	writel(reg_val_GCR, msp->registers + MSP_GCR);
+}
+
+int ux500_msp_i2s_open(struct ux500_msp *msp,
+		struct ux500_msp_config *config)
+{
+	u32 old_reg, new_reg, mask;
+	int res;
+	unsigned int tx_sel, rx_sel, tx_busy, rx_busy;
+
+	if (in_interrupt()) {
+		dev_err(msp->dev,
+			"%s: ERROR: Open called in interrupt context!\n",
+			__func__);
+		return -1;
+	}
+
+	tx_sel = (config->direction & MSP_DIR_TX) > 0;
+	rx_sel = (config->direction & MSP_DIR_RX) > 0;
+	if (!tx_sel && !rx_sel) {
+		dev_err(msp->dev, "%s: Error: No direction selected!\n",
+			__func__);
+		return -EINVAL;
+	}
+
+	tx_busy = (msp->dir_busy & MSP_DIR_TX) > 0;
+	rx_busy = (msp->dir_busy & MSP_DIR_RX) > 0;
+	if (tx_busy && tx_sel) {
+		dev_err(msp->dev, "%s: Error: TX is in use!\n", __func__);
+		return -EBUSY;
+	}
+	if (rx_busy && rx_sel) {
+		dev_err(msp->dev, "%s: Error: RX is in use!\n", __func__);
+		return -EBUSY;
+	}
+
+	msp->dir_busy |= (tx_sel ? MSP_DIR_TX : 0) | (rx_sel ? MSP_DIR_RX : 0);
+
+	/* First do the global config register */
+	mask = RX_CLK_SEL_MASK | TX_CLK_SEL_MASK | RX_FSYNC_MASK |
+	    TX_FSYNC_MASK | RX_SYNC_SEL_MASK | TX_SYNC_SEL_MASK |
+	    RX_FIFO_ENABLE_MASK | TX_FIFO_ENABLE_MASK | SRG_CLK_SEL_MASK |
+	    LOOPBACK_MASK | TX_EXTRA_DELAY_MASK;
+
+	new_reg = (config->tx_clk_sel | config->rx_clk_sel |
+		config->rx_fsync_pol | config->tx_fsync_pol |
+		config->rx_fsync_sel | config->tx_fsync_sel |
+		config->rx_fifo_config | config->tx_fifo_config |
+		config->srg_clk_sel | config->loopback_enable |
+		config->tx_data_enable);
+
+	old_reg = readl(msp->registers + MSP_GCR);
+	old_reg &= ~mask;
+	new_reg |= old_reg;
+	writel(new_reg, msp->registers + MSP_GCR);
+
+	res = enable_msp(msp, config);
+	if (res < 0) {
+		dev_err(msp->dev, "%s: ERROR: enable_msp failed (%d)!\n",
+			__func__, res);
+		return -EBUSY;
+	}
+	if (config->loopback_enable & 0x80)
+		msp->loopback_enable = 1;
+
+	/* Flush FIFOs */
+	flush_fifo_tx(msp);
+	flush_fifo_rx(msp);
+
+	msp->msp_state = MSP_STATE_CONFIGURED;
+	return 0;
+}
+
+static void disable_msp_rx(struct ux500_msp *msp)
+{
+	u32 reg_val_GCR, reg_val_DMACR, reg_val_IMSC;
+
+	reg_val_GCR = readl(msp->registers + MSP_GCR);
+	writel(reg_val_GCR & ~RX_ENABLE, msp->registers + MSP_GCR);
+	reg_val_DMACR = readl(msp->registers + MSP_DMACR);
+	writel(reg_val_DMACR & ~RX_DMA_ENABLE, msp->registers + MSP_DMACR);
+	reg_val_IMSC = readl(msp->registers + MSP_IMSC);
+	writel(reg_val_IMSC &
+			~(RX_SERVICE_INT | RX_OVERRUN_ERROR_INT),
+			msp->registers + MSP_IMSC);
+
+	msp->dir_busy &= ~MSP_DIR_RX;
+}
+
+static void disable_msp_tx(struct ux500_msp *msp)
+{
+	u32 reg_val_GCR, reg_val_DMACR, reg_val_IMSC;
+
+	reg_val_GCR = readl(msp->registers + MSP_GCR);
+	writel(reg_val_GCR & ~TX_ENABLE, msp->registers + MSP_GCR);
+	reg_val_DMACR = readl(msp->registers + MSP_DMACR);
+	writel(reg_val_DMACR & ~TX_DMA_ENABLE, msp->registers + MSP_DMACR);
+	reg_val_IMSC = readl(msp->registers + MSP_IMSC);
+	writel(reg_val_IMSC &
+			~(TX_SERVICE_INT | TX_UNDERRUN_ERR_INT),
+			msp->registers + MSP_IMSC);
+
+	msp->dir_busy &= ~MSP_DIR_TX;
+}
+
+static int disable_msp(struct ux500_msp *msp, unsigned int dir)
+{
+	u32 reg_val_GCR;
+	int status = 0;
+	unsigned int disable_tx, disable_rx;
+
+	reg_val_GCR = readl(msp->registers + MSP_GCR);
+	disable_tx = dir & MSP_DIR_TX;
+	disable_rx = dir & MSP_DIR_TX;
+	if (disable_tx && disable_rx) {
+		reg_val_GCR = readl(msp->registers + MSP_GCR);
+		writel(reg_val_GCR | LOOPBACK_MASK,
+				msp->registers + MSP_GCR);
+
+		/* Flush TX-FIFO */
+		flush_fifo_tx(msp);
+
+		/* Disable TX-channel */
+		writel((readl(msp->registers + MSP_GCR) &
+			       (~TX_ENABLE)), msp->registers + MSP_GCR);
+
+		/* Flush RX-FIFO */
+		flush_fifo_rx(msp);
+
+		/* Disable Loopback and Receive channel */
+		writel((readl(msp->registers + MSP_GCR) &
+				(~(RX_ENABLE | LOOPBACK_MASK))),
+				msp->registers + MSP_GCR);
+
+		disable_msp_tx(msp);
+		disable_msp_rx(msp);
+	} else if (disable_tx)
+		disable_msp_tx(msp);
+	else if (disable_rx)
+		disable_msp_rx(msp);
+
+	return status;
+}
+
+int ux500_msp_i2s_trigger(struct ux500_msp *msp, int cmd, int direction)
+{
+	u32 reg_val_GCR, enable_bit;
+
+	if (msp->msp_state == MSP_STATE_IDLE) {
+		dev_err(msp->dev, "%s: ERROR: MSP is not configured!\n",
+			__func__);
+		return -EINVAL;
+	}
+
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_START:
+	case SNDRV_PCM_TRIGGER_RESUME:
+	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+		if (direction == SNDRV_PCM_STREAM_PLAYBACK)
+			enable_bit = TX_ENABLE;
+		else
+			enable_bit = RX_ENABLE;
+		reg_val_GCR = readl(msp->registers + MSP_GCR);
+		writel(reg_val_GCR | enable_bit, msp->registers + MSP_GCR);
+		break;
+
+	case SNDRV_PCM_TRIGGER_STOP:
+	case SNDRV_PCM_TRIGGER_SUSPEND:
+	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+		if (direction == SNDRV_PCM_STREAM_PLAYBACK)
+			disable_msp_tx(msp);
+		else
+			disable_msp_rx(msp);
+		break;
+	default:
+		return -EINVAL;
+		break;
+	}
+
+	return 0;
+}
+
+int ux500_msp_i2s_close(struct ux500_msp *msp, unsigned int dir)
+{
+	int status = 0;
+
+	dev_dbg(msp->dev, "%s: Enter (dir = 0x%01x).\n", __func__, dir);
+
+	status = disable_msp(msp, dir);
+	if (msp->dir_busy == 0) {
+		/* disable sample rate and frame generators */
+		msp->msp_state = MSP_STATE_IDLE;
+		writel((readl(msp->registers + MSP_GCR) &
+			       (~(FRAME_GEN_ENABLE | SRG_ENABLE))),
+			      msp->registers + MSP_GCR);
+		if (msp->plat_exit)
+			status = msp->plat_exit();
+			if (status)
+				dev_warn(msp->dev,
+					"%s: WARN: ux500_msp_i2s_exit failed (%d)!\n",
+					__func__, status);
+		writel(0, msp->registers + MSP_GCR);
+		writel(0, msp->registers + MSP_TCF);
+		writel(0, msp->registers + MSP_RCF);
+		writel(0, msp->registers + MSP_DMACR);
+		writel(0, msp->registers + MSP_SRG);
+		writel(0, msp->registers + MSP_MCR);
+		writel(0, msp->registers + MSP_RCM);
+		writel(0, msp->registers + MSP_RCV);
+		writel(0, msp->registers + MSP_TCE0);
+		writel(0, msp->registers + MSP_TCE1);
+		writel(0, msp->registers + MSP_TCE2);
+		writel(0, msp->registers + MSP_TCE3);
+		writel(0, msp->registers + MSP_RCE0);
+		writel(0, msp->registers + MSP_RCE1);
+		writel(0, msp->registers + MSP_RCE2);
+		writel(0, msp->registers + MSP_RCE3);
+	}
+
+	return status;
+
+}
+
+int ux500_msp_i2s_init_msp(struct platform_device *pdev,
+			struct ux500_msp **msp_p,
+			struct msp_i2s_platform_data *platform_data)
+{
+	int ret = 0;
+	struct resource *res = NULL;
+	struct i2s_controller *i2s_cont;
+	struct ux500_msp *msp;
+
+	dev_dbg(&pdev->dev, "%s: Enter (name: %s, id: %d).\n", __func__,
+		pdev->name, platform_data->id);
+
+	*msp_p = devm_kzalloc(&pdev->dev, sizeof(struct ux500_msp), GFP_KERNEL);
+	msp = *msp_p;
+
+	msp->id = platform_data->id;
+	msp->dev = &pdev->dev;
+	msp->plat_init = platform_data->msp_i2s_init;
+	msp->plat_exit = platform_data->msp_i2s_exit;
+	msp->dma_cfg_rx = platform_data->msp_i2s_dma_rx;
+	msp->dma_cfg_tx = platform_data->msp_i2s_dma_tx;
+
+	res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+	if (res == NULL) {
+		dev_err(&pdev->dev, "%s: ERROR: Unable to get resource!\n",
+			__func__);
+		ret = -ENOMEM;
+		goto err_res;
+	}
+
+	msp->registers = ioremap(res->start, (res->end - res->start + 1));
+	if (msp->registers == NULL) {
+		dev_err(&pdev->dev, "%s: ERROR: ioremap failed!\n", __func__);
+		ret = -ENOMEM;
+		goto err_res;
+	}
+
+	msp->msp_state = MSP_STATE_IDLE;
+	msp->loopback_enable = 0;
+
+	/* I2S-controller is allocated and added in I2S controller class. */
+	i2s_cont = devm_kzalloc(&pdev->dev, sizeof(*i2s_cont), GFP_KERNEL);
+	if (!i2s_cont) {
+		dev_err(&pdev->dev,
+			"%s: ERROR: Failed to allocate I2S-controller!\n",
+			__func__);
+		goto err_i2s_cont;
+	}
+	i2s_cont->dev.parent = &pdev->dev;
+	i2s_cont->data = (void *)msp;
+	i2s_cont->id = (s16)msp->id;
+	snprintf(i2s_cont->name, sizeof(i2s_cont->name), "ux500-msp-i2s.%04x",
+		msp->id);
+	dev_dbg(&pdev->dev, "I2S device-name: '%s'\n", i2s_cont->name);
+	msp->i2s_cont = i2s_cont;
+
+	return 0;
+
+err_i2s_cont:
+	iounmap(msp->registers);
+
+err_res:
+	devm_kfree(&pdev->dev, msp);
+
+	return ret;
+}
+
+void ux500_msp_i2s_cleanup_msp(struct platform_device *pdev,
+			struct ux500_msp *msp)
+{
+	dev_dbg(msp->dev, "%s: Enter (id = %d).\n", __func__, msp->id);
+
+	device_unregister(&msp->i2s_cont->dev);
+	devm_kfree(&pdev->dev, msp->i2s_cont);
+
+	iounmap(msp->registers);
+
+	devm_kfree(&pdev->dev, msp);
+}
+
+MODULE_LICENSE("GPLv2");
diff --git a/sound/soc/ux500/ux500_msp_i2s.h b/sound/soc/ux500/ux500_msp_i2s.h
new file mode 100644
index 000000000000..7f71b4a0d4bc
--- /dev/null
+++ b/sound/soc/ux500/ux500_msp_i2s.h
@@ -0,0 +1,553 @@
+/*
+ * Copyright (C) ST-Ericsson SA 2012
+ *
+ * Author: Ola Lilja <ola.o.lilja@stericsson.com>,
+ *         for ST-Ericsson.
+ *
+ * License terms:
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as published
+ * by the Free Software Foundation.
+ */
+
+
+#ifndef UX500_MSP_I2S_H
+#define UX500_MSP_I2S_H
+
+#include <linux/platform_device.h>
+
+#include <mach/board-mop500-msp.h>
+
+#define MSP_INPUT_FREQ_APB 48000000
+
+/*** Stereo mode. Used for APB data accesses as 16 bits accesses (mono),
+ *   32 bits accesses (stereo).
+ ***/
+enum msp_stereo_mode {
+	MSP_MONO,
+	MSP_STEREO
+};
+
+/* Direction (Transmit/Receive mode) */
+enum msp_direction {
+	MSP_TX = 1,
+	MSP_RX = 2
+};
+
+/* Transmit and receive configuration register */
+#define MSP_BIG_ENDIAN           0x00000000
+#define MSP_LITTLE_ENDIAN        0x00001000
+#define MSP_UNEXPECTED_FS_ABORT  0x00000000
+#define MSP_UNEXPECTED_FS_IGNORE 0x00008000
+#define MSP_NON_MODE_BIT_MASK    0x00009000
+
+/* Global configuration register */
+#define RX_ENABLE             0x00000001
+#define RX_FIFO_ENABLE        0x00000002
+#define RX_SYNC_SRG           0x00000010
+#define RX_CLK_POL_RISING     0x00000020
+#define RX_CLK_SEL_SRG        0x00000040
+#define TX_ENABLE             0x00000100
+#define TX_FIFO_ENABLE        0x00000200
+#define TX_SYNC_SRG_PROG      0x00001800
+#define TX_SYNC_SRG_AUTO      0x00001000
+#define TX_CLK_POL_RISING     0x00002000
+#define TX_CLK_SEL_SRG        0x00004000
+#define TX_EXTRA_DELAY_ENABLE 0x00008000
+#define SRG_ENABLE            0x00010000
+#define FRAME_GEN_ENABLE      0x00100000
+#define SRG_CLK_SEL_APB       0x00000000
+#define RX_FIFO_SYNC_HI       0x00000000
+#define TX_FIFO_SYNC_HI       0x00000000
+#define SPI_CLK_MODE_NORMAL   0x00000000
+
+#define MSP_FRAME_SIZE_AUTO -1
+
+#define MSP_DR		0x00
+#define MSP_GCR		0x04
+#define MSP_TCF		0x08
+#define MSP_RCF		0x0c
+#define MSP_SRG		0x10
+#define MSP_FLR		0x14
+#define MSP_DMACR	0x18
+
+#define MSP_IMSC	0x20
+#define MSP_RIS		0x24
+#define MSP_MIS		0x28
+#define MSP_ICR		0x2c
+#define MSP_MCR		0x30
+#define MSP_RCV		0x34
+#define MSP_RCM		0x38
+
+#define MSP_TCE0	0x40
+#define MSP_TCE1	0x44
+#define MSP_TCE2	0x48
+#define MSP_TCE3	0x4c
+
+#define MSP_RCE0	0x60
+#define MSP_RCE1	0x64
+#define MSP_RCE2	0x68
+#define MSP_RCE3	0x6c
+#define MSP_IODLY	0x70
+
+#define MSP_ITCR	0x80
+#define MSP_ITIP	0x84
+#define MSP_ITOP	0x88
+#define MSP_TSTDR	0x8c
+
+#define MSP_PID0	0xfe0
+#define MSP_PID1	0xfe4
+#define MSP_PID2	0xfe8
+#define MSP_PID3	0xfec
+
+#define MSP_CID0	0xff0
+#define MSP_CID1	0xff4
+#define MSP_CID2	0xff8
+#define MSP_CID3	0xffc
+
+/* Protocol dependant parameters list */
+#define RX_ENABLE_MASK		BIT(0)
+#define RX_FIFO_ENABLE_MASK	BIT(1)
+#define RX_FSYNC_MASK		BIT(2)
+#define DIRECT_COMPANDING_MASK	BIT(3)
+#define RX_SYNC_SEL_MASK	BIT(4)
+#define RX_CLK_POL_MASK		BIT(5)
+#define RX_CLK_SEL_MASK		BIT(6)
+#define LOOPBACK_MASK		BIT(7)
+#define TX_ENABLE_MASK		BIT(8)
+#define TX_FIFO_ENABLE_MASK	BIT(9)
+#define TX_FSYNC_MASK		BIT(10)
+#define TX_MSP_TDR_TSR		BIT(11)
+#define TX_SYNC_SEL_MASK	(BIT(12) | BIT(11))
+#define TX_CLK_POL_MASK		BIT(13)
+#define TX_CLK_SEL_MASK		BIT(14)
+#define TX_EXTRA_DELAY_MASK	BIT(15)
+#define SRG_ENABLE_MASK		BIT(16)
+#define SRG_CLK_POL_MASK	BIT(17)
+#define SRG_CLK_SEL_MASK	(BIT(19) | BIT(18))
+#define FRAME_GEN_EN_MASK	BIT(20)
+#define SPI_CLK_MODE_MASK	(BIT(22) | BIT(21))
+#define SPI_BURST_MODE_MASK	BIT(23)
+
+#define RXEN_SHIFT		0
+#define RFFEN_SHIFT		1
+#define RFSPOL_SHIFT		2
+#define DCM_SHIFT		3
+#define RFSSEL_SHIFT		4
+#define RCKPOL_SHIFT		5
+#define RCKSEL_SHIFT		6
+#define LBM_SHIFT		7
+#define TXEN_SHIFT		8
+#define TFFEN_SHIFT		9
+#define TFSPOL_SHIFT		10
+#define TFSSEL_SHIFT		11
+#define TCKPOL_SHIFT		13
+#define TCKSEL_SHIFT		14
+#define TXDDL_SHIFT		15
+#define SGEN_SHIFT		16
+#define SCKPOL_SHIFT		17
+#define SCKSEL_SHIFT		18
+#define FGEN_SHIFT		20
+#define SPICKM_SHIFT		21
+#define TBSWAP_SHIFT		28
+
+#define RCKPOL_MASK		BIT(0)
+#define TCKPOL_MASK		BIT(0)
+#define SPICKM_MASK		(BIT(1) | BIT(0))
+#define MSP_RX_CLKPOL_BIT(n)     ((n & RCKPOL_MASK) << RCKPOL_SHIFT)
+#define MSP_TX_CLKPOL_BIT(n)     ((n & TCKPOL_MASK) << TCKPOL_SHIFT)
+
+#define P1ELEN_SHIFT		0
+#define P1FLEN_SHIFT		3
+#define DTYP_SHIFT		10
+#define ENDN_SHIFT		12
+#define DDLY_SHIFT		13
+#define FSIG_SHIFT		15
+#define P2ELEN_SHIFT		16
+#define P2FLEN_SHIFT		19
+#define P2SM_SHIFT		26
+#define P2EN_SHIFT		27
+#define FSYNC_SHIFT		15
+
+#define P1ELEN_MASK		0x00000007
+#define P2ELEN_MASK		0x00070000
+#define P1FLEN_MASK		0x00000378
+#define P2FLEN_MASK		0x03780000
+#define DDLY_MASK		0x00003000
+#define DTYP_MASK		0x00000600
+#define P2SM_MASK		0x04000000
+#define P2EN_MASK		0x08000000
+#define ENDN_MASK		0x00001000
+#define TFSPOL_MASK		0x00000400
+#define TBSWAP_MASK		0x30000000
+#define COMPANDING_MODE_MASK	0x00000c00
+#define FSYNC_MASK		0x00008000
+
+#define MSP_P1_ELEM_LEN_BITS(n)		(n & P1ELEN_MASK)
+#define MSP_P2_ELEM_LEN_BITS(n)		(((n) << P2ELEN_SHIFT) & P2ELEN_MASK)
+#define MSP_P1_FRAME_LEN_BITS(n)	(((n) << P1FLEN_SHIFT) & P1FLEN_MASK)
+#define MSP_P2_FRAME_LEN_BITS(n)	(((n) << P2FLEN_SHIFT) & P2FLEN_MASK)
+#define MSP_DATA_DELAY_BITS(n)		(((n) << DDLY_SHIFT) & DDLY_MASK)
+#define MSP_DATA_TYPE_BITS(n)		(((n) << DTYP_SHIFT) & DTYP_MASK)
+#define MSP_P2_START_MODE_BIT(n)	((n << P2SM_SHIFT) & P2SM_MASK)
+#define MSP_P2_ENABLE_BIT(n)		((n << P2EN_SHIFT) & P2EN_MASK)
+#define MSP_SET_ENDIANNES_BIT(n)	((n << ENDN_SHIFT) & ENDN_MASK)
+#define MSP_FSYNC_POL(n)		((n << TFSPOL_SHIFT) & TFSPOL_MASK)
+#define MSP_DATA_WORD_SWAP(n)		((n << TBSWAP_SHIFT) & TBSWAP_MASK)
+#define MSP_SET_COMPANDING_MODE(n)	((n << DTYP_SHIFT) & \
+						COMPANDING_MODE_MASK)
+#define MSP_SET_FSYNC_IGNORE(n)		((n << FSYNC_SHIFT) & FSYNC_MASK)
+
+/* Flag register */
+#define RX_BUSY			BIT(0)
+#define RX_FIFO_EMPTY		BIT(1)
+#define RX_FIFO_FULL		BIT(2)
+#define TX_BUSY			BIT(3)
+#define TX_FIFO_EMPTY		BIT(4)
+#define TX_FIFO_FULL		BIT(5)
+
+#define RBUSY_SHIFT		0
+#define RFE_SHIFT		1
+#define RFU_SHIFT		2
+#define TBUSY_SHIFT		3
+#define TFE_SHIFT		4
+#define TFU_SHIFT		5
+
+/* Multichannel control register */
+#define RMCEN_SHIFT		0
+#define RMCSF_SHIFT		1
+#define RCMPM_SHIFT		3
+#define TMCEN_SHIFT		5
+#define TNCSF_SHIFT		6
+
+/* Sample rate generator register */
+#define SCKDIV_SHIFT		0
+#define FRWID_SHIFT		10
+#define FRPER_SHIFT		16
+
+#define SCK_DIV_MASK		0x0000003FF
+#define FRAME_WIDTH_BITS(n)	(((n) << FRWID_SHIFT)  & 0x0000FC00)
+#define FRAME_PERIOD_BITS(n)	(((n) << FRPER_SHIFT) & 0x1FFF0000)
+
+/* DMA controller register */
+#define RX_DMA_ENABLE		BIT(0)
+#define TX_DMA_ENABLE		BIT(1)
+
+#define RDMAE_SHIFT		0
+#define TDMAE_SHIFT		1
+
+/* Interrupt Register */
+#define RX_SERVICE_INT		BIT(0)
+#define RX_OVERRUN_ERROR_INT	BIT(1)
+#define RX_FSYNC_ERR_INT	BIT(2)
+#define RX_FSYNC_INT		BIT(3)
+#define TX_SERVICE_INT		BIT(4)
+#define TX_UNDERRUN_ERR_INT	BIT(5)
+#define TX_FSYNC_ERR_INT	BIT(6)
+#define TX_FSYNC_INT		BIT(7)
+#define ALL_INT			0x000000ff
+
+/* MSP test control register */
+#define MSP_ITCR_ITEN		BIT(0)
+#define MSP_ITCR_TESTFIFO	BIT(1)
+
+#define RMCEN_BIT   0
+#define RMCSF_BIT   1
+#define RCMPM_BIT   3
+#define TMCEN_BIT   5
+#define TNCSF_BIT   6
+
+/* Single or dual phase mode */
+enum msp_phase_mode {
+	MSP_SINGLE_PHASE,
+	MSP_DUAL_PHASE
+};
+
+/* Frame length */
+enum msp_frame_length {
+	MSP_FRAME_LEN_1 = 0,
+	MSP_FRAME_LEN_2 = 1,
+	MSP_FRAME_LEN_4 = 3,
+	MSP_FRAME_LEN_8 = 7,
+	MSP_FRAME_LEN_12 = 11,
+	MSP_FRAME_LEN_16 = 15,
+	MSP_FRAME_LEN_20 = 19,
+	MSP_FRAME_LEN_32 = 31,
+	MSP_FRAME_LEN_48 = 47,
+	MSP_FRAME_LEN_64 = 63
+};
+
+/* Element length */
+enum msp_elem_length {
+	MSP_ELEM_LEN_8 = 0,
+	MSP_ELEM_LEN_10 = 1,
+	MSP_ELEM_LEN_12 = 2,
+	MSP_ELEM_LEN_14 = 3,
+	MSP_ELEM_LEN_16 = 4,
+	MSP_ELEM_LEN_20 = 5,
+	MSP_ELEM_LEN_24 = 6,
+	MSP_ELEM_LEN_32 = 7
+};
+
+enum msp_data_xfer_width {
+	MSP_DATA_TRANSFER_WIDTH_BYTE,
+	MSP_DATA_TRANSFER_WIDTH_HALFWORD,
+	MSP_DATA_TRANSFER_WIDTH_WORD
+};
+
+enum msp_frame_sync {
+	MSP_FSYNC_UNIGNORE = 0,
+	MSP_FSYNC_IGNORE = 1,
+};
+
+enum msp_phase2_start_mode {
+	MSP_PHASE2_START_MODE_IMEDIATE,
+	MSP_PHASE2_START_MODE_FSYNC
+};
+
+enum msp_btf {
+	MSP_BTF_MS_BIT_FIRST = 0,
+	MSP_BTF_LS_BIT_FIRST = 1
+};
+
+enum msp_fsync_pol {
+	MSP_FSYNC_POL_ACT_HI = 0,
+	MSP_FSYNC_POL_ACT_LO = 1
+};
+
+/* Data delay (in bit clock cycles) */
+enum msp_delay {
+	MSP_DELAY_0 = 0,
+	MSP_DELAY_1 = 1,
+	MSP_DELAY_2 = 2,
+	MSP_DELAY_3 = 3
+};
+
+/* Configurations of clocks (transmit, receive or sample rate generator) */
+enum msp_edge {
+	MSP_FALLING_EDGE = 0,
+	MSP_RISING_EDGE = 1,
+};
+
+enum msp_hws {
+	MSP_SWAP_NONE = 0,
+	MSP_SWAP_BYTE_PER_WORD = 1,
+	MSP_SWAP_BYTE_PER_HALF_WORD = 2,
+	MSP_SWAP_HALF_WORD_PER_WORD = 3
+};
+
+enum msp_compress_mode {
+	MSP_COMPRESS_MODE_LINEAR = 0,
+	MSP_COMPRESS_MODE_MU_LAW = 2,
+	MSP_COMPRESS_MODE_A_LAW = 3
+};
+
+enum msp_spi_burst_mode {
+	MSP_SPI_BURST_MODE_DISABLE = 0,
+	MSP_SPI_BURST_MODE_ENABLE = 1
+};
+
+enum msp_expand_mode {
+	MSP_EXPAND_MODE_LINEAR = 0,
+	MSP_EXPAND_MODE_LINEAR_SIGNED = 1,
+	MSP_EXPAND_MODE_MU_LAW = 2,
+	MSP_EXPAND_MODE_A_LAW = 3
+};
+
+#define MSP_FRAME_PERIOD_IN_MONO_MODE 256
+#define MSP_FRAME_PERIOD_IN_STEREO_MODE 32
+#define MSP_FRAME_WIDTH_IN_STEREO_MODE 16
+
+enum msp_protocol {
+	MSP_I2S_PROTOCOL,
+	MSP_PCM_PROTOCOL,
+	MSP_PCM_COMPAND_PROTOCOL,
+	MSP_INVALID_PROTOCOL
+};
+
+/*
+ * No of registers to backup during
+ * suspend resume
+ */
+#define MAX_MSP_BACKUP_REGS 36
+
+enum enum_i2s_controller {
+	MSP_0_I2S_CONTROLLER = 0,
+	MSP_1_I2S_CONTROLLER,
+	MSP_2_I2S_CONTROLLER,
+	MSP_3_I2S_CONTROLLER,
+};
+
+enum i2s_direction_t {
+	MSP_DIR_TX = 0x01,
+	MSP_DIR_RX = 0x02,
+};
+
+enum msp_data_size {
+	MSP_DATA_BITS_DEFAULT = -1,
+	MSP_DATA_BITS_8 = 0x00,
+	MSP_DATA_BITS_10,
+	MSP_DATA_BITS_12,
+	MSP_DATA_BITS_14,
+	MSP_DATA_BITS_16,
+	MSP_DATA_BITS_20,
+	MSP_DATA_BITS_24,
+	MSP_DATA_BITS_32,
+};
+
+enum msp_state {
+	MSP_STATE_IDLE = 0,
+	MSP_STATE_CONFIGURED = 1,
+	MSP_STATE_RUNNING = 2,
+};
+
+enum msp_rx_comparison_enable_mode {
+	MSP_COMPARISON_DISABLED = 0,
+	MSP_COMPARISON_NONEQUAL_ENABLED = 2,
+	MSP_COMPARISON_EQUAL_ENABLED = 3
+};
+
+struct msp_multichannel_config {
+	bool rx_multichannel_enable;
+	bool tx_multichannel_enable;
+	enum msp_rx_comparison_enable_mode rx_comparison_enable_mode;
+	u8 padding;
+	u32 comparison_value;
+	u32 comparison_mask;
+	u32 rx_channel_0_enable;
+	u32 rx_channel_1_enable;
+	u32 rx_channel_2_enable;
+	u32 rx_channel_3_enable;
+	u32 tx_channel_0_enable;
+	u32 tx_channel_1_enable;
+	u32 tx_channel_2_enable;
+	u32 tx_channel_3_enable;
+};
+
+struct msp_protdesc {
+	u32 rx_phase_mode;
+	u32 tx_phase_mode;
+	u32 rx_phase2_start_mode;
+	u32 tx_phase2_start_mode;
+	u32 rx_byte_order;
+	u32 tx_byte_order;
+	u32 rx_frame_len_1;
+	u32 rx_frame_len_2;
+	u32 tx_frame_len_1;
+	u32 tx_frame_len_2;
+	u32 rx_elem_len_1;
+	u32 rx_elem_len_2;
+	u32 tx_elem_len_1;
+	u32 tx_elem_len_2;
+	u32 rx_data_delay;
+	u32 tx_data_delay;
+	u32 rx_clk_pol;
+	u32 tx_clk_pol;
+	u32 rx_fsync_pol;
+	u32 tx_fsync_pol;
+	u32 rx_half_word_swap;
+	u32 tx_half_word_swap;
+	u32 compression_mode;
+	u32 expansion_mode;
+	u32 frame_sync_ignore;
+	u32 frame_period;
+	u32 frame_width;
+	u32 clocks_per_frame;
+};
+
+struct i2s_message {
+	enum i2s_direction_t i2s_direction;
+	void *txdata;
+	void *rxdata;
+	size_t txbytes;
+	size_t rxbytes;
+	int dma_flag;
+	int tx_offset;
+	int rx_offset;
+	bool cyclic_dma;
+	dma_addr_t buf_addr;
+	size_t buf_len;
+	size_t period_len;
+};
+
+struct i2s_controller {
+	struct module *owner;
+	unsigned int id;
+	unsigned int class;
+	const struct i2s_algorithm *algo; /* the algorithm to access the bus */
+	void *data;
+	struct mutex bus_lock;
+	struct device dev; /* the controller device */
+	char name[48];
+};
+
+struct ux500_msp_config {
+	unsigned int f_inputclk;
+	unsigned int rx_clk_sel;
+	unsigned int tx_clk_sel;
+	unsigned int srg_clk_sel;
+	unsigned int rx_fsync_pol;
+	unsigned int tx_fsync_pol;
+	unsigned int rx_fsync_sel;
+	unsigned int tx_fsync_sel;
+	unsigned int rx_fifo_config;
+	unsigned int tx_fifo_config;
+	unsigned int spi_clk_mode;
+	unsigned int spi_burst_mode;
+	unsigned int loopback_enable;
+	unsigned int tx_data_enable;
+	unsigned int default_protdesc;
+	struct msp_protdesc protdesc;
+	int multichannel_configured;
+	struct msp_multichannel_config multichannel_config;
+	unsigned int direction;
+	unsigned int protocol;
+	unsigned int frame_freq;
+	unsigned int frame_size;
+	enum msp_data_size data_size;
+	unsigned int def_elem_len;
+	unsigned int iodelay;
+	void (*handler) (void *data);
+	void *tx_callback_data;
+	void *rx_callback_data;
+};
+
+struct ux500_msp {
+	enum enum_i2s_controller id;
+	void __iomem *registers;
+	struct device *dev;
+	struct i2s_controller *i2s_cont;
+	struct stedma40_chan_cfg *dma_cfg_rx;
+	struct stedma40_chan_cfg *dma_cfg_tx;
+	struct dma_chan *tx_pipeid;
+	struct dma_chan *rx_pipeid;
+	enum msp_state msp_state;
+	int (*transfer) (struct ux500_msp *msp, struct i2s_message *message);
+	int (*plat_init) (void);
+	int (*plat_exit) (void);
+	struct timer_list notify_timer;
+	int def_elem_len;
+	unsigned int dir_busy;
+	int loopback_enable;
+	u32 backup_regs[MAX_MSP_BACKUP_REGS];
+	unsigned int f_bitclk;
+};
+
+struct ux500_msp_dma_params {
+	unsigned int data_size;
+	struct stedma40_chan_cfg *dma_cfg;
+};
+
+int ux500_msp_i2s_init_msp(struct platform_device *pdev,
+			struct ux500_msp **msp_p,
+			struct msp_i2s_platform_data *platform_data);
+void ux500_msp_i2s_cleanup_msp(struct platform_device *pdev,
+			struct ux500_msp *msp);
+int ux500_msp_i2s_open(struct ux500_msp *msp, struct ux500_msp_config *config);
+int ux500_msp_i2s_close(struct ux500_msp *msp,
+			unsigned int dir);
+int ux500_msp_i2s_trigger(struct ux500_msp *msp, int cmd,
+			int direction);
+
+#endif
diff --git a/sound/sound_core.c b/sound/sound_core.c
index c6e81fb928e9..fb9255cca214 100644
--- a/sound/sound_core.c
+++ b/sound/sound_core.c
@@ -361,7 +361,7 @@ int register_sound_special_device(const struct file_operations *fops, int unit,
 				  struct device *dev)
 {
 	const int chain = unit % SOUND_STEP;
-	int max_unit = 128 + chain;
+	int max_unit = 256;
 	const char *name;
 	char _name[16];
 
diff --git a/sound/usb/card.c b/sound/usb/card.c
index 4a7be7b98331..d5b5c3388e28 100644
--- a/sound/usb/card.c
+++ b/sound/usb/card.c
@@ -131,8 +131,9 @@ static void snd_usb_stream_disconnect(struct list_head *head)
 		subs = &as->substream[idx];
 		if (!subs->num_formats)
 			continue;
-		snd_usb_release_substream_urbs(subs, 1);
 		subs->interface = -1;
+		subs->data_endpoint = NULL;
+		subs->sync_endpoint = NULL;
 	}
 }
 
@@ -276,6 +277,7 @@ static int snd_usb_create_streams(struct snd_usb_audio *chip, int ctrlif)
 
 static int snd_usb_audio_free(struct snd_usb_audio *chip)
 {
+	mutex_destroy(&chip->mutex);
 	kfree(chip);
 	return 0;
 }
@@ -336,6 +338,7 @@ static int snd_usb_audio_create(struct usb_device *dev, int idx,
 		return -ENOMEM;
 	}
 
+	mutex_init(&chip->mutex);
 	mutex_init(&chip->shutdown_mutex);
 	chip->index = idx;
 	chip->dev = dev;
@@ -348,6 +351,7 @@ static int snd_usb_audio_create(struct usb_device *dev, int idx,
 	chip->usb_id = USB_ID(le16_to_cpu(dev->descriptor.idVendor),
 			      le16_to_cpu(dev->descriptor.idProduct));
 	INIT_LIST_HEAD(&chip->pcm_list);
+	INIT_LIST_HEAD(&chip->ep_list);
 	INIT_LIST_HEAD(&chip->midi_list);
 	INIT_LIST_HEAD(&chip->mixer_list);
 
@@ -565,6 +569,10 @@ static void snd_usb_audio_disconnect(struct usb_device *dev,
 		list_for_each(p, &chip->pcm_list) {
 			snd_usb_stream_disconnect(p);
 		}
+		/* release the endpoint resources */
+		list_for_each(p, &chip->ep_list) {
+			snd_usb_endpoint_free(p);
+		}
 		/* release the midi resources */
 		list_for_each(p, &chip->midi_list) {
 			snd_usbmidi_disconnect(p);
diff --git a/sound/usb/card.h b/sound/usb/card.h
index da5fa1ac4eda..0d37238b8457 100644
--- a/sound/usb/card.h
+++ b/sound/usb/card.h
@@ -30,20 +30,71 @@ struct audioformat {
 };
 
 struct snd_usb_substream;
+struct snd_usb_endpoint;
 
 struct snd_urb_ctx {
 	struct urb *urb;
 	unsigned int buffer_size;	/* size of data buffer, if data URB */
 	struct snd_usb_substream *subs;
+	struct snd_usb_endpoint *ep;
 	int index;	/* index for urb array */
 	int packets;	/* number of packets per urb */
+	int packet_size[MAX_PACKS_HS]; /* size of packets for next submission */
+	struct list_head ready_list;
 };
 
-struct snd_urb_ops {
-	int (*prepare)(struct snd_usb_substream *subs, struct snd_pcm_runtime *runtime, struct urb *u);
-	int (*retire)(struct snd_usb_substream *subs, struct snd_pcm_runtime *runtime, struct urb *u);
-	int (*prepare_sync)(struct snd_usb_substream *subs, struct snd_pcm_runtime *runtime, struct urb *u);
-	int (*retire_sync)(struct snd_usb_substream *subs, struct snd_pcm_runtime *runtime, struct urb *u);
+struct snd_usb_endpoint {
+	struct snd_usb_audio *chip;
+
+	int use_count;
+	int ep_num;		/* the referenced endpoint number */
+	int type;		/* SND_USB_ENDPOINT_TYPE_* */
+	unsigned long flags;
+
+	void (*prepare_data_urb) (struct snd_usb_substream *subs,
+				  struct urb *urb);
+	void (*retire_data_urb) (struct snd_usb_substream *subs,
+				 struct urb *urb);
+
+	struct snd_usb_substream *data_subs;
+	struct snd_usb_endpoint *sync_master;
+	struct snd_usb_endpoint *sync_slave;
+
+	struct snd_urb_ctx urb[MAX_URBS];
+
+	struct snd_usb_packet_info {
+		uint32_t packet_size[MAX_PACKS_HS];
+		int packets;
+	} next_packet[MAX_URBS];
+	int next_packet_read_pos, next_packet_write_pos;
+	struct list_head ready_playback_urbs;
+
+	unsigned int nurbs;		/* # urbs */
+	unsigned long active_mask;	/* bitmask of active urbs */
+	unsigned long unlink_mask;	/* bitmask of unlinked urbs */
+	char *syncbuf;			/* sync buffer for all sync URBs */
+	dma_addr_t sync_dma;		/* DMA address of syncbuf */
+
+	unsigned int pipe;		/* the data i/o pipe */
+	unsigned int freqn;		/* nominal sampling rate in fs/fps in Q16.16 format */
+	unsigned int freqm;		/* momentary sampling rate in fs/fps in Q16.16 format */
+	int	   freqshift;		/* how much to shift the feedback value to get Q16.16 */
+	unsigned int freqmax;		/* maximum sampling rate, used for buffer management */
+	unsigned int phase;		/* phase accumulator */
+	unsigned int maxpacksize;	/* max packet size in bytes */
+	unsigned int maxframesize;      /* max packet size in frames */
+	unsigned int curpacksize;	/* current packet size in bytes (for capture) */
+	unsigned int curframesize;      /* current packet size in frames (for capture) */
+	unsigned int syncmaxsize;	/* sync endpoint packet size */
+	unsigned int fill_max:1;	/* fill max packet size always */
+	unsigned int datainterval;      /* log_2 of data packet interval */
+	unsigned int syncinterval;	/* P for adaptive mode, 0 otherwise */
+	unsigned char silence_value;
+	unsigned int stride;
+	int iface, alt_idx;
+
+	spinlock_t lock;
+	struct list_head list;
 };
 
 struct snd_usb_substream {
@@ -57,21 +108,6 @@ struct snd_usb_substream {
 	unsigned int cur_rate;		/* current rate (for hw_params callback) */
 	unsigned int period_bytes;	/* current period bytes (for hw_params callback) */
 	unsigned int altset_idx;     /* USB data format: index of alternate setting */
-	unsigned int datapipe;   /* the data i/o pipe */
-	unsigned int syncpipe;   /* 1 - async out or adaptive in */
-	unsigned int datainterval;	/* log_2 of data packet interval */
-	unsigned int syncinterval;  /* P for adaptive mode, 0 otherwise */
-	unsigned int freqn;      /* nominal sampling rate in fs/fps in Q16.16 format */
-	unsigned int freqm;      /* momentary sampling rate in fs/fps in Q16.16 format */
-	int          freqshift;  /* how much to shift the feedback value to get Q16.16 */
-	unsigned int freqmax;    /* maximum sampling rate, used for buffer management */
-	unsigned int phase;      /* phase accumulator */
-	unsigned int maxpacksize;	/* max packet size in bytes */
-	unsigned int maxframesize;	/* max packet size in frames */
-	unsigned int curpacksize;	/* current packet size in bytes (for capture) */
-	unsigned int curframesize;	/* current packet size in frames (for capture) */
-	unsigned int syncmaxsize;	/* sync endpoint packet size */
-	unsigned int fill_max: 1;	/* fill max packet size always */
 	unsigned int txfr_quirk:1;	/* allow sub-frame alignment */
 	unsigned int fmt_type;		/* USB audio format type (1-3) */
 
@@ -82,11 +118,10 @@ struct snd_usb_substream {
 	unsigned long active_mask;	/* bitmask of active urbs */
 	unsigned long unlink_mask;	/* bitmask of unlinked urbs */
 
-	unsigned int nurbs;			/* # urbs */
-	struct snd_urb_ctx dataurb[MAX_URBS];	/* data urb table */
-	struct snd_urb_ctx syncurb[SYNC_URBS];	/* sync urb table */
-	char *syncbuf;				/* sync buffer for all sync URBs */
-	dma_addr_t sync_dma;			/* DMA address of syncbuf */
+	/* data and sync endpoints for this stream */
+	struct snd_usb_endpoint *data_endpoint;
+	struct snd_usb_endpoint *sync_endpoint;
+	unsigned long flags;
 
 	u64 formats;			/* format bitmasks (all or'ed) */
 	unsigned int num_formats;		/* number of supported audio formats (list) */
@@ -94,7 +129,6 @@ struct snd_usb_substream {
 	struct snd_pcm_hw_constraint_list rate_list;	/* limited rates */
 	spinlock_t lock;
 
-	struct snd_urb_ops ops;		/* callbacks (must be filled at init) */
 	int last_frame_number;          /* stored frame number */
 	int last_delay;                 /* stored delay */
 };
diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c
index 08dcce53720b..e6906901debb 100644
--- a/sound/usb/endpoint.c
+++ b/sound/usb/endpoint.c
@@ -20,9 +20,11 @@
 #include <linux/ratelimit.h>
 #include <linux/usb.h>
 #include <linux/usb/audio.h>
+#include <linux/slab.h>
 
 #include <sound/core.h>
 #include <sound/pcm.h>
+#include <sound/pcm_params.h>
 
 #include "usbaudio.h"
 #include "helper.h"
@@ -30,6 +32,36 @@
 #include "endpoint.h"
 #include "pcm.h"
 
+#define EP_FLAG_ACTIVATED	0
+#define EP_FLAG_RUNNING		1
+
+/*
+ * snd_usb_endpoint is a model that abstracts everything related to an
+ * USB endpoint and its streaming.
+ *
+ * There are functions to activate and deactivate the streaming URBs and
+ * optional callbacks to let the pcm logic handle the actual content of the
+ * packets for playback and record. Thus, the bus streaming and the audio
+ * handlers are fully decoupled.
+ *
+ * There are two different types of endpoints in audio applications.
+ *
+ * SND_USB_ENDPOINT_TYPE_DATA handles full audio data payload for both
+ * inbound and outbound traffic.
+ *
+ * SND_USB_ENDPOINT_TYPE_SYNC endpoints are for inbound traffic only and
+ * expect the payload to carry Q10.14 / Q16.16 formatted sync information
+ * (3 or 4 bytes).
+ *
+ * Each endpoint has to be configured prior to being used by calling
+ * snd_usb_endpoint_set_params().
+ *
+ * The model incorporates a reference counting, so that multiple users
+ * can call snd_usb_endpoint_start() and snd_usb_endpoint_stop(), and
+ * only the first user will effectively start the URBs, and only the last
+ * one to stop it will tear the URBs down again.
+ */
+
 /*
  * convert a sampling rate into our full speed format (fs/1000 in Q16.16)
  * this will overflow at approx 524 kHz
@@ -48,728 +80,17 @@ static inline unsigned get_usb_high_speed_rate(unsigned int rate)
 	return ((rate << 10) + 62) / 125;
 }
 
-/*
- * unlink active urbs.
- */
-static int deactivate_urbs(struct snd_usb_substream *subs, int force, int can_sleep)
-{
-	struct snd_usb_audio *chip = subs->stream->chip;
-	unsigned int i;
-	int async;
-
-	subs->running = 0;
-
-	if (!force && subs->stream->chip->shutdown) /* to be sure... */
-		return -EBADFD;
-
-	async = !can_sleep && chip->async_unlink;
-
-	if (!async && in_interrupt())
-		return 0;
-
-	for (i = 0; i < subs->nurbs; i++) {
-		if (test_bit(i, &subs->active_mask)) {
-			if (!test_and_set_bit(i, &subs->unlink_mask)) {
-				struct urb *u = subs->dataurb[i].urb;
-				if (async)
-					usb_unlink_urb(u);
-				else
-					usb_kill_urb(u);
-			}
-		}
-	}
-	if (subs->syncpipe) {
-		for (i = 0; i < SYNC_URBS; i++) {
-			if (test_bit(i+16, &subs->active_mask)) {
-				if (!test_and_set_bit(i+16, &subs->unlink_mask)) {
-					struct urb *u = subs->syncurb[i].urb;
-					if (async)
-						usb_unlink_urb(u);
-					else
-						usb_kill_urb(u);
-				}
-			}
-		}
-	}
-	return 0;
-}
-
-
 /*
  * release a urb data
  */
 static void release_urb_ctx(struct snd_urb_ctx *u)
 {
-	if (u->urb) {
-		if (u->buffer_size)
-			usb_free_coherent(u->subs->dev, u->buffer_size,
-					u->urb->transfer_buffer,
-					u->urb->transfer_dma);
-		usb_free_urb(u->urb);
-		u->urb = NULL;
-	}
-}
-
-/*
- *  wait until all urbs are processed.
- */
-static int wait_clear_urbs(struct snd_usb_substream *subs)
-{
-	unsigned long end_time = jiffies + msecs_to_jiffies(1000);
-	unsigned int i;
-	int alive;
-
-	do {
-		alive = 0;
-		for (i = 0; i < subs->nurbs; i++) {
-			if (test_bit(i, &subs->active_mask))
-				alive++;
-		}
-		if (subs->syncpipe) {
-			for (i = 0; i < SYNC_URBS; i++) {
-				if (test_bit(i + 16, &subs->active_mask))
-					alive++;
-			}
-		}
-		if (! alive)
-			break;
-		schedule_timeout_uninterruptible(1);
-	} while (time_before(jiffies, end_time));
-	if (alive)
-		snd_printk(KERN_ERR "timeout: still %d active urbs..\n", alive);
-	return 0;
-}
-
-/*
- * release a substream
- */
-void snd_usb_release_substream_urbs(struct snd_usb_substream *subs, int force)
-{
-	int i;
-
-	/* stop urbs (to be sure) */
-	deactivate_urbs(subs, force, 1);
-	wait_clear_urbs(subs);
-
-	for (i = 0; i < MAX_URBS; i++)
-		release_urb_ctx(&subs->dataurb[i]);
-	for (i = 0; i < SYNC_URBS; i++)
-		release_urb_ctx(&subs->syncurb[i]);
-	usb_free_coherent(subs->dev, SYNC_URBS * 4,
-			subs->syncbuf, subs->sync_dma);
-	subs->syncbuf = NULL;
-	subs->nurbs = 0;
-}
-
-/*
- * complete callback from data urb
- */
-static void snd_complete_urb(struct urb *urb)
-{
-	struct snd_urb_ctx *ctx = urb->context;
-	struct snd_usb_substream *subs = ctx->subs;
-	struct snd_pcm_substream *substream = ctx->subs->pcm_substream;
-	int err = 0;
-
-	if ((subs->running && subs->ops.retire(subs, substream->runtime, urb)) ||
-	    !subs->running || /* can be stopped during retire callback */
-	    (err = subs->ops.prepare(subs, substream->runtime, urb)) < 0 ||
-	    (err = usb_submit_urb(urb, GFP_ATOMIC)) < 0) {
-		clear_bit(ctx->index, &subs->active_mask);
-		if (err < 0) {
-			snd_printd(KERN_ERR "cannot submit urb (err = %d)\n", err);
-			snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN);
-		}
-	}
-}
-
-
-/*
- * complete callback from sync urb
- */
-static void snd_complete_sync_urb(struct urb *urb)
-{
-	struct snd_urb_ctx *ctx = urb->context;
-	struct snd_usb_substream *subs = ctx->subs;
-	struct snd_pcm_substream *substream = ctx->subs->pcm_substream;
-	int err = 0;
-
-	if ((subs->running && subs->ops.retire_sync(subs, substream->runtime, urb)) ||
-	    !subs->running || /* can be stopped during retire callback */
-	    (err = subs->ops.prepare_sync(subs, substream->runtime, urb)) < 0 ||
-	    (err = usb_submit_urb(urb, GFP_ATOMIC)) < 0) {
-		clear_bit(ctx->index + 16, &subs->active_mask);
-		if (err < 0) {
-			snd_printd(KERN_ERR "cannot submit sync urb (err = %d)\n", err);
-			snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN);
-		}
-	}
-}
-
-
-/*
- * initialize a substream for plaback/capture
- */
-int snd_usb_init_substream_urbs(struct snd_usb_substream *subs,
-				unsigned int period_bytes,
-				unsigned int rate,
-				unsigned int frame_bits)
-{
-	unsigned int maxsize, i;
-	int is_playback = subs->direction == SNDRV_PCM_STREAM_PLAYBACK;
-	unsigned int urb_packs, total_packs, packs_per_ms;
-	struct snd_usb_audio *chip = subs->stream->chip;
-
-	/* calculate the frequency in 16.16 format */
-	if (snd_usb_get_speed(subs->dev) == USB_SPEED_FULL)
-		subs->freqn = get_usb_full_speed_rate(rate);
-	else
-		subs->freqn = get_usb_high_speed_rate(rate);
-	subs->freqm = subs->freqn;
-	subs->freqshift = INT_MIN;
-	/* calculate max. frequency */
-	if (subs->maxpacksize) {
-		/* whatever fits into a max. size packet */
-		maxsize = subs->maxpacksize;
-		subs->freqmax = (maxsize / (frame_bits >> 3))
-				<< (16 - subs->datainterval);
-	} else {
-		/* no max. packet size: just take 25% higher than nominal */
-		subs->freqmax = subs->freqn + (subs->freqn >> 2);
-		maxsize = ((subs->freqmax + 0xffff) * (frame_bits >> 3))
-				>> (16 - subs->datainterval);
-	}
-	subs->phase = 0;
-
-	if (subs->fill_max)
-		subs->curpacksize = subs->maxpacksize;
-	else
-		subs->curpacksize = maxsize;
-
-	if (snd_usb_get_speed(subs->dev) != USB_SPEED_FULL)
-		packs_per_ms = 8 >> subs->datainterval;
-	else
-		packs_per_ms = 1;
-
-	if (is_playback) {
-		urb_packs = max(chip->nrpacks, 1);
-		urb_packs = min(urb_packs, (unsigned int)MAX_PACKS);
-	} else
-		urb_packs = 1;
-	urb_packs *= packs_per_ms;
-	if (subs->syncpipe)
-		urb_packs = min(urb_packs, 1U << subs->syncinterval);
-
-	/* decide how many packets to be used */
-	if (is_playback) {
-		unsigned int minsize, maxpacks;
-		/* determine how small a packet can be */
-		minsize = (subs->freqn >> (16 - subs->datainterval))
-			  * (frame_bits >> 3);
-		/* with sync from device, assume it can be 12% lower */
-		if (subs->syncpipe)
-			minsize -= minsize >> 3;
-		minsize = max(minsize, 1u);
-		total_packs = (period_bytes + minsize - 1) / minsize;
-		/* we need at least two URBs for queueing */
-		if (total_packs < 2) {
-			total_packs = 2;
-		} else {
-			/* and we don't want too long a queue either */
-			maxpacks = max(MAX_QUEUE * packs_per_ms, urb_packs * 2);
-			total_packs = min(total_packs, maxpacks);
-		}
-	} else {
-		while (urb_packs > 1 && urb_packs * maxsize >= period_bytes)
-			urb_packs >>= 1;
-		total_packs = MAX_URBS * urb_packs;
-	}
-	subs->nurbs = (total_packs + urb_packs - 1) / urb_packs;
-	if (subs->nurbs > MAX_URBS) {
-		/* too much... */
-		subs->nurbs = MAX_URBS;
-		total_packs = MAX_URBS * urb_packs;
-	} else if (subs->nurbs < 2) {
-		/* too little - we need at least two packets
-		 * to ensure contiguous playback/capture
-		 */
-		subs->nurbs = 2;
-	}
-
-	/* allocate and initialize data urbs */
-	for (i = 0; i < subs->nurbs; i++) {
-		struct snd_urb_ctx *u = &subs->dataurb[i];
-		u->index = i;
-		u->subs = subs;
-		u->packets = (i + 1) * total_packs / subs->nurbs
-			- i * total_packs / subs->nurbs;
-		u->buffer_size = maxsize * u->packets;
-		if (subs->fmt_type == UAC_FORMAT_TYPE_II)
-			u->packets++; /* for transfer delimiter */
-		u->urb = usb_alloc_urb(u->packets, GFP_KERNEL);
-		if (!u->urb)
-			goto out_of_memory;
-		u->urb->transfer_buffer =
-			usb_alloc_coherent(subs->dev, u->buffer_size,
-					   GFP_KERNEL, &u->urb->transfer_dma);
-		if (!u->urb->transfer_buffer)
-			goto out_of_memory;
-		u->urb->pipe = subs->datapipe;
-		u->urb->transfer_flags = URB_ISO_ASAP | URB_NO_TRANSFER_DMA_MAP;
-		u->urb->interval = 1 << subs->datainterval;
-		u->urb->context = u;
-		u->urb->complete = snd_complete_urb;
-	}
-
-	if (subs->syncpipe) {
-		/* allocate and initialize sync urbs */
-		subs->syncbuf = usb_alloc_coherent(subs->dev, SYNC_URBS * 4,
-						 GFP_KERNEL, &subs->sync_dma);
-		if (!subs->syncbuf)
-			goto out_of_memory;
-		for (i = 0; i < SYNC_URBS; i++) {
-			struct snd_urb_ctx *u = &subs->syncurb[i];
-			u->index = i;
-			u->subs = subs;
-			u->packets = 1;
-			u->urb = usb_alloc_urb(1, GFP_KERNEL);
-			if (!u->urb)
-				goto out_of_memory;
-			u->urb->transfer_buffer = subs->syncbuf + i * 4;
-			u->urb->transfer_dma = subs->sync_dma + i * 4;
-			u->urb->transfer_buffer_length = 4;
-			u->urb->pipe = subs->syncpipe;
-			u->urb->transfer_flags = URB_ISO_ASAP |
-						 URB_NO_TRANSFER_DMA_MAP;
-			u->urb->number_of_packets = 1;
-			u->urb->interval = 1 << subs->syncinterval;
-			u->urb->context = u;
-			u->urb->complete = snd_complete_sync_urb;
-		}
-	}
-	return 0;
-
-out_of_memory:
-	snd_usb_release_substream_urbs(subs, 0);
-	return -ENOMEM;
-}
-
-/*
- * prepare urb for full speed capture sync pipe
- *
- * fill the length and offset of each urb descriptor.
- * the fixed 10.14 frequency is passed through the pipe.
- */
-static int prepare_capture_sync_urb(struct snd_usb_substream *subs,
-				    struct snd_pcm_runtime *runtime,
-				    struct urb *urb)
-{
-	unsigned char *cp = urb->transfer_buffer;
-	struct snd_urb_ctx *ctx = urb->context;
-
-	urb->dev = ctx->subs->dev; /* we need to set this at each time */
-	urb->iso_frame_desc[0].length = 3;
-	urb->iso_frame_desc[0].offset = 0;
-	cp[0] = subs->freqn >> 2;
-	cp[1] = subs->freqn >> 10;
-	cp[2] = subs->freqn >> 18;
-	return 0;
-}
-
-/*
- * prepare urb for high speed capture sync pipe
- *
- * fill the length and offset of each urb descriptor.
- * the fixed 12.13 frequency is passed as 16.16 through the pipe.
- */
-static int prepare_capture_sync_urb_hs(struct snd_usb_substream *subs,
-				       struct snd_pcm_runtime *runtime,
-				       struct urb *urb)
-{
-	unsigned char *cp = urb->transfer_buffer;
-	struct snd_urb_ctx *ctx = urb->context;
-
-	urb->dev = ctx->subs->dev; /* we need to set this at each time */
-	urb->iso_frame_desc[0].length = 4;
-	urb->iso_frame_desc[0].offset = 0;
-	cp[0] = subs->freqn;
-	cp[1] = subs->freqn >> 8;
-	cp[2] = subs->freqn >> 16;
-	cp[3] = subs->freqn >> 24;
-	return 0;
-}
-
-/*
- * process after capture sync complete
- * - nothing to do
- */
-static int retire_capture_sync_urb(struct snd_usb_substream *subs,
-				   struct snd_pcm_runtime *runtime,
-				   struct urb *urb)
-{
-	return 0;
-}
-
-/*
- * prepare urb for capture data pipe
- *
- * fill the offset and length of each descriptor.
- *
- * we use a temporary buffer to write the captured data.
- * since the length of written data is determined by host, we cannot
- * write onto the pcm buffer directly...  the data is thus copied
- * later at complete callback to the global buffer.
- */
-static int prepare_capture_urb(struct snd_usb_substream *subs,
-			       struct snd_pcm_runtime *runtime,
-			       struct urb *urb)
-{
-	int i, offs;
-	struct snd_urb_ctx *ctx = urb->context;
-
-	offs = 0;
-	urb->dev = ctx->subs->dev; /* we need to set this at each time */
-	for (i = 0; i < ctx->packets; i++) {
-		urb->iso_frame_desc[i].offset = offs;
-		urb->iso_frame_desc[i].length = subs->curpacksize;
-		offs += subs->curpacksize;
-	}
-	urb->transfer_buffer_length = offs;
-	urb->number_of_packets = ctx->packets;
-	return 0;
-}
-
-/*
- * process after capture complete
- *
- * copy the data from each desctiptor to the pcm buffer, and
- * update the current position.
- */
-static int retire_capture_urb(struct snd_usb_substream *subs,
-			      struct snd_pcm_runtime *runtime,
-			      struct urb *urb)
-{
-	unsigned long flags;
-	unsigned char *cp;
-	int i;
-	unsigned int stride, frames, bytes, oldptr;
-	int period_elapsed = 0;
-
-	stride = runtime->frame_bits >> 3;
-
-	for (i = 0; i < urb->number_of_packets; i++) {
-		cp = (unsigned char *)urb->transfer_buffer + urb->iso_frame_desc[i].offset;
-		if (urb->iso_frame_desc[i].status && printk_ratelimit()) {
-			snd_printdd("frame %d active: %d\n", i, urb->iso_frame_desc[i].status);
-			// continue;
-		}
-		bytes = urb->iso_frame_desc[i].actual_length;
-		frames = bytes / stride;
-		if (!subs->txfr_quirk)
-			bytes = frames * stride;
-		if (bytes % (runtime->sample_bits >> 3) != 0) {
-#ifdef CONFIG_SND_DEBUG_VERBOSE
-			int oldbytes = bytes;
-#endif
-			bytes = frames * stride;
-			snd_printdd(KERN_ERR "Corrected urb data len. %d->%d\n",
-							oldbytes, bytes);
-		}
-		/* update the current pointer */
-		spin_lock_irqsave(&subs->lock, flags);
-		oldptr = subs->hwptr_done;
-		subs->hwptr_done += bytes;
-		if (subs->hwptr_done >= runtime->buffer_size * stride)
-			subs->hwptr_done -= runtime->buffer_size * stride;
-		frames = (bytes + (oldptr % stride)) / stride;
-		subs->transfer_done += frames;
-		if (subs->transfer_done >= runtime->period_size) {
-			subs->transfer_done -= runtime->period_size;
-			period_elapsed = 1;
-		}
-		spin_unlock_irqrestore(&subs->lock, flags);
-		/* copy a data chunk */
-		if (oldptr + bytes > runtime->buffer_size * stride) {
-			unsigned int bytes1 =
-					runtime->buffer_size * stride - oldptr;
-			memcpy(runtime->dma_area + oldptr, cp, bytes1);
-			memcpy(runtime->dma_area, cp + bytes1, bytes - bytes1);
-		} else {
-			memcpy(runtime->dma_area + oldptr, cp, bytes);
-		}
-	}
-	if (period_elapsed)
-		snd_pcm_period_elapsed(subs->pcm_substream);
-	return 0;
-}
-
-/*
- * Process after capture complete when paused.  Nothing to do.
- */
-static int retire_paused_capture_urb(struct snd_usb_substream *subs,
-				     struct snd_pcm_runtime *runtime,
-				     struct urb *urb)
-{
-	return 0;
-}
-
-
-/*
- * prepare urb for playback sync pipe
- *
- * set up the offset and length to receive the current frequency.
- */
-static int prepare_playback_sync_urb(struct snd_usb_substream *subs,
-				     struct snd_pcm_runtime *runtime,
-				     struct urb *urb)
-{
-	struct snd_urb_ctx *ctx = urb->context;
-
-	urb->dev = ctx->subs->dev; /* we need to set this at each time */
-	urb->iso_frame_desc[0].length = min(4u, ctx->subs->syncmaxsize);
-	urb->iso_frame_desc[0].offset = 0;
-	return 0;
-}
-
-/*
- * process after playback sync complete
- *
- * Full speed devices report feedback values in 10.14 format as samples per
- * frame, high speed devices in 16.16 format as samples per microframe.
- * Because the Audio Class 1 spec was written before USB 2.0, many high speed
- * devices use a wrong interpretation, some others use an entirely different
- * format.  Therefore, we cannot predict what format any particular device uses
- * and must detect it automatically.
- */
-static int retire_playback_sync_urb(struct snd_usb_substream *subs,
-				    struct snd_pcm_runtime *runtime,
-				    struct urb *urb)
-{
-	unsigned int f;
-	int shift;
-	unsigned long flags;
-
-	if (urb->iso_frame_desc[0].status != 0 ||
-	    urb->iso_frame_desc[0].actual_length < 3)
-		return 0;
-
-	f = le32_to_cpup(urb->transfer_buffer);
-	if (urb->iso_frame_desc[0].actual_length == 3)
-		f &= 0x00ffffff;
-	else
-		f &= 0x0fffffff;
-	if (f == 0)
-		return 0;
-
-	if (unlikely(subs->freqshift == INT_MIN)) {
-		/*
-		 * The first time we see a feedback value, determine its format
-		 * by shifting it left or right until it matches the nominal
-		 * frequency value.  This assumes that the feedback does not
-		 * differ from the nominal value more than +50% or -25%.
-		 */
-		shift = 0;
-		while (f < subs->freqn - subs->freqn / 4) {
-			f <<= 1;
-			shift++;
-		}
-		while (f > subs->freqn + subs->freqn / 2) {
-			f >>= 1;
-			shift--;
-		}
-		subs->freqshift = shift;
-	}
-	else if (subs->freqshift >= 0)
-		f <<= subs->freqshift;
-	else
-		f >>= -subs->freqshift;
-
-	if (likely(f >= subs->freqn - subs->freqn / 8 && f <= subs->freqmax)) {
-		/*
-		 * If the frequency looks valid, set it.
-		 * This value is referred to in prepare_playback_urb().
-		 */
-		spin_lock_irqsave(&subs->lock, flags);
-		subs->freqm = f;
-		spin_unlock_irqrestore(&subs->lock, flags);
-	} else {
-		/*
-		 * Out of range; maybe the shift value is wrong.
-		 * Reset it so that we autodetect again the next time.
-		 */
-		subs->freqshift = INT_MIN;
-	}
-
-	return 0;
-}
-
-/* determine the number of frames in the next packet */
-static int snd_usb_audio_next_packet_size(struct snd_usb_substream *subs)
-{
-	if (subs->fill_max)
-		return subs->maxframesize;
-	else {
-		subs->phase = (subs->phase & 0xffff)
-			+ (subs->freqm << subs->datainterval);
-		return min(subs->phase >> 16, subs->maxframesize);
-	}
-}
-
-/*
- * Prepare urb for streaming before playback starts or when paused.
- *
- * We don't have any data, so we send silence.
- */
-static int prepare_nodata_playback_urb(struct snd_usb_substream *subs,
-				       struct snd_pcm_runtime *runtime,
-				       struct urb *urb)
-{
-	unsigned int i, offs, counts;
-	struct snd_urb_ctx *ctx = urb->context;
-	int stride = runtime->frame_bits >> 3;
-
-	offs = 0;
-	urb->dev = ctx->subs->dev;
-	for (i = 0; i < ctx->packets; ++i) {
-		counts = snd_usb_audio_next_packet_size(subs);
-		urb->iso_frame_desc[i].offset = offs * stride;
-		urb->iso_frame_desc[i].length = counts * stride;
-		offs += counts;
-	}
-	urb->number_of_packets = ctx->packets;
-	urb->transfer_buffer_length = offs * stride;
-	memset(urb->transfer_buffer,
-	       runtime->format == SNDRV_PCM_FORMAT_U8 ? 0x80 : 0,
-	       offs * stride);
-	return 0;
-}
-
-/*
- * prepare urb for playback data pipe
- *
- * Since a URB can handle only a single linear buffer, we must use double
- * buffering when the data to be transferred overflows the buffer boundary.
- * To avoid inconsistencies when updating hwptr_done, we use double buffering
- * for all URBs.
- */
-static int prepare_playback_urb(struct snd_usb_substream *subs,
-				struct snd_pcm_runtime *runtime,
-				struct urb *urb)
-{
-	int i, stride;
-	unsigned int counts, frames, bytes;
-	unsigned long flags;
-	int period_elapsed = 0;
-	struct snd_urb_ctx *ctx = urb->context;
-
-	stride = runtime->frame_bits >> 3;
-
-	frames = 0;
-	urb->dev = ctx->subs->dev; /* we need to set this at each time */
-	urb->number_of_packets = 0;
-	spin_lock_irqsave(&subs->lock, flags);
-	for (i = 0; i < ctx->packets; i++) {
-		counts = snd_usb_audio_next_packet_size(subs);
-		/* set up descriptor */
-		urb->iso_frame_desc[i].offset = frames * stride;
-		urb->iso_frame_desc[i].length = counts * stride;
-		frames += counts;
-		urb->number_of_packets++;
-		subs->transfer_done += counts;
-		if (subs->transfer_done >= runtime->period_size) {
-			subs->transfer_done -= runtime->period_size;
-			period_elapsed = 1;
-			if (subs->fmt_type == UAC_FORMAT_TYPE_II) {
-				if (subs->transfer_done > 0) {
-					/* FIXME: fill-max mode is not
-					 * supported yet */
-					frames -= subs->transfer_done;
-					counts -= subs->transfer_done;
-					urb->iso_frame_desc[i].length =
-						counts * stride;
-					subs->transfer_done = 0;
-				}
-				i++;
-				if (i < ctx->packets) {
-					/* add a transfer delimiter */
-					urb->iso_frame_desc[i].offset =
-						frames * stride;
-					urb->iso_frame_desc[i].length = 0;
-					urb->number_of_packets++;
-				}
-				break;
-			}
-		}
-		if (period_elapsed) /* finish at the period boundary */
-			break;
-	}
-	bytes = frames * stride;
-	if (subs->hwptr_done + bytes > runtime->buffer_size * stride) {
-		/* err, the transferred area goes over buffer boundary. */
-		unsigned int bytes1 =
-			runtime->buffer_size * stride - subs->hwptr_done;
-		memcpy(urb->transfer_buffer,
-		       runtime->dma_area + subs->hwptr_done, bytes1);
-		memcpy(urb->transfer_buffer + bytes1,
-		       runtime->dma_area, bytes - bytes1);
-	} else {
-		memcpy(urb->transfer_buffer,
-		       runtime->dma_area + subs->hwptr_done, bytes);
-	}
-	subs->hwptr_done += bytes;
-	if (subs->hwptr_done >= runtime->buffer_size * stride)
-		subs->hwptr_done -= runtime->buffer_size * stride;
-
-	/* update delay with exact number of samples queued */
-	runtime->delay = subs->last_delay;
-	runtime->delay += frames;
-	subs->last_delay = runtime->delay;
-
-	/* realign last_frame_number */
-	subs->last_frame_number = usb_get_current_frame_number(subs->dev);
-	subs->last_frame_number &= 0xFF; /* keep 8 LSBs */
-
-	spin_unlock_irqrestore(&subs->lock, flags);
-	urb->transfer_buffer_length = bytes;
-	if (period_elapsed)
-		snd_pcm_period_elapsed(subs->pcm_substream);
-	return 0;
-}
-
-/*
- * process after playback data complete
- * - decrease the delay count again
- */
-static int retire_playback_urb(struct snd_usb_substream *subs,
-			       struct snd_pcm_runtime *runtime,
-			       struct urb *urb)
-{
-	unsigned long flags;
-	int stride = runtime->frame_bits >> 3;
-	int processed = urb->transfer_buffer_length / stride;
-	int est_delay;
-
-	spin_lock_irqsave(&subs->lock, flags);
-
-	est_delay = snd_usb_pcm_delay(subs, runtime->rate);
-	/* update delay with exact number of samples played */
-	if (processed > subs->last_delay)
-		subs->last_delay = 0;
-	else
-		subs->last_delay -= processed;
-	runtime->delay = subs->last_delay;
-
-	/*
-	 * Report when delay estimate is off by more than 2ms.
-	 * The error should be lower than 2ms since the estimate relies
-	 * on two reads of a counter updated every ms.
-	 */
-	if (abs(est_delay - subs->last_delay) * 1000 > runtime->rate * 2)
-		snd_printk(KERN_DEBUG "delay: estimated %d, actual %d\n",
-			est_delay, subs->last_delay);
-
-	spin_unlock_irqrestore(&subs->lock, flags);
-	return 0;
+	if (u->buffer_size)
+		usb_free_coherent(u->ep->chip->dev, u->buffer_size,
+				  u->urb->transfer_buffer,
+				  u->urb->transfer_dma);
+	usb_free_urb(u->urb);
+	u->urb = NULL;
 }
 
 static const char *usb_error_string(int err)
@@ -797,170 +118,1044 @@ static const char *usb_error_string(int err)
 	}
 }
 
-/*
- * set up and start data/sync urbs
+/**
+ * snd_usb_endpoint_implicit_feedback_sink: Report endpoint usage type
+ *
+ * @ep: The snd_usb_endpoint
+ *
+ * Determine whether an endpoint is driven by an implicit feedback
+ * data endpoint source.
  */
-static int start_urbs(struct snd_usb_substream *subs, struct snd_pcm_runtime *runtime)
+int snd_usb_endpoint_implict_feedback_sink(struct snd_usb_endpoint *ep)
 {
-	unsigned int i;
+	return  ep->sync_master &&
+		ep->sync_master->type == SND_USB_ENDPOINT_TYPE_DATA &&
+		ep->type == SND_USB_ENDPOINT_TYPE_DATA &&
+		usb_pipeout(ep->pipe);
+}
+
+/*
+ * For streaming based on information derived from sync endpoints,
+ * prepare_outbound_urb_sizes() will call next_packet_size() to
+ * determine the number of samples to be sent in the next packet.
+ *
+ * For implicit feedback, next_packet_size() is unused.
+ */
+static int next_packet_size(struct snd_usb_endpoint *ep)
+{
+	unsigned long flags;
+	int ret;
+
+	if (ep->fill_max)
+		return ep->maxframesize;
+
+	spin_lock_irqsave(&ep->lock, flags);
+	ep->phase = (ep->phase & 0xffff)
+		+ (ep->freqm << ep->datainterval);
+	ret = min(ep->phase >> 16, ep->maxframesize);
+	spin_unlock_irqrestore(&ep->lock, flags);
+
+	return ret;
+}
+
+static void retire_outbound_urb(struct snd_usb_endpoint *ep,
+				struct snd_urb_ctx *urb_ctx)
+{
+	if (ep->retire_data_urb)
+		ep->retire_data_urb(ep->data_subs, urb_ctx->urb);
+}
+
+static void retire_inbound_urb(struct snd_usb_endpoint *ep,
+			       struct snd_urb_ctx *urb_ctx)
+{
+	struct urb *urb = urb_ctx->urb;
+
+	if (ep->sync_slave)
+		snd_usb_handle_sync_urb(ep->sync_slave, ep, urb);
+
+	if (ep->retire_data_urb)
+		ep->retire_data_urb(ep->data_subs, urb);
+}
+
+static void prepare_outbound_urb_sizes(struct snd_usb_endpoint *ep,
+				       struct snd_urb_ctx *ctx)
+{
+	int i;
+
+	for (i = 0; i < ctx->packets; ++i)
+		ctx->packet_size[i] = next_packet_size(ep);
+}
+
+/*
+ * Prepare a PLAYBACK urb for submission to the bus.
+ */
+static void prepare_outbound_urb(struct snd_usb_endpoint *ep,
+				 struct snd_urb_ctx *ctx)
+{
+	int i;
+	struct urb *urb = ctx->urb;
+	unsigned char *cp = urb->transfer_buffer;
+
+	urb->dev = ep->chip->dev; /* we need to set this at each time */
+
+	switch (ep->type) {
+	case SND_USB_ENDPOINT_TYPE_DATA:
+		if (ep->prepare_data_urb) {
+			ep->prepare_data_urb(ep->data_subs, urb);
+		} else {
+			/* no data provider, so send silence */
+			unsigned int offs = 0;
+			for (i = 0; i < ctx->packets; ++i) {
+				int counts = ctx->packet_size[i];
+				urb->iso_frame_desc[i].offset = offs * ep->stride;
+				urb->iso_frame_desc[i].length = counts * ep->stride;
+				offs += counts;
+			}
+
+			urb->number_of_packets = ctx->packets;
+			urb->transfer_buffer_length = offs * ep->stride;
+			memset(urb->transfer_buffer, ep->silence_value,
+			       offs * ep->stride);
+		}
+		break;
+
+	case SND_USB_ENDPOINT_TYPE_SYNC:
+		if (snd_usb_get_speed(ep->chip->dev) >= USB_SPEED_HIGH) {
+			/*
+			 * fill the length and offset of each urb descriptor.
+			 * the fixed 12.13 frequency is passed as 16.16 through the pipe.
+			 */
+			urb->iso_frame_desc[0].length = 4;
+			urb->iso_frame_desc[0].offset = 0;
+			cp[0] = ep->freqn;
+			cp[1] = ep->freqn >> 8;
+			cp[2] = ep->freqn >> 16;
+			cp[3] = ep->freqn >> 24;
+		} else {
+			/*
+			 * fill the length and offset of each urb descriptor.
+			 * the fixed 10.14 frequency is passed through the pipe.
+			 */
+			urb->iso_frame_desc[0].length = 3;
+			urb->iso_frame_desc[0].offset = 0;
+			cp[0] = ep->freqn >> 2;
+			cp[1] = ep->freqn >> 10;
+			cp[2] = ep->freqn >> 18;
+		}
+
+		break;
+	}
+}
+
+/*
+ * Prepare a CAPTURE or SYNC urb for submission to the bus.
+ */
+static inline void prepare_inbound_urb(struct snd_usb_endpoint *ep,
+				       struct snd_urb_ctx *urb_ctx)
+{
+	int i, offs;
+	struct urb *urb = urb_ctx->urb;
+
+	urb->dev = ep->chip->dev; /* we need to set this at each time */
+
+	switch (ep->type) {
+	case SND_USB_ENDPOINT_TYPE_DATA:
+		offs = 0;
+		for (i = 0; i < urb_ctx->packets; i++) {
+			urb->iso_frame_desc[i].offset = offs;
+			urb->iso_frame_desc[i].length = ep->curpacksize;
+			offs += ep->curpacksize;
+		}
+
+		urb->transfer_buffer_length = offs;
+		urb->number_of_packets = urb_ctx->packets;
+		break;
+
+	case SND_USB_ENDPOINT_TYPE_SYNC:
+		urb->iso_frame_desc[0].length = min(4u, ep->syncmaxsize);
+		urb->iso_frame_desc[0].offset = 0;
+		break;
+	}
+}
+
+/*
+ * Send output urbs that have been prepared previously. URBs are dequeued
+ * from ep->ready_playback_urbs and in case there there aren't any available
+ * or there are no packets that have been prepared, this function does
+ * nothing.
+ *
+ * The reason why the functionality of sending and preparing URBs is separated
+ * is that host controllers don't guarantee the order in which they return
+ * inbound and outbound packets to their submitters.
+ *
+ * This function is only used for implicit feedback endpoints. For endpoints
+ * driven by dedicated sync endpoints, URBs are immediately re-submitted
+ * from their completion handler.
+ */
+static void queue_pending_output_urbs(struct snd_usb_endpoint *ep)
+{
+	while (test_bit(EP_FLAG_RUNNING, &ep->flags)) {
+
+		unsigned long flags;
+		struct snd_usb_packet_info *uninitialized_var(packet);
+		struct snd_urb_ctx *ctx = NULL;
+		struct urb *urb;
+		int err, i;
+
+		spin_lock_irqsave(&ep->lock, flags);
+		if (ep->next_packet_read_pos != ep->next_packet_write_pos) {
+			packet = ep->next_packet + ep->next_packet_read_pos;
+			ep->next_packet_read_pos++;
+			ep->next_packet_read_pos %= MAX_URBS;
+
+			/* take URB out of FIFO */
+			if (!list_empty(&ep->ready_playback_urbs))
+				ctx = list_first_entry(&ep->ready_playback_urbs,
+					       struct snd_urb_ctx, ready_list);
+		}
+		spin_unlock_irqrestore(&ep->lock, flags);
+
+		if (ctx == NULL)
+			return;
+
+		list_del_init(&ctx->ready_list);
+		urb = ctx->urb;
+
+		/* copy over the length information */
+		for (i = 0; i < packet->packets; i++)
+			ctx->packet_size[i] = packet->packet_size[i];
+
+		/* call the data handler to fill in playback data */
+		prepare_outbound_urb(ep, ctx);
+
+		err = usb_submit_urb(ctx->urb, GFP_ATOMIC);
+		if (err < 0)
+			snd_printk(KERN_ERR "Unable to submit urb #%d: %d (urb %p)\n",
+				   ctx->index, err, ctx->urb);
+		else
+			set_bit(ctx->index, &ep->active_mask);
+	}
+}
+
+/*
+ * complete callback for urbs
+ */
+static void snd_complete_urb(struct urb *urb)
+{
+	struct snd_urb_ctx *ctx = urb->context;
+	struct snd_usb_endpoint *ep = ctx->ep;
 	int err;
 
-	if (subs->stream->chip->shutdown)
-		return -EBADFD;
+	if (unlikely(urb->status == -ENOENT ||		/* unlinked */
+		     urb->status == -ENODEV ||		/* device removed */
+		     urb->status == -ECONNRESET ||	/* unlinked */
+		     urb->status == -ESHUTDOWN ||	/* device disabled */
+		     ep->chip->shutdown))		/* device disconnected */
+		goto exit_clear;
 
-	for (i = 0; i < subs->nurbs; i++) {
-		if (snd_BUG_ON(!subs->dataurb[i].urb))
-			return -EINVAL;
-		if (subs->ops.prepare(subs, runtime, subs->dataurb[i].urb) < 0) {
-			snd_printk(KERN_ERR "cannot prepare datapipe for urb %d\n", i);
-			goto __error;
+	if (usb_pipeout(ep->pipe)) {
+		retire_outbound_urb(ep, ctx);
+		/* can be stopped during retire callback */
+		if (unlikely(!test_bit(EP_FLAG_RUNNING, &ep->flags)))
+			goto exit_clear;
+
+		if (snd_usb_endpoint_implict_feedback_sink(ep)) {
+			unsigned long flags;
+
+			spin_lock_irqsave(&ep->lock, flags);
+			list_add_tail(&ctx->ready_list, &ep->ready_playback_urbs);
+			spin_unlock_irqrestore(&ep->lock, flags);
+			queue_pending_output_urbs(ep);
+
+			goto exit_clear;
+		}
+
+		prepare_outbound_urb_sizes(ep, ctx);
+		prepare_outbound_urb(ep, ctx);
+	} else {
+		retire_inbound_urb(ep, ctx);
+		/* can be stopped during retire callback */
+		if (unlikely(!test_bit(EP_FLAG_RUNNING, &ep->flags)))
+			goto exit_clear;
+
+		prepare_inbound_urb(ep, ctx);
+	}
+
+	err = usb_submit_urb(urb, GFP_ATOMIC);
+	if (err == 0)
+		return;
+
+	snd_printk(KERN_ERR "cannot submit urb (err = %d)\n", err);
+	//snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN);
+
+exit_clear:
+	clear_bit(ctx->index, &ep->active_mask);
+}
+
+/**
+ * snd_usb_add_endpoint: Add an endpoint to an USB audio chip
+ *
+ * @chip: The chip
+ * @alts: The USB host interface
+ * @ep_num: The number of the endpoint to use
+ * @direction: SNDRV_PCM_STREAM_PLAYBACK or SNDRV_PCM_STREAM_CAPTURE
+ * @type: SND_USB_ENDPOINT_TYPE_DATA or SND_USB_ENDPOINT_TYPE_SYNC
+ *
+ * If the requested endpoint has not been added to the given chip before,
+ * a new instance is created. Otherwise, a pointer to the previoulsy
+ * created instance is returned. In case of any error, NULL is returned.
+ *
+ * New endpoints will be added to chip->ep_list and must be freed by
+ * calling snd_usb_endpoint_free().
+ */
+struct snd_usb_endpoint *snd_usb_add_endpoint(struct snd_usb_audio *chip,
+					      struct usb_host_interface *alts,
+					      int ep_num, int direction, int type)
+{
+	struct list_head *p;
+	struct snd_usb_endpoint *ep;
+	int ret, is_playback = direction == SNDRV_PCM_STREAM_PLAYBACK;
+
+	mutex_lock(&chip->mutex);
+
+	list_for_each(p, &chip->ep_list) {
+		ep = list_entry(p, struct snd_usb_endpoint, list);
+		if (ep->ep_num == ep_num &&
+		    ep->iface == alts->desc.bInterfaceNumber &&
+		    ep->alt_idx == alts->desc.bAlternateSetting) {
+			snd_printdd(KERN_DEBUG "Re-using EP %x in iface %d,%d @%p\n",
+					ep_num, ep->iface, ep->alt_idx, ep);
+			goto __exit_unlock;
 		}
 	}
-	if (subs->syncpipe) {
-		for (i = 0; i < SYNC_URBS; i++) {
-			if (snd_BUG_ON(!subs->syncurb[i].urb))
-				return -EINVAL;
-			if (subs->ops.prepare_sync(subs, runtime, subs->syncurb[i].urb) < 0) {
-				snd_printk(KERN_ERR "cannot prepare syncpipe for urb %d\n", i);
-				goto __error;
+
+	snd_printdd(KERN_DEBUG "Creating new %s %s endpoint #%x\n",
+		    is_playback ? "playback" : "capture",
+		    type == SND_USB_ENDPOINT_TYPE_DATA ? "data" : "sync",
+		    ep_num);
+
+	/* select the alt setting once so the endpoints become valid */
+	ret = usb_set_interface(chip->dev, alts->desc.bInterfaceNumber,
+				alts->desc.bAlternateSetting);
+	if (ret < 0) {
+		snd_printk(KERN_ERR "%s(): usb_set_interface() failed, ret = %d\n",
+					__func__, ret);
+		ep = NULL;
+		goto __exit_unlock;
+	}
+
+	ep = kzalloc(sizeof(*ep), GFP_KERNEL);
+	if (!ep)
+		goto __exit_unlock;
+
+	ep->chip = chip;
+	spin_lock_init(&ep->lock);
+	ep->type = type;
+	ep->ep_num = ep_num;
+	ep->iface = alts->desc.bInterfaceNumber;
+	ep->alt_idx = alts->desc.bAlternateSetting;
+	INIT_LIST_HEAD(&ep->ready_playback_urbs);
+	ep_num &= USB_ENDPOINT_NUMBER_MASK;
+
+	if (is_playback)
+		ep->pipe = usb_sndisocpipe(chip->dev, ep_num);
+	else
+		ep->pipe = usb_rcvisocpipe(chip->dev, ep_num);
+
+	if (type == SND_USB_ENDPOINT_TYPE_SYNC) {
+		if (get_endpoint(alts, 1)->bLength >= USB_DT_ENDPOINT_AUDIO_SIZE &&
+		    get_endpoint(alts, 1)->bRefresh >= 1 &&
+		    get_endpoint(alts, 1)->bRefresh <= 9)
+			ep->syncinterval = get_endpoint(alts, 1)->bRefresh;
+		else if (snd_usb_get_speed(chip->dev) == USB_SPEED_FULL)
+			ep->syncinterval = 1;
+		else if (get_endpoint(alts, 1)->bInterval >= 1 &&
+			 get_endpoint(alts, 1)->bInterval <= 16)
+			ep->syncinterval = get_endpoint(alts, 1)->bInterval - 1;
+		else
+			ep->syncinterval = 3;
+
+		ep->syncmaxsize = le16_to_cpu(get_endpoint(alts, 1)->wMaxPacketSize);
+	}
+
+	list_add_tail(&ep->list, &chip->ep_list);
+
+__exit_unlock:
+	mutex_unlock(&chip->mutex);
+
+	return ep;
+}
+
+/*
+ *  wait until all urbs are processed.
+ */
+static int wait_clear_urbs(struct snd_usb_endpoint *ep)
+{
+	unsigned long end_time = jiffies + msecs_to_jiffies(1000);
+	unsigned int i;
+	int alive;
+
+	do {
+		alive = 0;
+		for (i = 0; i < ep->nurbs; i++)
+			if (test_bit(i, &ep->active_mask))
+				alive++;
+
+		if (!alive)
+			break;
+
+		schedule_timeout_uninterruptible(1);
+	} while (time_before(jiffies, end_time));
+
+	if (alive)
+		snd_printk(KERN_ERR "timeout: still %d active urbs on EP #%x\n",
+					alive, ep->ep_num);
+
+	return 0;
+}
+
+/*
+ * unlink active urbs.
+ */
+static int deactivate_urbs(struct snd_usb_endpoint *ep, int force, int can_sleep)
+{
+	unsigned int i;
+	int async;
+
+	if (!force && ep->chip->shutdown) /* to be sure... */
+		return -EBADFD;
+
+	async = !can_sleep && ep->chip->async_unlink;
+
+	clear_bit(EP_FLAG_RUNNING, &ep->flags);
+
+	INIT_LIST_HEAD(&ep->ready_playback_urbs);
+	ep->next_packet_read_pos = 0;
+	ep->next_packet_write_pos = 0;
+
+	if (!async && in_interrupt())
+		return 0;
+
+	for (i = 0; i < ep->nurbs; i++) {
+		if (test_bit(i, &ep->active_mask)) {
+			if (!test_and_set_bit(i, &ep->unlink_mask)) {
+				struct urb *u = ep->urb[i].urb;
+				if (async)
+					usb_unlink_urb(u);
+				else
+					usb_kill_urb(u);
 			}
 		}
 	}
 
-	subs->active_mask = 0;
-	subs->unlink_mask = 0;
-	subs->running = 1;
-	for (i = 0; i < subs->nurbs; i++) {
-		err = usb_submit_urb(subs->dataurb[i].urb, GFP_ATOMIC);
+	return 0;
+}
+
+/*
+ * release an endpoint's urbs
+ */
+static void release_urbs(struct snd_usb_endpoint *ep, int force)
+{
+	int i;
+
+	/* route incoming urbs to nirvana */
+	ep->retire_data_urb = NULL;
+	ep->prepare_data_urb = NULL;
+
+	/* stop urbs */
+	deactivate_urbs(ep, force, 1);
+	wait_clear_urbs(ep);
+
+	for (i = 0; i < ep->nurbs; i++)
+		release_urb_ctx(&ep->urb[i]);
+
+	if (ep->syncbuf)
+		usb_free_coherent(ep->chip->dev, SYNC_URBS * 4,
+				  ep->syncbuf, ep->sync_dma);
+
+	ep->syncbuf = NULL;
+	ep->nurbs = 0;
+}
+
+/*
+ * configure a data endpoint
+ */
+static int data_ep_set_params(struct snd_usb_endpoint *ep,
+			      struct snd_pcm_hw_params *hw_params,
+			      struct audioformat *fmt,
+			      struct snd_usb_endpoint *sync_ep)
+{
+	unsigned int maxsize, i, urb_packs, total_packs, packs_per_ms;
+	int period_bytes = params_period_bytes(hw_params);
+	int format = params_format(hw_params);
+	int is_playback = usb_pipeout(ep->pipe);
+	int frame_bits = snd_pcm_format_physical_width(params_format(hw_params)) *
+							params_channels(hw_params);
+
+	ep->datainterval = fmt->datainterval;
+	ep->stride = frame_bits >> 3;
+	ep->silence_value = format == SNDRV_PCM_FORMAT_U8 ? 0x80 : 0;
+
+	/* calculate max. frequency */
+	if (ep->maxpacksize) {
+		/* whatever fits into a max. size packet */
+		maxsize = ep->maxpacksize;
+		ep->freqmax = (maxsize / (frame_bits >> 3))
+				<< (16 - ep->datainterval);
+	} else {
+		/* no max. packet size: just take 25% higher than nominal */
+		ep->freqmax = ep->freqn + (ep->freqn >> 2);
+		maxsize = ((ep->freqmax + 0xffff) * (frame_bits >> 3))
+				>> (16 - ep->datainterval);
+	}
+
+	if (ep->fill_max)
+		ep->curpacksize = ep->maxpacksize;
+	else
+		ep->curpacksize = maxsize;
+
+	if (snd_usb_get_speed(ep->chip->dev) != USB_SPEED_FULL)
+		packs_per_ms = 8 >> ep->datainterval;
+	else
+		packs_per_ms = 1;
+
+	if (is_playback && !snd_usb_endpoint_implict_feedback_sink(ep)) {
+		urb_packs = max(ep->chip->nrpacks, 1);
+		urb_packs = min(urb_packs, (unsigned int) MAX_PACKS);
+	} else {
+		urb_packs = 1;
+	}
+
+	urb_packs *= packs_per_ms;
+
+	if (sync_ep && !snd_usb_endpoint_implict_feedback_sink(ep))
+		urb_packs = min(urb_packs, 1U << sync_ep->syncinterval);
+
+	/* decide how many packets to be used */
+	if (is_playback && !snd_usb_endpoint_implict_feedback_sink(ep)) {
+		unsigned int minsize, maxpacks;
+		/* determine how small a packet can be */
+		minsize = (ep->freqn >> (16 - ep->datainterval))
+			  * (frame_bits >> 3);
+		/* with sync from device, assume it can be 12% lower */
+		if (sync_ep)
+			minsize -= minsize >> 3;
+		minsize = max(minsize, 1u);
+		total_packs = (period_bytes + minsize - 1) / minsize;
+		/* we need at least two URBs for queueing */
+		if (total_packs < 2) {
+			total_packs = 2;
+		} else {
+			/* and we don't want too long a queue either */
+			maxpacks = max(MAX_QUEUE * packs_per_ms, urb_packs * 2);
+			total_packs = min(total_packs, maxpacks);
+		}
+	} else {
+		while (urb_packs > 1 && urb_packs * maxsize >= period_bytes)
+			urb_packs >>= 1;
+		total_packs = MAX_URBS * urb_packs;
+	}
+
+	ep->nurbs = (total_packs + urb_packs - 1) / urb_packs;
+	if (ep->nurbs > MAX_URBS) {
+		/* too much... */
+		ep->nurbs = MAX_URBS;
+		total_packs = MAX_URBS * urb_packs;
+	} else if (ep->nurbs < 2) {
+		/* too little - we need at least two packets
+		 * to ensure contiguous playback/capture
+		 */
+		ep->nurbs = 2;
+	}
+
+	/* allocate and initialize data urbs */
+	for (i = 0; i < ep->nurbs; i++) {
+		struct snd_urb_ctx *u = &ep->urb[i];
+		u->index = i;
+		u->ep = ep;
+		u->packets = (i + 1) * total_packs / ep->nurbs
+			- i * total_packs / ep->nurbs;
+		u->buffer_size = maxsize * u->packets;
+
+		if (fmt->fmt_type == UAC_FORMAT_TYPE_II)
+			u->packets++; /* for transfer delimiter */
+		u->urb = usb_alloc_urb(u->packets, GFP_KERNEL);
+		if (!u->urb)
+			goto out_of_memory;
+
+		u->urb->transfer_buffer =
+			usb_alloc_coherent(ep->chip->dev, u->buffer_size,
+					   GFP_KERNEL, &u->urb->transfer_dma);
+		if (!u->urb->transfer_buffer)
+			goto out_of_memory;
+		u->urb->pipe = ep->pipe;
+		u->urb->transfer_flags = URB_ISO_ASAP | URB_NO_TRANSFER_DMA_MAP;
+		u->urb->interval = 1 << ep->datainterval;
+		u->urb->context = u;
+		u->urb->complete = snd_complete_urb;
+		INIT_LIST_HEAD(&u->ready_list);
+	}
+
+	return 0;
+
+out_of_memory:
+	release_urbs(ep, 0);
+	return -ENOMEM;
+}
+
+/*
+ * configure a sync endpoint
+ */
+static int sync_ep_set_params(struct snd_usb_endpoint *ep,
+			      struct snd_pcm_hw_params *hw_params,
+			      struct audioformat *fmt)
+{
+	int i;
+
+	ep->syncbuf = usb_alloc_coherent(ep->chip->dev, SYNC_URBS * 4,
+					 GFP_KERNEL, &ep->sync_dma);
+	if (!ep->syncbuf)
+		return -ENOMEM;
+
+	for (i = 0; i < SYNC_URBS; i++) {
+		struct snd_urb_ctx *u = &ep->urb[i];
+		u->index = i;
+		u->ep = ep;
+		u->packets = 1;
+		u->urb = usb_alloc_urb(1, GFP_KERNEL);
+		if (!u->urb)
+			goto out_of_memory;
+		u->urb->transfer_buffer = ep->syncbuf + i * 4;
+		u->urb->transfer_dma = ep->sync_dma + i * 4;
+		u->urb->transfer_buffer_length = 4;
+		u->urb->pipe = ep->pipe;
+		u->urb->transfer_flags = URB_ISO_ASAP |
+					 URB_NO_TRANSFER_DMA_MAP;
+		u->urb->number_of_packets = 1;
+		u->urb->interval = 1 << ep->syncinterval;
+		u->urb->context = u;
+		u->urb->complete = snd_complete_urb;
+	}
+
+	ep->nurbs = SYNC_URBS;
+
+	return 0;
+
+out_of_memory:
+	release_urbs(ep, 0);
+	return -ENOMEM;
+}
+
+/**
+ * snd_usb_endpoint_set_params: configure an snd_usb_endpoint
+ *
+ * @ep: the snd_usb_endpoint to configure
+ * @hw_params: the hardware parameters
+ * @fmt: the USB audio format information
+ * @sync_ep: the sync endpoint to use, if any
+ *
+ * Determine the number of URBs to be used on this endpoint.
+ * An endpoint must be configured before it can be started.
+ * An endpoint that is already running can not be reconfigured.
+ */
+int snd_usb_endpoint_set_params(struct snd_usb_endpoint *ep,
+				struct snd_pcm_hw_params *hw_params,
+				struct audioformat *fmt,
+				struct snd_usb_endpoint *sync_ep)
+{
+	int err;
+
+	if (ep->use_count != 0) {
+		snd_printk(KERN_WARNING "Unable to change format on ep #%x: already in use\n",
+			   ep->ep_num);
+		return -EBUSY;
+	}
+
+	/* release old buffers, if any */
+	release_urbs(ep, 0);
+
+	ep->datainterval = fmt->datainterval;
+	ep->maxpacksize = fmt->maxpacksize;
+	ep->fill_max = !!(fmt->attributes & UAC_EP_CS_ATTR_FILL_MAX);
+
+	if (snd_usb_get_speed(ep->chip->dev) == USB_SPEED_FULL)
+		ep->freqn = get_usb_full_speed_rate(params_rate(hw_params));
+	else
+		ep->freqn = get_usb_high_speed_rate(params_rate(hw_params));
+
+	/* calculate the frequency in 16.16 format */
+	ep->freqm = ep->freqn;
+	ep->freqshift = INT_MIN;
+
+	ep->phase = 0;
+
+	switch (ep->type) {
+	case  SND_USB_ENDPOINT_TYPE_DATA:
+		err = data_ep_set_params(ep, hw_params, fmt, sync_ep);
+		break;
+	case  SND_USB_ENDPOINT_TYPE_SYNC:
+		err = sync_ep_set_params(ep, hw_params, fmt);
+		break;
+	default:
+		err = -EINVAL;
+	}
+
+	snd_printdd(KERN_DEBUG "Setting params for ep #%x (type %d, %d urbs), ret=%d\n",
+		   ep->ep_num, ep->type, ep->nurbs, err);
+
+	return err;
+}
+
+/**
+ * snd_usb_endpoint_start: start an snd_usb_endpoint
+ *
+ * @ep: the endpoint to start
+ *
+ * A call to this function will increment the use count of the endpoint.
+ * In case it is not already running, the URBs for this endpoint will be
+ * submitted. Otherwise, this function does nothing.
+ *
+ * Must be balanced to calls of snd_usb_endpoint_stop().
+ *
+ * Returns an error if the URB submission failed, 0 in all other cases.
+ */
+int snd_usb_endpoint_start(struct snd_usb_endpoint *ep)
+{
+	int err;
+	unsigned int i;
+
+	if (ep->chip->shutdown)
+		return -EBADFD;
+
+	/* already running? */
+	if (++ep->use_count != 1)
+		return 0;
+
+	if (snd_BUG_ON(!test_bit(EP_FLAG_ACTIVATED, &ep->flags)))
+		return -EINVAL;
+
+	/* just to be sure */
+	deactivate_urbs(ep, 0, 1);
+	wait_clear_urbs(ep);
+
+	ep->active_mask = 0;
+	ep->unlink_mask = 0;
+	ep->phase = 0;
+
+	/*
+	 * If this endpoint has a data endpoint as implicit feedback source,
+	 * don't start the urbs here. Instead, mark them all as available,
+	 * wait for the record urbs to return and queue the playback urbs
+	 * from that context.
+	 */
+
+	set_bit(EP_FLAG_RUNNING, &ep->flags);
+
+	if (snd_usb_endpoint_implict_feedback_sink(ep)) {
+		for (i = 0; i < ep->nurbs; i++) {
+			struct snd_urb_ctx *ctx = ep->urb + i;
+			list_add_tail(&ctx->ready_list, &ep->ready_playback_urbs);
+		}
+
+		return 0;
+	}
+
+	for (i = 0; i < ep->nurbs; i++) {
+		struct urb *urb = ep->urb[i].urb;
+
+		if (snd_BUG_ON(!urb))
+			goto __error;
+
+		if (usb_pipeout(ep->pipe)) {
+			prepare_outbound_urb_sizes(ep, urb->context);
+			prepare_outbound_urb(ep, urb->context);
+		} else {
+			prepare_inbound_urb(ep, urb->context);
+		}
+
+		err = usb_submit_urb(urb, GFP_ATOMIC);
 		if (err < 0) {
-			snd_printk(KERN_ERR "cannot submit datapipe "
-				   "for urb %d, error %d: %s\n",
+			snd_printk(KERN_ERR "cannot submit urb %d, error %d: %s\n",
 				   i, err, usb_error_string(err));
 			goto __error;
 		}
-		set_bit(i, &subs->active_mask);
-	}
-	if (subs->syncpipe) {
-		for (i = 0; i < SYNC_URBS; i++) {
-			err = usb_submit_urb(subs->syncurb[i].urb, GFP_ATOMIC);
-			if (err < 0) {
-				snd_printk(KERN_ERR "cannot submit syncpipe "
-					   "for urb %d, error %d: %s\n",
-					   i, err, usb_error_string(err));
-				goto __error;
-			}
-			set_bit(i + 16, &subs->active_mask);
-		}
+		set_bit(i, &ep->active_mask);
 	}
+
 	return 0;
 
- __error:
-	// snd_pcm_stop(subs->pcm_substream, SNDRV_PCM_STATE_XRUN);
-	deactivate_urbs(subs, 0, 0);
+__error:
+	clear_bit(EP_FLAG_RUNNING, &ep->flags);
+	ep->use_count--;
+	deactivate_urbs(ep, 0, 0);
 	return -EPIPE;
 }
 
-
-/*
+/**
+ * snd_usb_endpoint_stop: stop an snd_usb_endpoint
+ *
+ * @ep: the endpoint to stop (may be NULL)
+ *
+ * A call to this function will decrement the use count of the endpoint.
+ * In case the last user has requested the endpoint stop, the URBs will
+ * actually be deactivated.
+ *
+ * Must be balanced to calls of snd_usb_endpoint_start().
  */
-static struct snd_urb_ops audio_urb_ops[2] = {
-	{
-		.prepare =	prepare_nodata_playback_urb,
-		.retire =	retire_playback_urb,
-		.prepare_sync =	prepare_playback_sync_urb,
-		.retire_sync =	retire_playback_sync_urb,
-	},
-	{
-		.prepare =	prepare_capture_urb,
-		.retire =	retire_capture_urb,
-		.prepare_sync =	prepare_capture_sync_urb,
-		.retire_sync =	retire_capture_sync_urb,
-	},
-};
+void snd_usb_endpoint_stop(struct snd_usb_endpoint *ep,
+			   int force, int can_sleep, int wait)
+{
+	if (!ep)
+		return;
 
-/*
- * initialize the substream instance.
+	if (snd_BUG_ON(ep->use_count == 0))
+		return;
+
+	if (snd_BUG_ON(!test_bit(EP_FLAG_ACTIVATED, &ep->flags)))
+		return;
+
+	if (--ep->use_count == 0) {
+		deactivate_urbs(ep, force, can_sleep);
+		ep->data_subs = NULL;
+		ep->sync_slave = NULL;
+		ep->retire_data_urb = NULL;
+		ep->prepare_data_urb = NULL;
+
+		if (wait)
+			wait_clear_urbs(ep);
+	}
+}
+
+/**
+ * snd_usb_endpoint_activate: activate an snd_usb_endpoint
+ *
+ * @ep: the endpoint to activate
+ *
+ * If the endpoint is not currently in use, this functions will select the
+ * correct alternate interface setting for the interface of this endpoint.
+ *
+ * In case of any active users, this functions does nothing.
+ *
+ * Returns an error if usb_set_interface() failed, 0 in all other
+ * cases.
  */
-
-void snd_usb_init_substream(struct snd_usb_stream *as,
-			    int stream, struct audioformat *fp)
+int snd_usb_endpoint_activate(struct snd_usb_endpoint *ep)
 {
-	struct snd_usb_substream *subs = &as->substream[stream];
-
-	INIT_LIST_HEAD(&subs->fmt_list);
-	spin_lock_init(&subs->lock);
-
-	subs->stream = as;
-	subs->direction = stream;
-	subs->dev = as->chip->dev;
-	subs->txfr_quirk = as->chip->txfr_quirk;
-	subs->ops = audio_urb_ops[stream];
-	if (snd_usb_get_speed(subs->dev) >= USB_SPEED_HIGH)
-		subs->ops.prepare_sync = prepare_capture_sync_urb_hs;
-
-	snd_usb_set_pcm_ops(as->pcm, stream);
-
-	list_add_tail(&fp->list, &subs->fmt_list);
-	subs->formats |= fp->formats;
-	subs->endpoint = fp->endpoint;
-	subs->num_formats++;
-	subs->fmt_type = fp->fmt_type;
-}
-
-int snd_usb_substream_playback_trigger(struct snd_pcm_substream *substream, int cmd)
-{
-	struct snd_usb_substream *subs = substream->runtime->private_data;
-
-	switch (cmd) {
-	case SNDRV_PCM_TRIGGER_START:
-	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
-		subs->ops.prepare = prepare_playback_urb;
+	if (ep->use_count != 0)
 		return 0;
-	case SNDRV_PCM_TRIGGER_STOP:
-		return deactivate_urbs(subs, 0, 0);
-	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
-		subs->ops.prepare = prepare_nodata_playback_urb;
+
+	if (!ep->chip->shutdown &&
+	    !test_and_set_bit(EP_FLAG_ACTIVATED, &ep->flags)) {
+		int ret;
+
+		ret = usb_set_interface(ep->chip->dev, ep->iface, ep->alt_idx);
+		if (ret < 0) {
+			snd_printk(KERN_ERR "%s() usb_set_interface() failed, ret = %d\n",
+						__func__, ret);
+			clear_bit(EP_FLAG_ACTIVATED, &ep->flags);
+			return ret;
+		}
+
 		return 0;
 	}
 
-	return -EINVAL;
+	return -EBUSY;
 }
 
-int snd_usb_substream_capture_trigger(struct snd_pcm_substream *substream, int cmd)
+/**
+ * snd_usb_endpoint_deactivate: deactivate an snd_usb_endpoint
+ *
+ * @ep: the endpoint to deactivate
+ *
+ * If the endpoint is not currently in use, this functions will select the
+ * alternate interface setting 0 for the interface of this endpoint.
+ *
+ * In case of any active users, this functions does nothing.
+ *
+ * Returns an error if usb_set_interface() failed, 0 in all other
+ * cases.
+ */
+int snd_usb_endpoint_deactivate(struct snd_usb_endpoint *ep)
 {
-	struct snd_usb_substream *subs = substream->runtime->private_data;
+	if (!ep)
+		return -EINVAL;
 
-	switch (cmd) {
-	case SNDRV_PCM_TRIGGER_START:
-		subs->ops.retire = retire_capture_urb;
-		return start_urbs(subs, substream->runtime);
-	case SNDRV_PCM_TRIGGER_STOP:
-		return deactivate_urbs(subs, 0, 0);
-	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
-		subs->ops.retire = retire_paused_capture_urb;
+	if (ep->use_count != 0)
 		return 0;
-	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
-		subs->ops.retire = retire_capture_urb;
+
+	if (!ep->chip->shutdown &&
+	    test_and_clear_bit(EP_FLAG_ACTIVATED, &ep->flags)) {
+		int ret;
+
+		ret = usb_set_interface(ep->chip->dev, ep->iface, 0);
+		if (ret < 0) {
+			snd_printk(KERN_ERR "%s(): usb_set_interface() failed, ret = %d\n",
+						__func__, ret);
+			return ret;
+		}
+
 		return 0;
 	}
 
-	return -EINVAL;
+	return -EBUSY;
 }
 
-int snd_usb_substream_prepare(struct snd_usb_substream *subs,
-			      struct snd_pcm_runtime *runtime)
+/**
+ * snd_usb_endpoint_free: Free the resources of an snd_usb_endpoint
+ *
+ * @ep: the list header of the endpoint to free
+ *
+ * This function does not care for the endpoint's use count but will tear
+ * down all the streaming URBs immediately and free all resources.
+ */
+void snd_usb_endpoint_free(struct list_head *head)
 {
-	/* clear urbs (to be sure) */
-	deactivate_urbs(subs, 0, 1);
-	wait_clear_urbs(subs);
+	struct snd_usb_endpoint *ep;
 
-	/* for playback, submit the URBs now; otherwise, the first hwptr_done
-	 * updates for all URBs would happen at the same time when starting */
-	if (subs->direction == SNDRV_PCM_STREAM_PLAYBACK) {
-		subs->ops.prepare = prepare_nodata_playback_urb;
-		return start_urbs(subs, runtime);
+	ep = list_entry(head, struct snd_usb_endpoint, list);
+	release_urbs(ep, 1);
+	kfree(ep);
+}
+
+/**
+ * snd_usb_handle_sync_urb: parse an USB sync packet
+ *
+ * @ep: the endpoint to handle the packet
+ * @sender: the sending endpoint
+ * @urb: the received packet
+ *
+ * This function is called from the context of an endpoint that received
+ * the packet and is used to let another endpoint object handle the payload.
+ */
+void snd_usb_handle_sync_urb(struct snd_usb_endpoint *ep,
+			     struct snd_usb_endpoint *sender,
+			     const struct urb *urb)
+{
+	int shift;
+	unsigned int f;
+	unsigned long flags;
+
+	snd_BUG_ON(ep == sender);
+
+	/*
+	 * In case the endpoint is operating in implicit feedback mode, prepare
+	 * a new outbound URB that has the same layout as the received packet
+	 * and add it to the list of pending urbs. queue_pending_output_urbs()
+	 * will take care of them later.
+	 */
+	if (snd_usb_endpoint_implict_feedback_sink(ep) &&
+	    ep->use_count != 0) {
+
+		/* implicit feedback case */
+		int i, bytes = 0;
+		struct snd_urb_ctx *in_ctx;
+		struct snd_usb_packet_info *out_packet;
+
+		in_ctx = urb->context;
+
+		/* Count overall packet size */
+		for (i = 0; i < in_ctx->packets; i++)
+			if (urb->iso_frame_desc[i].status == 0)
+				bytes += urb->iso_frame_desc[i].actual_length;
+
+		/*
+		 * skip empty packets. At least M-Audio's Fast Track Ultra stops
+		 * streaming once it received a 0-byte OUT URB
+		 */
+		if (bytes == 0)
+			return;
+
+		spin_lock_irqsave(&ep->lock, flags);
+		out_packet = ep->next_packet + ep->next_packet_write_pos;
+
+		/*
+		 * Iterate through the inbound packet and prepare the lengths
+		 * for the output packet. The OUT packet we are about to send
+		 * will have the same amount of payload bytes than the IN
+		 * packet we just received.
+		 */
+
+		out_packet->packets = in_ctx->packets;
+		for (i = 0; i < in_ctx->packets; i++) {
+			if (urb->iso_frame_desc[i].status == 0)
+				out_packet->packet_size[i] =
+					urb->iso_frame_desc[i].actual_length / ep->stride;
+			else
+				out_packet->packet_size[i] = 0;
+		}
+
+		ep->next_packet_write_pos++;
+		ep->next_packet_write_pos %= MAX_URBS;
+		spin_unlock_irqrestore(&ep->lock, flags);
+		queue_pending_output_urbs(ep);
+
+		return;
 	}
 
-	return 0;
+	/*
+	 * process after playback sync complete
+	 *
+	 * Full speed devices report feedback values in 10.14 format as samples
+	 * per frame, high speed devices in 16.16 format as samples per
+	 * microframe.
+	 *
+	 * Because the Audio Class 1 spec was written before USB 2.0, many high
+	 * speed devices use a wrong interpretation, some others use an
+	 * entirely different format.
+	 *
+	 * Therefore, we cannot predict what format any particular device uses
+	 * and must detect it automatically.
+	 */
+
+	if (urb->iso_frame_desc[0].status != 0 ||
+	    urb->iso_frame_desc[0].actual_length < 3)
+		return;
+
+	f = le32_to_cpup(urb->transfer_buffer);
+	if (urb->iso_frame_desc[0].actual_length == 3)
+		f &= 0x00ffffff;
+	else
+		f &= 0x0fffffff;
+
+	if (f == 0)
+		return;
+
+	if (unlikely(ep->freqshift == INT_MIN)) {
+		/*
+		 * The first time we see a feedback value, determine its format
+		 * by shifting it left or right until it matches the nominal
+		 * frequency value.  This assumes that the feedback does not
+		 * differ from the nominal value more than +50% or -25%.
+		 */
+		shift = 0;
+		while (f < ep->freqn - ep->freqn / 4) {
+			f <<= 1;
+			shift++;
+		}
+		while (f > ep->freqn + ep->freqn / 2) {
+			f >>= 1;
+			shift--;
+		}
+		ep->freqshift = shift;
+	} else if (ep->freqshift >= 0)
+		f <<= ep->freqshift;
+	else
+		f >>= -ep->freqshift;
+
+	if (likely(f >= ep->freqn - ep->freqn / 8 && f <= ep->freqmax)) {
+		/*
+		 * If the frequency looks valid, set it.
+		 * This value is referred to in prepare_playback_urb().
+		 */
+		spin_lock_irqsave(&ep->lock, flags);
+		ep->freqm = f;
+		spin_unlock_irqrestore(&ep->lock, flags);
+	} else {
+		/*
+		 * Out of range; maybe the shift value is wrong.
+		 * Reset it so that we autodetect again the next time.
+		 */
+		ep->freqshift = INT_MIN;
+	}
 }
 
diff --git a/sound/usb/endpoint.h b/sound/usb/endpoint.h
index 88eb63a636eb..ee2723fb174f 100644
--- a/sound/usb/endpoint.h
+++ b/sound/usb/endpoint.h
@@ -1,21 +1,29 @@
 #ifndef __USBAUDIO_ENDPOINT_H
 #define __USBAUDIO_ENDPOINT_H
 
-void snd_usb_init_substream(struct snd_usb_stream *as,
-			    int stream,
-			    struct audioformat *fp);
+#define SND_USB_ENDPOINT_TYPE_DATA     0
+#define SND_USB_ENDPOINT_TYPE_SYNC     1
 
-int snd_usb_init_substream_urbs(struct snd_usb_substream *subs,
-				unsigned int period_bytes,
-				unsigned int rate,
-				unsigned int frame_bits);
+struct snd_usb_endpoint *snd_usb_add_endpoint(struct snd_usb_audio *chip,
+					      struct usb_host_interface *alts,
+					      int ep_num, int direction, int type);
 
-void snd_usb_release_substream_urbs(struct snd_usb_substream *subs, int force);
+int snd_usb_endpoint_set_params(struct snd_usb_endpoint *ep,
+				struct snd_pcm_hw_params *hw_params,
+				struct audioformat *fmt,
+				struct snd_usb_endpoint *sync_ep);
 
-int snd_usb_substream_prepare(struct snd_usb_substream *subs,
-			      struct snd_pcm_runtime *runtime);
+int  snd_usb_endpoint_start(struct snd_usb_endpoint *ep);
+void snd_usb_endpoint_stop(struct snd_usb_endpoint *ep,
+			   int force, int can_sleep, int wait);
+int  snd_usb_endpoint_activate(struct snd_usb_endpoint *ep);
+int  snd_usb_endpoint_deactivate(struct snd_usb_endpoint *ep);
+void snd_usb_endpoint_free(struct list_head *head);
 
-int snd_usb_substream_playback_trigger(struct snd_pcm_substream *substream, int cmd);
-int snd_usb_substream_capture_trigger(struct snd_pcm_substream *substream, int cmd);
+int snd_usb_endpoint_implict_feedback_sink(struct snd_usb_endpoint *ep);
+
+void snd_usb_handle_sync_urb(struct snd_usb_endpoint *ep,
+			     struct snd_usb_endpoint *sender,
+			     const struct urb *urb);
 
 #endif /* __USBAUDIO_ENDPOINT_H */
diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c
index ab23869c01bb..4f40ba823163 100644
--- a/sound/usb/mixer.c
+++ b/sound/usb/mixer.c
@@ -486,7 +486,7 @@ static int set_cur_mix_value(struct usb_mixer_elem_info *cval, int channel,
 /*
  * TLV callback for mixer volume controls
  */
-static int mixer_vol_tlv(struct snd_kcontrol *kcontrol, int op_flag,
+int snd_usb_mixer_vol_tlv(struct snd_kcontrol *kcontrol, int op_flag,
 			 unsigned int size, unsigned int __user *_tlv)
 {
 	struct usb_mixer_elem_info *cval = kcontrol->private_data;
@@ -770,6 +770,26 @@ static void volume_control_quirks(struct usb_mixer_elem_info *cval,
 				  struct snd_kcontrol *kctl)
 {
 	switch (cval->mixer->chip->usb_id) {
+	case USB_ID(0x0763, 0x2081): /* M-Audio Fast Track Ultra 8R */
+	case USB_ID(0x0763, 0x2080): /* M-Audio Fast Track Ultra */
+		if (strcmp(kctl->id.name, "Effect Duration") == 0) {
+			snd_printk(KERN_INFO
+				"usb-audio: set quirk for FTU Effect Duration\n");
+			cval->min = 0x0000;
+			cval->max = 0x7f00;
+			cval->res = 0x0100;
+			break;
+		}
+		if (strcmp(kctl->id.name, "Effect Volume") == 0 ||
+		    strcmp(kctl->id.name, "Effect Feedback Volume") == 0) {
+			snd_printk(KERN_INFO
+				"usb-audio: set quirks for FTU Effect Feedback/Volume\n");
+			cval->min = 0x00;
+			cval->max = 0x7f;
+			break;
+		}
+		break;
+
 	case USB_ID(0x0471, 0x0101):
 	case USB_ID(0x0471, 0x0104):
 	case USB_ID(0x0471, 0x0105):
@@ -1121,9 +1141,6 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc,
 		len = snd_usb_copy_string_desc(state, nameid,
 				kctl->id.name, sizeof(kctl->id.name));
 
-	/* get min/max values */
-	get_min_max_with_quirks(cval, 0, kctl);
-
 	switch (control) {
 	case UAC_FU_MUTE:
 	case UAC_FU_VOLUME:
@@ -1155,17 +1172,7 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc,
 		}
 		append_ctl_name(kctl, control == UAC_FU_MUTE ?
 				" Switch" : " Volume");
-		if (control == UAC_FU_VOLUME) {
-			check_mapped_dB(map, cval);
-			if (cval->dBmin < cval->dBmax || !cval->initialized) {
-				kctl->tlv.c = mixer_vol_tlv;
-				kctl->vd[0].access |= 
-					SNDRV_CTL_ELEM_ACCESS_TLV_READ |
-					SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK;
-			}
-		}
 		break;
-
 	default:
 		if (! len)
 			strlcpy(kctl->id.name, audio_feature_info[control-1].name,
@@ -1173,6 +1180,19 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc,
 		break;
 	}
 
+	/* get min/max values */
+	get_min_max_with_quirks(cval, 0, kctl);
+
+	if (control == UAC_FU_VOLUME) {
+		check_mapped_dB(map, cval);
+		if (cval->dBmin < cval->dBmax || !cval->initialized) {
+			kctl->tlv.c = snd_usb_mixer_vol_tlv;
+			kctl->vd[0].access |=
+				SNDRV_CTL_ELEM_ACCESS_TLV_READ |
+				SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK;
+		}
+	}
+
 	range = (cval->max - cval->min) / cval->res;
 	/* Are there devices with volume range more than 255? I use a bit more
 	 * to be sure. 384 is a resolution magic number found on Logitech
@@ -1388,7 +1408,7 @@ static int parse_audio_mixer_unit(struct mixer_build *state, int unitid, void *r
 	for (pin = 0; pin < input_pins; pin++) {
 		err = parse_audio_unit(state, desc->baSourceID[pin]);
 		if (err < 0)
-			return err;
+			continue;
 		err = check_input_term(state, desc->baSourceID[pin], &iterm);
 		if (err < 0)
 			return err;
diff --git a/sound/usb/mixer.h b/sound/usb/mixer.h
index 81b2d8a32fb0..a7f3d45a8acf 100644
--- a/sound/usb/mixer.h
+++ b/sound/usb/mixer.h
@@ -68,4 +68,7 @@ int snd_usb_mixer_activate(struct usb_mixer_interface *mixer);
 int snd_usb_mixer_add_control(struct usb_mixer_interface *mixer,
 			      struct snd_kcontrol *kctl);
 
+int snd_usb_mixer_vol_tlv(struct snd_kcontrol *kcontrol, int op_flag,
+			  unsigned int size, unsigned int __user *_tlv);
+
 #endif /* __USBMIXER_H */
diff --git a/sound/usb/mixer_maps.c b/sound/usb/mixer_maps.c
index f1324c423835..41daaa24c25f 100644
--- a/sound/usb/mixer_maps.c
+++ b/sound/usb/mixer_maps.c
@@ -288,6 +288,15 @@ static struct usbmix_name_map scratch_live_map[] = {
 	{ 0 } /* terminator */
 };
 
+static struct usbmix_name_map ebox44_map[] = {
+	{ 4, NULL }, /* FU */
+	{ 6, NULL }, /* MU */
+	{ 7, NULL }, /* FU */
+	{ 10, NULL }, /* FU */
+	{ 11, NULL }, /* MU */
+	{ 0 }
+};
+
 /* "Gamesurround Muse Pocket LT" looks same like "Sound Blaster MP3+"
  *  most importand difference is SU[8], it should be set to "Capture Source"
  *  to make alsamixer and PA working properly.
@@ -371,6 +380,10 @@ static struct usbmix_ctl_map usbmix_ctl_maps[] = {
 		.map = scratch_live_map,
 		.ignore_ctl_error = 1,
 	},
+	{
+		.id = USB_ID(0x200c, 0x1018),
+		.map = ebox44_map,
+	},
 	{ 0 } /* terminator */
 };
 
diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c
index ab125ee0b0f0..41f4b6911920 100644
--- a/sound/usb/mixer_quirks.c
+++ b/sound/usb/mixer_quirks.c
@@ -42,6 +42,77 @@
 
 extern struct snd_kcontrol_new *snd_usb_feature_unit_ctl;
 
+/* private_free callback */
+static void usb_mixer_elem_free(struct snd_kcontrol *kctl)
+{
+	kfree(kctl->private_data);
+	kctl->private_data = NULL;
+}
+
+/* This function allows for the creation of standard UAC controls.
+ * See the quirks for M-Audio FTUs or Ebox-44.
+ * If you don't want to set a TLV callback pass NULL.
+ *
+ * Since there doesn't seem to be a devices that needs a multichannel
+ * version, we keep it mono for simplicity.
+ */
+static int snd_create_std_mono_ctl(struct usb_mixer_interface *mixer,
+				unsigned int unitid,
+				unsigned int control,
+				unsigned int cmask,
+				int val_type,
+				const char *name,
+				snd_kcontrol_tlv_rw_t *tlv_callback)
+{
+	int err;
+	struct usb_mixer_elem_info *cval;
+	struct snd_kcontrol *kctl;
+
+	cval = kzalloc(sizeof(*cval), GFP_KERNEL);
+	if (!cval)
+		return -ENOMEM;
+
+	cval->id = unitid;
+	cval->mixer = mixer;
+	cval->val_type = val_type;
+	cval->channels = 1;
+	cval->control = control;
+	cval->cmask = cmask;
+
+	/* get_min_max() is called only for integer volumes later,
+	 * so provide a short-cut for booleans */
+	cval->min = 0;
+	cval->max = 1;
+	cval->res = 0;
+	cval->dBmin = 0;
+	cval->dBmax = 0;
+
+	/* Create control */
+	kctl = snd_ctl_new1(snd_usb_feature_unit_ctl, cval);
+	if (!kctl) {
+		kfree(cval);
+		return -ENOMEM;
+	}
+
+	/* Set name */
+	snprintf(kctl->id.name, sizeof(kctl->id.name), name);
+	kctl->private_free = usb_mixer_elem_free;
+
+	/* set TLV */
+	if (tlv_callback) {
+		kctl->tlv.c = tlv_callback;
+		kctl->vd[0].access |=
+			SNDRV_CTL_ELEM_ACCESS_TLV_READ |
+			SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK;
+	}
+	/* Add control to mixer */
+	err = snd_usb_mixer_add_control(mixer, kctl);
+	if (err < 0)
+		return err;
+
+	return 0;
+}
+
 /*
  * Sound Blaster remote control configuration
  *
@@ -495,60 +566,218 @@ static int snd_nativeinstruments_create_mixer(struct usb_mixer_interface *mixer,
 }
 
 /* M-Audio FastTrack Ultra quirks */
+/* FTU Effect switch */
+struct snd_ftu_eff_switch_priv_val {
+	struct usb_mixer_interface *mixer;
+	int cached_value;
+	int is_cached;
+};
 
-/* private_free callback */
-static void usb_mixer_elem_free(struct snd_kcontrol *kctl)
+static int snd_ftu_eff_switch_info(struct snd_kcontrol *kcontrol,
+					struct snd_ctl_elem_info *uinfo)
 {
-	kfree(kctl->private_data);
-	kctl->private_data = NULL;
+	static const char *texts[8] = {"Room 1",
+				       "Room 2",
+				       "Room 3",
+				       "Hall 1",
+				       "Hall 2",
+				       "Plate",
+				       "Delay",
+				       "Echo"
+	};
+
+	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+	uinfo->count = 1;
+	uinfo->value.enumerated.items = 8;
+	if (uinfo->value.enumerated.item > 7)
+		uinfo->value.enumerated.item = 7;
+	strcpy(uinfo->value.enumerated.name,
+		texts[uinfo->value.enumerated.item]);
+
+	return 0;
 }
 
-static int snd_maudio_ftu_create_ctl(struct usb_mixer_interface *mixer,
-				     int in, int out, const char *name)
+static int snd_ftu_eff_switch_get(struct snd_kcontrol *kctl,
+					struct snd_ctl_elem_value *ucontrol)
 {
-	struct usb_mixer_elem_info *cval;
-	struct snd_kcontrol *kctl;
+	struct snd_usb_audio *chip;
+	struct usb_mixer_interface *mixer;
+	struct snd_ftu_eff_switch_priv_val *pval;
+	int err;
+	unsigned char value[2];
 
-	cval = kzalloc(sizeof(*cval), GFP_KERNEL);
-	if (!cval)
+	const int id = 6;
+	const int validx = 1;
+	const int val_len = 2;
+
+	value[0] = 0x00;
+	value[1] = 0x00;
+
+	pval = (struct snd_ftu_eff_switch_priv_val *)
+		kctl->private_value;
+
+	if (pval->is_cached) {
+		ucontrol->value.enumerated.item[0] = pval->cached_value;
+		return 0;
+	}
+
+	mixer = (struct usb_mixer_interface *) pval->mixer;
+	if (snd_BUG_ON(!mixer))
+		return -EINVAL;
+
+	chip = (struct snd_usb_audio *) mixer->chip;
+	if (snd_BUG_ON(!chip))
+		return -EINVAL;
+
+
+	err = snd_usb_ctl_msg(chip->dev,
+			usb_rcvctrlpipe(chip->dev, 0), UAC_GET_CUR,
+			USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN,
+			validx << 8, snd_usb_ctrl_intf(chip) | (id << 8),
+			value, val_len);
+	if (err < 0)
+		return err;
+
+	ucontrol->value.enumerated.item[0] = value[0];
+	pval->cached_value = value[0];
+	pval->is_cached = 1;
+
+	return 0;
+}
+
+static int snd_ftu_eff_switch_put(struct snd_kcontrol *kctl,
+					struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_usb_audio *chip;
+	struct snd_ftu_eff_switch_priv_val *pval;
+
+	struct usb_mixer_interface *mixer;
+	int changed, cur_val, err, new_val;
+	unsigned char value[2];
+
+
+	const int id = 6;
+	const int validx = 1;
+	const int val_len = 2;
+
+	changed = 0;
+
+	pval = (struct snd_ftu_eff_switch_priv_val *)
+		kctl->private_value;
+	cur_val = pval->cached_value;
+	new_val = ucontrol->value.enumerated.item[0];
+
+	mixer = (struct usb_mixer_interface *) pval->mixer;
+	if (snd_BUG_ON(!mixer))
+		return -EINVAL;
+
+	chip = (struct snd_usb_audio *) mixer->chip;
+	if (snd_BUG_ON(!chip))
+		return -EINVAL;
+
+	if (!pval->is_cached) {
+		/* Read current value */
+		err = snd_usb_ctl_msg(chip->dev,
+				usb_rcvctrlpipe(chip->dev, 0), UAC_GET_CUR,
+				USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN,
+				validx << 8, snd_usb_ctrl_intf(chip) | (id << 8),
+				value, val_len);
+		if (err < 0)
+			return err;
+
+		cur_val = value[0];
+		pval->cached_value = cur_val;
+		pval->is_cached = 1;
+	}
+	/* update value if needed */
+	if (cur_val != new_val) {
+		value[0] = new_val;
+		value[1] = 0;
+		err = snd_usb_ctl_msg(chip->dev,
+				usb_sndctrlpipe(chip->dev, 0), UAC_SET_CUR,
+				USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_OUT,
+				validx << 8, snd_usb_ctrl_intf(chip) | (id << 8),
+				value, val_len);
+		if (err < 0)
+			return err;
+
+		pval->cached_value = new_val;
+		pval->is_cached = 1;
+		changed = 1;
+	}
+
+	return changed;
+}
+
+static int snd_ftu_create_effect_switch(struct usb_mixer_interface *mixer)
+{
+	static struct snd_kcontrol_new template = {
+		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+		.name = "Effect Program Switch",
+		.index = 0,
+		.access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+		.info = snd_ftu_eff_switch_info,
+		.get = snd_ftu_eff_switch_get,
+		.put = snd_ftu_eff_switch_put
+	};
+
+	int err;
+	struct snd_kcontrol *kctl;
+	struct snd_ftu_eff_switch_priv_val *pval;
+
+	pval = kzalloc(sizeof(*pval), GFP_KERNEL);
+	if (!pval)
 		return -ENOMEM;
 
-	cval->id = 5;
-	cval->mixer = mixer;
-	cval->val_type = USB_MIXER_S16;
-	cval->channels = 1;
-	cval->control = out + 1;
-	cval->cmask = 1 << in;
+	pval->cached_value = 0;
+	pval->is_cached = 0;
+	pval->mixer = mixer;
 
-	kctl = snd_ctl_new1(snd_usb_feature_unit_ctl, cval);
+	template.private_value = (unsigned long) pval;
+	kctl = snd_ctl_new1(&template, mixer->chip);
 	if (!kctl) {
-		kfree(cval);
+		kfree(pval);
 		return -ENOMEM;
 	}
 
-	snprintf(kctl->id.name, sizeof(kctl->id.name), name);
-	kctl->private_free = usb_mixer_elem_free;
-	return snd_usb_mixer_add_control(mixer, kctl);
+	err = snd_ctl_add(mixer->chip->card, kctl);
+	if (err < 0)
+		return err;
+
+	return 0;
 }
 
-static int snd_maudio_ftu_create_mixer(struct usb_mixer_interface *mixer)
+/* Create volume controls for FTU devices*/
+static int snd_ftu_create_volume_ctls(struct usb_mixer_interface *mixer)
 {
 	char name[64];
+	unsigned int control, cmask;
 	int in, out, err;
 
+	const unsigned int id = 5;
+	const int val_type = USB_MIXER_S16;
+
 	for (out = 0; out < 8; out++) {
+		control = out + 1;
 		for (in = 0; in < 8; in++) {
+			cmask = 1 << in;
 			snprintf(name, sizeof(name),
-				 "AIn%d - Out%d Capture Volume", in  + 1, out + 1);
-			err = snd_maudio_ftu_create_ctl(mixer, in, out, name);
+				"AIn%d - Out%d Capture Volume",
+				in  + 1, out + 1);
+			err = snd_create_std_mono_ctl(mixer, id, control,
+							cmask, val_type, name,
+							&snd_usb_mixer_vol_tlv);
 			if (err < 0)
 				return err;
 		}
-
 		for (in = 8; in < 16; in++) {
+			cmask = 1 << in;
 			snprintf(name, sizeof(name),
-				 "DIn%d - Out%d Playback Volume", in - 7, out + 1);
-			err = snd_maudio_ftu_create_ctl(mixer, in, out, name);
+				"DIn%d - Out%d Playback Volume",
+				in - 7, out + 1);
+			err = snd_create_std_mono_ctl(mixer, id, control,
+							cmask, val_type, name,
+							&snd_usb_mixer_vol_tlv);
 			if (err < 0)
 				return err;
 		}
@@ -557,6 +786,191 @@ static int snd_maudio_ftu_create_mixer(struct usb_mixer_interface *mixer)
 	return 0;
 }
 
+/* This control needs a volume quirk, see mixer.c */
+static int snd_ftu_create_effect_volume_ctl(struct usb_mixer_interface *mixer)
+{
+	static const char name[] = "Effect Volume";
+	const unsigned int id = 6;
+	const int val_type = USB_MIXER_U8;
+	const unsigned int control = 2;
+	const unsigned int cmask = 0;
+
+	return snd_create_std_mono_ctl(mixer, id, control, cmask, val_type,
+					name, snd_usb_mixer_vol_tlv);
+}
+
+/* This control needs a volume quirk, see mixer.c */
+static int snd_ftu_create_effect_duration_ctl(struct usb_mixer_interface *mixer)
+{
+	static const char name[] = "Effect Duration";
+	const unsigned int id = 6;
+	const int val_type = USB_MIXER_S16;
+	const unsigned int control = 3;
+	const unsigned int cmask = 0;
+
+	return snd_create_std_mono_ctl(mixer, id, control, cmask, val_type,
+					name, snd_usb_mixer_vol_tlv);
+}
+
+/* This control needs a volume quirk, see mixer.c */
+static int snd_ftu_create_effect_feedback_ctl(struct usb_mixer_interface *mixer)
+{
+	static const char name[] = "Effect Feedback Volume";
+	const unsigned int id = 6;
+	const int val_type = USB_MIXER_U8;
+	const unsigned int control = 4;
+	const unsigned int cmask = 0;
+
+	return snd_create_std_mono_ctl(mixer, id, control, cmask, val_type,
+					name, NULL);
+}
+
+static int snd_ftu_create_effect_return_ctls(struct usb_mixer_interface *mixer)
+{
+	unsigned int cmask;
+	int err, ch;
+	char name[48];
+
+	const unsigned int id = 7;
+	const int val_type = USB_MIXER_S16;
+	const unsigned int control = 7;
+
+	for (ch = 0; ch < 4; ++ch) {
+		cmask = 1 << ch;
+		snprintf(name, sizeof(name),
+			"Effect Return %d Volume", ch + 1);
+		err = snd_create_std_mono_ctl(mixer, id, control,
+						cmask, val_type, name,
+						snd_usb_mixer_vol_tlv);
+		if (err < 0)
+			return err;
+	}
+
+	return 0;
+}
+
+static int snd_ftu_create_effect_send_ctls(struct usb_mixer_interface *mixer)
+{
+	unsigned int  cmask;
+	int err, ch;
+	char name[48];
+
+	const unsigned int id = 5;
+	const int val_type = USB_MIXER_S16;
+	const unsigned int control = 9;
+
+	for (ch = 0; ch < 8; ++ch) {
+		cmask = 1 << ch;
+		snprintf(name, sizeof(name),
+			"Effect Send AIn%d Volume", ch + 1);
+		err = snd_create_std_mono_ctl(mixer, id, control, cmask,
+						val_type, name,
+						snd_usb_mixer_vol_tlv);
+		if (err < 0)
+			return err;
+	}
+	for (ch = 8; ch < 16; ++ch) {
+		cmask = 1 << ch;
+		snprintf(name, sizeof(name),
+			"Effect Send DIn%d Volume", ch - 7);
+		err = snd_create_std_mono_ctl(mixer, id, control, cmask,
+						val_type, name,
+						snd_usb_mixer_vol_tlv);
+		if (err < 0)
+			return err;
+	}
+	return 0;
+}
+
+static int snd_ftu_create_mixer(struct usb_mixer_interface *mixer)
+{
+	int err;
+
+	err = snd_ftu_create_volume_ctls(mixer);
+	if (err < 0)
+		return err;
+
+	err = snd_ftu_create_effect_switch(mixer);
+	if (err < 0)
+		return err;
+	err = snd_ftu_create_effect_volume_ctl(mixer);
+	if (err < 0)
+		return err;
+
+	err = snd_ftu_create_effect_duration_ctl(mixer);
+	if (err < 0)
+		return err;
+
+	err = snd_ftu_create_effect_feedback_ctl(mixer);
+	if (err < 0)
+		return err;
+
+	err = snd_ftu_create_effect_return_ctls(mixer);
+	if (err < 0)
+		return err;
+
+	err = snd_ftu_create_effect_send_ctls(mixer);
+	if (err < 0)
+		return err;
+
+	return 0;
+}
+
+
+/*
+ * Create mixer for Electrix Ebox-44
+ *
+ * The mixer units from this device are corrupt, and even where they
+ * are valid they presents mono controls as L and R channels of
+ * stereo. So we create a good mixer in code.
+ */
+
+static int snd_ebox44_create_mixer(struct usb_mixer_interface *mixer)
+{
+	int err;
+
+	err = snd_create_std_mono_ctl(mixer, 4, 1, 0x0, USB_MIXER_INV_BOOLEAN,
+				"Headphone Playback Switch", NULL);
+	if (err < 0)
+		return err;
+	err = snd_create_std_mono_ctl(mixer, 4, 2, 0x1, USB_MIXER_S16,
+				"Headphone A Mix Playback Volume", NULL);
+	if (err < 0)
+		return err;
+	err = snd_create_std_mono_ctl(mixer, 4, 2, 0x2, USB_MIXER_S16,
+				"Headphone B Mix Playback Volume", NULL);
+	if (err < 0)
+		return err;
+
+	err = snd_create_std_mono_ctl(mixer, 7, 1, 0x0, USB_MIXER_INV_BOOLEAN,
+				"Output Playback Switch", NULL);
+	if (err < 0)
+		return err;
+	err = snd_create_std_mono_ctl(mixer, 7, 2, 0x1, USB_MIXER_S16,
+				"Output A Playback Volume", NULL);
+	if (err < 0)
+		return err;
+	err = snd_create_std_mono_ctl(mixer, 7, 2, 0x2, USB_MIXER_S16,
+				"Output B Playback Volume", NULL);
+	if (err < 0)
+		return err;
+
+	err = snd_create_std_mono_ctl(mixer, 10, 1, 0x0, USB_MIXER_INV_BOOLEAN,
+				"Input Capture Switch", NULL);
+	if (err < 0)
+		return err;
+	err = snd_create_std_mono_ctl(mixer, 10, 2, 0x1, USB_MIXER_S16,
+				"Input A Capture Volume", NULL);
+	if (err < 0)
+		return err;
+	err = snd_create_std_mono_ctl(mixer, 10, 2, 0x2, USB_MIXER_S16,
+				"Input B Capture Volume", NULL);
+	if (err < 0)
+		return err;
+
+	return 0;
+}
+
 void snd_emuusb_set_samplerate(struct snd_usb_audio *chip,
 			       unsigned char samplerate_id)
 {
@@ -600,7 +1014,7 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer)
 
 	case USB_ID(0x0763, 0x2080): /* M-Audio Fast Track Ultra */
 	case USB_ID(0x0763, 0x2081): /* M-Audio Fast Track Ultra 8R */
-		err = snd_maudio_ftu_create_mixer(mixer);
+		err = snd_ftu_create_mixer(mixer);
 		break;
 
 	case USB_ID(0x0b05, 0x1739):
@@ -619,6 +1033,10 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer)
 				snd_nativeinstruments_ta10_mixers,
 				ARRAY_SIZE(snd_nativeinstruments_ta10_mixers));
 		break;
+
+	case USB_ID(0x200c, 0x1018): /* Electrix Ebox-44 */
+		err = snd_ebox44_create_mixer(mixer);
+		break;
 	}
 
 	return err;
diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c
index 0eed6115c2d4..24839d932648 100644
--- a/sound/usb/pcm.c
+++ b/sound/usb/pcm.c
@@ -16,6 +16,7 @@
 
 #include <linux/init.h>
 #include <linux/slab.h>
+#include <linux/ratelimit.h>
 #include <linux/usb.h>
 #include <linux/usb/audio.h>
 #include <linux/usb/audio-v2.h>
@@ -34,6 +35,9 @@
 #include "clock.h"
 #include "power.h"
 
+#define SUBSTREAM_FLAG_DATA_EP_STARTED	0
+#define SUBSTREAM_FLAG_SYNC_EP_STARTED	1
+
 /* return the estimated delay based on USB frame counters */
 snd_pcm_uframes_t snd_usb_pcm_delay(struct snd_usb_substream *subs,
 				    unsigned int rate)
@@ -208,6 +212,84 @@ int snd_usb_init_pitch(struct snd_usb_audio *chip, int iface,
 	}
 }
 
+static int start_endpoints(struct snd_usb_substream *subs)
+{
+	int err;
+
+	if (!subs->data_endpoint)
+		return -EINVAL;
+
+	if (!test_and_set_bit(SUBSTREAM_FLAG_DATA_EP_STARTED, &subs->flags)) {
+		struct snd_usb_endpoint *ep = subs->data_endpoint;
+
+		snd_printdd(KERN_DEBUG "Starting data EP @%p\n", ep);
+
+		ep->data_subs = subs;
+		err = snd_usb_endpoint_start(ep);
+		if (err < 0) {
+			clear_bit(SUBSTREAM_FLAG_DATA_EP_STARTED, &subs->flags);
+			return err;
+		}
+	}
+
+	if (subs->sync_endpoint &&
+	    !test_and_set_bit(SUBSTREAM_FLAG_SYNC_EP_STARTED, &subs->flags)) {
+		struct snd_usb_endpoint *ep = subs->sync_endpoint;
+
+		snd_printdd(KERN_DEBUG "Starting sync EP @%p\n", ep);
+
+		ep->sync_slave = subs->data_endpoint;
+		err = snd_usb_endpoint_start(ep);
+		if (err < 0) {
+			clear_bit(SUBSTREAM_FLAG_SYNC_EP_STARTED, &subs->flags);
+			return err;
+		}
+	}
+
+	return 0;
+}
+
+static void stop_endpoints(struct snd_usb_substream *subs,
+			   int force, int can_sleep, int wait)
+{
+	if (test_and_clear_bit(SUBSTREAM_FLAG_SYNC_EP_STARTED, &subs->flags))
+		snd_usb_endpoint_stop(subs->sync_endpoint,
+				      force, can_sleep, wait);
+
+	if (test_and_clear_bit(SUBSTREAM_FLAG_DATA_EP_STARTED, &subs->flags))
+		snd_usb_endpoint_stop(subs->data_endpoint,
+				      force, can_sleep, wait);
+}
+
+static int activate_endpoints(struct snd_usb_substream *subs)
+{
+	if (subs->sync_endpoint) {
+		int ret;
+
+		ret = snd_usb_endpoint_activate(subs->sync_endpoint);
+		if (ret < 0)
+			return ret;
+	}
+
+	return snd_usb_endpoint_activate(subs->data_endpoint);
+}
+
+static int deactivate_endpoints(struct snd_usb_substream *subs)
+{
+	int reta, retb;
+
+	reta = snd_usb_endpoint_deactivate(subs->sync_endpoint);
+	retb = snd_usb_endpoint_deactivate(subs->data_endpoint);
+
+	if (reta < 0)
+		return reta;
+
+	if (retb < 0)
+		return retb;
+
+	return 0;
+}
+
 /*
  * find a matching format and set up the interface
  */
@@ -219,7 +301,7 @@ static int set_format(struct snd_usb_substream *subs, struct audioformat *fmt)
 	struct usb_interface *iface;
 	unsigned int ep, attr;
 	int is_playback = subs->direction == SNDRV_PCM_STREAM_PLAYBACK;
-	int err;
+	int err, implicit_fb = 0;
 
 	iface = usb_ifnum_to_if(dev, fmt->iface);
 	if (WARN_ON(!iface))
@@ -232,40 +314,11 @@ static int set_format(struct snd_usb_substream *subs, struct audioformat *fmt)
 	if (fmt == subs->cur_audiofmt)
 		return 0;
 
-	/* close the old interface */
-	if (subs->interface >= 0 && subs->interface != fmt->iface) {
-		if (usb_set_interface(subs->dev, subs->interface, 0) < 0) {
-			snd_printk(KERN_ERR "%d:%d:%d: return to setting 0 failed\n",
-				dev->devnum, fmt->iface, fmt->altsetting);
-			return -EIO;
-		}
-		subs->interface = -1;
-		subs->altset_idx = 0;
-	}
-
-	/* set interface */
-	if (subs->interface != fmt->iface || subs->altset_idx != fmt->altset_idx) {
-		if (usb_set_interface(dev, fmt->iface, fmt->altsetting) < 0) {
-			snd_printk(KERN_ERR "%d:%d:%d: usb_set_interface failed\n",
-				   dev->devnum, fmt->iface, fmt->altsetting);
-			return -EIO;
-		}
-		snd_printdd(KERN_INFO "setting usb interface %d:%d\n", fmt->iface, fmt->altsetting);
-		subs->interface = fmt->iface;
-		subs->altset_idx = fmt->altset_idx;
-	}
-
-	/* create a data pipe */
-	ep = fmt->endpoint & USB_ENDPOINT_NUMBER_MASK;
-	if (is_playback)
-		subs->datapipe = usb_sndisocpipe(dev, ep);
-	else
-		subs->datapipe = usb_rcvisocpipe(dev, ep);
-	subs->datainterval = fmt->datainterval;
-	subs->syncpipe = subs->syncinterval = 0;
-	subs->maxpacksize = fmt->maxpacksize;
-	subs->syncmaxsize = 0;
-	subs->fill_max = 0;
+	subs->data_endpoint = snd_usb_add_endpoint(subs->stream->chip,
+						   alts, fmt->endpoint, subs->direction,
+						   SND_USB_ENDPOINT_TYPE_DATA);
+	if (!subs->data_endpoint)
+		return -EINVAL;
 
 	/* we need a sync pipe in async OUT or adaptive IN mode */
 	/* check the number of EP, since some devices have broken
@@ -273,8 +326,25 @@ static int set_format(struct snd_usb_substream *subs, struct audioformat *fmt)
 	 * assume it as adaptive-out or sync-in.
 	 */
 	attr = fmt->ep_attr & USB_ENDPOINT_SYNCTYPE;
+
+	switch (subs->stream->chip->usb_id) {
+	case USB_ID(0x0763, 0x2080): /* M-Audio FastTrack Ultra */
+	case USB_ID(0x0763, 0x2081):
+		if (is_playback) {
+			implicit_fb = 1;
+			ep = 0x81;
+			iface = usb_ifnum_to_if(dev, 2);
+
+			if (!iface || iface->num_altsetting == 0)
+				return -EINVAL;
+
+			alts = &iface->altsetting[1];
+			goto add_sync_ep;
+		}
+	}
+
 	if (((is_playback && attr == USB_ENDPOINT_SYNC_ASYNC) ||
-	     (! is_playback && attr == USB_ENDPOINT_SYNC_ADAPTIVE)) &&
+	     (!is_playback && attr == USB_ENDPOINT_SYNC_ADAPTIVE)) &&
 	    altsd->bNumEndpoints >= 2) {
 		/* check sync-pipe endpoint */
 		/* ... and check descriptor size before accessing bSynchAddress
@@ -282,7 +352,8 @@ static int set_format(struct snd_usb_substream *subs, struct audioformat *fmt)
 		   the audio fields in the endpoint descriptors */
 		if ((get_endpoint(alts, 1)->bmAttributes & USB_ENDPOINT_XFERTYPE_MASK) != 0x01 ||
 		    (get_endpoint(alts, 1)->bLength >= USB_DT_ENDPOINT_AUDIO_SIZE &&
-		     get_endpoint(alts, 1)->bSynchAddress != 0)) {
+		     get_endpoint(alts, 1)->bSynchAddress != 0 &&
+		     !implicit_fb)) {
 			snd_printk(KERN_ERR "%d:%d:%d : invalid synch pipe\n",
 				   dev->devnum, fmt->iface, fmt->altsetting);
 			return -EINVAL;
@@ -290,33 +361,27 @@ static int set_format(struct snd_usb_substream *subs, struct audioformat *fmt)
 		ep = get_endpoint(alts, 1)->bEndpointAddress;
 		if (get_endpoint(alts, 0)->bLength >= USB_DT_ENDPOINT_AUDIO_SIZE &&
 		    (( is_playback && ep != (unsigned int)(get_endpoint(alts, 0)->bSynchAddress | USB_DIR_IN)) ||
-		     (!is_playback && ep != (unsigned int)(get_endpoint(alts, 0)->bSynchAddress & ~USB_DIR_IN)))) {
+		     (!is_playback && ep != (unsigned int)(get_endpoint(alts, 0)->bSynchAddress & ~USB_DIR_IN)) ||
+		     ( is_playback && !implicit_fb))) {
 			snd_printk(KERN_ERR "%d:%d:%d : invalid synch pipe\n",
 				   dev->devnum, fmt->iface, fmt->altsetting);
 			return -EINVAL;
 		}
-		ep &= USB_ENDPOINT_NUMBER_MASK;
-		if (is_playback)
-			subs->syncpipe = usb_rcvisocpipe(dev, ep);
-		else
-			subs->syncpipe = usb_sndisocpipe(dev, ep);
-		if (get_endpoint(alts, 1)->bLength >= USB_DT_ENDPOINT_AUDIO_SIZE &&
-		    get_endpoint(alts, 1)->bRefresh >= 1 &&
-		    get_endpoint(alts, 1)->bRefresh <= 9)
-			subs->syncinterval = get_endpoint(alts, 1)->bRefresh;
-		else if (snd_usb_get_speed(subs->dev) == USB_SPEED_FULL)
-			subs->syncinterval = 1;
-		else if (get_endpoint(alts, 1)->bInterval >= 1 &&
-			 get_endpoint(alts, 1)->bInterval <= 16)
-			subs->syncinterval = get_endpoint(alts, 1)->bInterval - 1;
-		else
-			subs->syncinterval = 3;
-		subs->syncmaxsize = le16_to_cpu(get_endpoint(alts, 1)->wMaxPacketSize);
-	}
 
-	/* always fill max packet size */
-	if (fmt->attributes & UAC_EP_CS_ATTR_FILL_MAX)
-		subs->fill_max = 1;
+		implicit_fb = (get_endpoint(alts, 1)->bmAttributes & USB_ENDPOINT_USAGE_MASK)
+				== USB_ENDPOINT_USAGE_IMPLICIT_FB;
+
+add_sync_ep:
+		subs->sync_endpoint = snd_usb_add_endpoint(subs->stream->chip,
+							   alts, ep, !subs->direction,
+							   implicit_fb ?
+								SND_USB_ENDPOINT_TYPE_DATA :
+								SND_USB_ENDPOINT_TYPE_SYNC);
+		if (!subs->sync_endpoint)
+			return -EINVAL;
+
+		subs->data_endpoint->sync_master = subs->sync_endpoint;
+	}
 
 	if ((err = snd_usb_init_pitch(subs->stream->chip, subs->interface, alts, fmt)) < 0)
 		return err;
@@ -390,15 +455,30 @@ static int snd_usb_hw_params(struct snd_pcm_substream *substream,
 	if (changed) {
 		mutex_lock(&subs->stream->chip->shutdown_mutex);
 		/* format changed */
-		snd_usb_release_substream_urbs(subs, 0);
-		/* influenced: period_bytes, channels, rate, format, */
-		ret = snd_usb_init_substream_urbs(subs, params_period_bytes(hw_params),
-						  params_rate(hw_params),
-						  snd_pcm_format_physical_width(params_format(hw_params)) *
-							params_channels(hw_params));
+		stop_endpoints(subs, 0, 0, 0);
+		deactivate_endpoints(subs);
+
+		ret = activate_endpoints(subs);
+		if (ret < 0)
+			goto unlock;
+
+		ret = snd_usb_endpoint_set_params(subs->data_endpoint, hw_params, fmt,
+						  subs->sync_endpoint);
+		if (ret < 0)
+			goto unlock;
+
+		if (subs->sync_endpoint)
+			ret = snd_usb_endpoint_set_params(subs->sync_endpoint,
+							  hw_params, fmt, NULL);
+unlock:
 		mutex_unlock(&subs->stream->chip->shutdown_mutex);
 	}
 
+	if (ret == 0) {
+		subs->interface = fmt->iface;
+		subs->altset_idx = fmt->altset_idx;
+	}
+
 	return ret;
 }
 
@@ -415,7 +495,7 @@ static int snd_usb_hw_free(struct snd_pcm_substream *substream)
 	subs->cur_rate = 0;
 	subs->period_bytes = 0;
 	mutex_lock(&subs->stream->chip->shutdown_mutex);
-	snd_usb_release_substream_urbs(subs, 0);
+	stop_endpoints(subs, 0, 1, 1);
 	mutex_unlock(&subs->stream->chip->shutdown_mutex);
 	return snd_pcm_lib_free_vmalloc_buffer(substream);
 }
@@ -435,19 +515,28 @@ static int snd_usb_pcm_prepare(struct snd_pcm_substream *substream)
 		return -ENXIO;
 	}
 
+	if (snd_BUG_ON(!subs->data_endpoint))
+		return -EIO;
+
 	/* some unit conversions in runtime */
-	subs->maxframesize = bytes_to_frames(runtime, subs->maxpacksize);
-	subs->curframesize = bytes_to_frames(runtime, subs->curpacksize);
+	subs->data_endpoint->maxframesize =
+		bytes_to_frames(runtime, subs->data_endpoint->maxpacksize);
+	subs->data_endpoint->curframesize =
+		bytes_to_frames(runtime, subs->data_endpoint->curpacksize);
 
 	/* reset the pointer */
 	subs->hwptr_done = 0;
 	subs->transfer_done = 0;
-	subs->phase = 0;
 	subs->last_delay = 0;
 	subs->last_frame_number = 0;
 	runtime->delay = 0;
 
-	return snd_usb_substream_prepare(subs, runtime);
+	/* for playback, submit the URBs now; otherwise, the first hwptr_done
+	 * updates for all URBs would happen at the same time when starting */
+	if (subs->direction == SNDRV_PCM_STREAM_PLAYBACK)
+		return start_endpoints(subs);
+
+	return 0;
 }
 
 static struct snd_pcm_hardware snd_usb_hardware =
@@ -842,16 +931,171 @@ static int snd_usb_pcm_open(struct snd_pcm_substream *substream, int direction)
 
 static int snd_usb_pcm_close(struct snd_pcm_substream *substream, int direction)
 {
+	int ret;
 	struct snd_usb_stream *as = snd_pcm_substream_chip(substream);
 	struct snd_usb_substream *subs = &as->substream[direction];
 
-	if (!as->chip->shutdown && subs->interface >= 0) {
-		usb_set_interface(subs->dev, subs->interface, 0);
-		subs->interface = -1;
-	}
+	stop_endpoints(subs, 0, 0, 0);
+	ret = deactivate_endpoints(subs);
 	subs->pcm_substream = NULL;
 	snd_usb_autosuspend(subs->stream->chip);
-	return 0;
+
+	return ret;
+}
+
+/* Since a URB can handle only a single linear buffer, we must use double
+ * buffering when the data to be transferred overflows the buffer boundary.
+ * To avoid inconsistencies when updating hwptr_done, we use double buffering
+ * for all URBs.
+ */
+static void retire_capture_urb(struct snd_usb_substream *subs,
+			       struct urb *urb)
+{
+	struct snd_pcm_runtime *runtime = subs->pcm_substream->runtime;
+	unsigned int stride, frames, bytes, oldptr;
+	int i, period_elapsed = 0;
+	unsigned long flags;
+	unsigned char *cp;
+
+	stride = runtime->frame_bits >> 3;
+
+	for (i = 0; i < urb->number_of_packets; i++) {
+		cp = (unsigned char *)urb->transfer_buffer + urb->iso_frame_desc[i].offset;
+		if (urb->iso_frame_desc[i].status && printk_ratelimit()) {
+			snd_printdd(KERN_ERR "frame %d active: %d\n", i, urb->iso_frame_desc[i].status);
+			// continue;
+		}
+		bytes = urb->iso_frame_desc[i].actual_length;
+		frames = bytes / stride;
+		if (!subs->txfr_quirk)
+			bytes = frames * stride;
+		if (bytes % (runtime->sample_bits >> 3) != 0) {
+#ifdef CONFIG_SND_DEBUG_VERBOSE
+			int oldbytes = bytes;
+#endif
+			bytes = frames * stride;
+			snd_printdd(KERN_ERR "Corrected urb data len. %d->%d\n",
+							oldbytes, bytes);
+		}
+		/* update the current pointer */
+		spin_lock_irqsave(&subs->lock, flags);
+		oldptr = subs->hwptr_done;
+		subs->hwptr_done += bytes;
+		if (subs->hwptr_done >= runtime->buffer_size * stride)
+			subs->hwptr_done -= runtime->buffer_size * stride;
+		frames = (bytes + (oldptr % stride)) / stride;
+		subs->transfer_done += frames;
+		if (subs->transfer_done >= runtime->period_size) {
+			subs->transfer_done -= runtime->period_size;
+			period_elapsed = 1;
+		}
+		spin_unlock_irqrestore(&subs->lock, flags);
+		/* copy a data chunk */
+		if (oldptr + bytes > runtime->buffer_size * stride) {
+			unsigned int bytes1 =
+					runtime->buffer_size * stride - oldptr;
+			memcpy(runtime->dma_area + oldptr, cp, bytes1);
+			memcpy(runtime->dma_area, cp + bytes1, bytes - bytes1);
+		} else {
+			memcpy(runtime->dma_area + oldptr, cp, bytes);
+		}
+	}
+
+	if (period_elapsed)
+		snd_pcm_period_elapsed(subs->pcm_substream);
+}
+
+static void prepare_playback_urb(struct snd_usb_substream *subs,
+				 struct urb *urb)
+{
+	struct snd_pcm_runtime *runtime = subs->pcm_substream->runtime;
+	struct snd_urb_ctx *ctx = urb->context;
+	unsigned int counts, frames, bytes;
+	int i, stride, period_elapsed = 0;
+	unsigned long flags;
+
+	stride = runtime->frame_bits >> 3;
+
+	frames = 0;
+	urb->number_of_packets = 0;
+	spin_lock_irqsave(&subs->lock, flags);
+	for (i = 0; i < ctx->packets; i++) {
+		counts = ctx->packet_size[i];
+		/* set up descriptor */
+		urb->iso_frame_desc[i].offset = frames * stride;
+		urb->iso_frame_desc[i].length = counts * stride;
+		frames += counts;
+		urb->number_of_packets++;
+		subs->transfer_done += counts;
+		if (subs->transfer_done >= runtime->period_size) {
+			subs->transfer_done -= runtime->period_size;
+			period_elapsed = 1;
+			if (subs->fmt_type == UAC_FORMAT_TYPE_II) {
+				if (subs->transfer_done > 0) {
+					/* FIXME: fill-max mode is not
+					 * supported yet */
+					frames -= subs->transfer_done;
+					counts -= subs->transfer_done;
+					urb->iso_frame_desc[i].length =
+						counts * stride;
+					subs->transfer_done = 0;
+				}
+				i++;
+				if (i < ctx->packets) {
+					/* add a transfer delimiter */
+					urb->iso_frame_desc[i].offset =
+						frames * stride;
+					urb->iso_frame_desc[i].length = 0;
+					urb->number_of_packets++;
+				}
+				break;
+			}
+		}
+		if (period_elapsed &&
+		    !snd_usb_endpoint_implict_feedback_sink(subs->data_endpoint)) /* finish at the period boundary */
+			break;
+	}
+	bytes = frames * stride;
+	if (subs->hwptr_done + bytes > runtime->buffer_size * stride) {
+		/* err, the transferred area goes over buffer boundary. */
+		unsigned int bytes1 =
+			runtime->buffer_size * stride - subs->hwptr_done;
+		memcpy(urb->transfer_buffer,
+		       runtime->dma_area + subs->hwptr_done, bytes1);
+		memcpy(urb->transfer_buffer + bytes1,
+		       runtime->dma_area, bytes - bytes1);
+	} else {
+		memcpy(urb->transfer_buffer,
+		       runtime->dma_area + subs->hwptr_done, bytes);
+	}
+	subs->hwptr_done += bytes;
+	if (subs->hwptr_done >= runtime->buffer_size * stride)
+		subs->hwptr_done -= runtime->buffer_size * stride;
+	runtime->delay += frames;
+	spin_unlock_irqrestore(&subs->lock, flags);
+	urb->transfer_buffer_length = bytes;
+	if (period_elapsed)
+		snd_pcm_period_elapsed(subs->pcm_substream);
+}
+
+/*
+ * process after playback data complete
+ * - decrease the delay count again
+ */
+static void retire_playback_urb(struct snd_usb_substream *subs,
+			       struct urb *urb)
+{
+	unsigned long flags;
+	struct snd_pcm_runtime *runtime = subs->pcm_substream->runtime;
+	int stride = runtime->frame_bits >> 3;
+	int processed = urb->transfer_buffer_length / stride;
+
+	spin_lock_irqsave(&subs->lock, flags);
+	if (processed > runtime->delay)
+		runtime->delay = 0;
+	else
+		runtime->delay -= processed;
+	spin_unlock_irqrestore(&subs->lock, flags);
 }
 
 static int snd_usb_playback_open(struct snd_pcm_substream *substream)
@@ -874,6 +1118,63 @@ static int snd_usb_capture_close(struct snd_pcm_substream *substream)
 	return snd_usb_pcm_close(substream, SNDRV_PCM_STREAM_CAPTURE);
 }
 
+static int snd_usb_substream_playback_trigger(struct snd_pcm_substream *substream,
+					      int cmd)
+{
+	struct snd_usb_substream *subs = substream->runtime->private_data;
+
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_START:
+	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+		subs->data_endpoint->prepare_data_urb = prepare_playback_urb;
+		subs->data_endpoint->retire_data_urb = retire_playback_urb;
+		subs->running = 1;
+		return 0;
+	case SNDRV_PCM_TRIGGER_STOP:
+		stop_endpoints(subs, 0, 0, 0);
+		subs->running = 0;
+		return 0;
+	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+		subs->data_endpoint->prepare_data_urb = NULL;
+		subs->data_endpoint->retire_data_urb = NULL;
+		subs->running = 0;
+		return 0;
+	}
+
+	return -EINVAL;
+}
+
+int snd_usb_substream_capture_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+	int err;
+	struct snd_usb_substream *subs = substream->runtime->private_data;
+
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_START:
+		err = start_endpoints(subs);
+		if (err < 0)
+			return err;
+
+		subs->data_endpoint->retire_data_urb = retire_capture_urb;
+		subs->running = 1;
+		return 0;
+	case SNDRV_PCM_TRIGGER_STOP:
+		stop_endpoints(subs, 0, 0, 0);
+		subs->running = 0;
+		return 0;
+	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+		subs->data_endpoint->retire_data_urb = NULL;
+		subs->running = 0;
+		return 0;
+	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+		subs->data_endpoint->retire_data_urb = retire_capture_urb;
+		subs->running = 1;
+		return 0;
+	}
+
+	return -EINVAL;
+}
+
 static struct snd_pcm_ops snd_usb_playback_ops = {
 	.open =		snd_usb_playback_open,
 	.close =	snd_usb_playback_close,
diff --git a/sound/usb/proc.c b/sound/usb/proc.c
index 961c9a250686..ebc1a5b5b3f1 100644
--- a/sound/usb/proc.c
+++ b/sound/usb/proc.c
@@ -25,6 +25,7 @@
 #include "usbaudio.h"
 #include "helper.h"
 #include "card.h"
+#include "endpoint.h"
 #include "proc.h"
 
 /* convert our full speed USB rate into sampling rate in Hz */
@@ -115,28 +116,33 @@ static void proc_dump_substream_formats(struct snd_usb_substream *subs, struct s
 	}
 }
 
+static void proc_dump_ep_status(struct snd_usb_substream *subs,
+				struct snd_usb_endpoint *ep,
+				struct snd_info_buffer *buffer)
+{
+	if (!ep)
+		return;
+	snd_iprintf(buffer, "    Packet Size = %d\n", ep->curpacksize);
+	snd_iprintf(buffer, "    Momentary freq = %u Hz (%#x.%04x)\n",
+		    snd_usb_get_speed(subs->dev) == USB_SPEED_FULL
+		    ? get_full_speed_hz(ep->freqm)
+		    : get_high_speed_hz(ep->freqm),
+		    ep->freqm >> 16, ep->freqm & 0xffff);
+	if (ep->freqshift != INT_MIN) {
+		int res = 16 - ep->freqshift;
+		snd_iprintf(buffer, "    Feedback Format = %d.%d\n",
+			    (ep->syncmaxsize > 3 ? 32 : 24) - res, res);
+	}
+}
+
 static void proc_dump_substream_status(struct snd_usb_substream *subs, struct snd_info_buffer *buffer)
 {
 	if (subs->running) {
-		unsigned int i;
 		snd_iprintf(buffer, "  Status: Running\n");
 		snd_iprintf(buffer, "    Interface = %d\n", subs->interface);
 		snd_iprintf(buffer, "    Altset = %d\n", subs->altset_idx);
-		snd_iprintf(buffer, "    URBs = %d [ ", subs->nurbs);
-		for (i = 0; i < subs->nurbs; i++)
-			snd_iprintf(buffer, "%d ", subs->dataurb[i].packets);
-		snd_iprintf(buffer, "]\n");
-		snd_iprintf(buffer, "    Packet Size = %d\n", subs->curpacksize);
-		snd_iprintf(buffer, "    Momentary freq = %u Hz (%#x.%04x)\n",
-			    snd_usb_get_speed(subs->dev) == USB_SPEED_FULL
-			    ? get_full_speed_hz(subs->freqm)
-			    : get_high_speed_hz(subs->freqm),
-			    subs->freqm >> 16, subs->freqm & 0xffff);
-		if (subs->freqshift != INT_MIN)
-			snd_iprintf(buffer, "    Feedback Format = %d.%d\n",
-				    (subs->syncmaxsize > 3 ? 32 : 24)
-						- (16 - subs->freqshift),
-				    16 - subs->freqshift);
+		proc_dump_ep_status(subs, subs->data_endpoint, buffer);
+		proc_dump_ep_status(subs, subs->sync_endpoint, buffer);
 	} else {
 		snd_iprintf(buffer, "  Status: Stop\n");
 	}
diff --git a/sound/usb/stream.c b/sound/usb/stream.c
index 5ff8010b2d6f..6b7d7a2b7baa 100644
--- a/sound/usb/stream.c
+++ b/sound/usb/stream.c
@@ -73,6 +73,31 @@ static void snd_usb_audio_pcm_free(struct snd_pcm *pcm)
 	}
 }
 
+/*
+ * initialize the substream instance.
+ */
+
+static void snd_usb_init_substream(struct snd_usb_stream *as,
+				   int stream,
+				   struct audioformat *fp)
+{
+	struct snd_usb_substream *subs = &as->substream[stream];
+
+	INIT_LIST_HEAD(&subs->fmt_list);
+	spin_lock_init(&subs->lock);
+
+	subs->stream = as;
+	subs->direction = stream;
+	subs->dev = as->chip->dev;
+	subs->txfr_quirk = as->chip->txfr_quirk;
+
+	snd_usb_set_pcm_ops(as->pcm, stream);
+
+	list_add_tail(&fp->list, &subs->fmt_list);
+	subs->formats |= fp->formats;
+	subs->num_formats++;
+	subs->fmt_type = fp->fmt_type;
+}
 
 /*
  * add this endpoint to the chip instance.
@@ -94,9 +119,9 @@ int snd_usb_add_audio_stream(struct snd_usb_audio *chip,
 		if (as->fmt_type != fp->fmt_type)
 			continue;
 		subs = &as->substream[stream];
-		if (!subs->endpoint)
+		if (!subs->data_endpoint)
 			continue;
-		if (subs->endpoint == fp->endpoint) {
+		if (subs->data_endpoint->ep_num == fp->endpoint) {
 			list_add_tail(&fp->list, &subs->fmt_list);
 			subs->num_formats++;
 			subs->formats |= fp->formats;
@@ -109,7 +134,7 @@ int snd_usb_add_audio_stream(struct snd_usb_audio *chip,
 		if (as->fmt_type != fp->fmt_type)
 			continue;
 		subs = &as->substream[stream];
-		if (subs->endpoint)
+		if (subs->data_endpoint)
 			continue;
 		err = snd_pcm_new_stream(as->pcm, stream, 1);
 		if (err < 0)
diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h
index 3e2b03577936..b8233ebe250f 100644
--- a/sound/usb/usbaudio.h
+++ b/sound/usb/usbaudio.h
@@ -36,6 +36,7 @@ struct snd_usb_audio {
 	struct snd_card *card;
 	struct usb_interface *pm_intf;
 	u32 usb_id;
+	struct mutex mutex;
 	struct mutex shutdown_mutex;
 	unsigned int shutdown:1;
 	unsigned int probing:1;
@@ -46,6 +47,7 @@ struct snd_usb_audio {
 	int num_suspended_intf;
 
 	struct list_head pcm_list;	/* list of pcm streams */
+	struct list_head ep_list;	/* list of audio-related endpoints */
 	int pcm_devs;
 
 	struct list_head midi_list;	/* list of midi interfaces */