Merge tag 'sound-3.12' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound updates from Takashi Iwai:
 "Changes are seen in a wide range of codes, mainly due to ASoC DAPM
  requirements; HD-audio shows a high peak in diffstat, it's just a
  removal of bunch of old static quirks.

  Some highlights:

   - HDPM: Updates for AIO/RayDAT support, TCO/sync support

   - RME96: Add PCM sync support

   - HD-audio:

     * A few HDMI/DP audio updates (CA assignment fix, stream switching
       fix, Intel DP device list support)
     * Device specific fixes (ASUS/CXT HP mic support, Thinkpad mic
       improvements, Chromebook fixes, STAC9228 Dell fixes)
     * Replace the all static quirks for AD codecs with the generic
       parser
     * WAKEEN support for handling irqs in the power saving mode

   - USB-audio: Clean up implicit fb handling and related codes

   - DAPM is now mandatory for ASoC CODEC drivers; all existing drivers
     have had some level of DAPM support added.  In addition, a lot of
     cleanups and improvements in DAPM.

   - Support for ASoC cross-platform compile test

   - New drivers and support for Analog Devices ADAU1702 and
     ADAU1401(a), Asahi Kasei Microdevices AK4554, Atmel AT91ASM9x5 and
     WM8904 based machines, Freescale S/PDIF and SSI AC'97, Renesas
     R-Car SoCs, Samsung Exynos5420 SoCs, Texas Instruments PCM1681 and
     PCM1792A and Wolfson Microelectronics WM8997

   - DT bindings for kirkwood and i.MX S/PDIF

   - Clean up and bug fixes: ssm2602, rt5640 and sgtl5000.

   - Core helpers for bitbanged AC'97 reset"

* tag 'sound-3.12' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (375 commits)
  ALSA: hda - Re-setup HDMI pin and audio infoframe on stream switches
  ALSA: hda - hdmi: Fallback to ALSA allocation when selecting CA
  ASoC: mxs-sgtl5000: Configure the dai_links as unidirectional
  ASoC: soc-pcm: Allow to specify unidirectional dai_link
  ASoC: fsl_spdif: Staticse non-exported symbols
  ASoC: ssm2602: Fix cache sync
  ASoC: Remove unused sysfs_registered field from snd_soc_codec struct
  ASoC: Remove unused debugfs_dapm field from snd_soc_{platform,codec} struct
  ASoC: Remove unused control_type field from snd_soc_codec struct
  ASoC: fsl: Add one blank space after ':=' in Makefile
  ASoC: fsl: Add wrapping for dev_dbg() in fsl_spdif.c
  ASoC: rt5640: change widget sequence for depop
  ASoC: dapm: Fix auto-disable for inverted controls
  ASoC: fsl: Drop SND_SOC_FSL_UTILS from SND_SOC_IMX_SPDIF
  ASoC: Samsung: Do not queue cyclic buffers multiple times
  ASoC: ep93xx-i2s: Remove unnecessary dev_set_drvdata()
  ASoC: designware_i2s: Remove unnecessary dev_set_drvdata()
  ASoC: fsl_spdif: remove redundant dev_err call in fsl_spdif_probe()
  ASoC: fsl: Add S/PDIF machine driver
  ASoc: kirkwood: Use the Kirkwood audio driver in Dove boards
  ...

Documentation/devicetree/bindings/misc/atmel-ssc.txt

@@ -7,9 +7,30 @@ Required properties:
 - reg: Should contain SSC registers location and length
 - interrupts: Should contain SSC interrupt
 
-Example:
+
+Required properties for devices compatible with "atmel,at91sam9g45-ssc":
+- dmas: DMA specifier, consisting of a phandle to DMA controller node,
+  the memory interface and SSC DMA channel ID (for tx and rx).
+  See Documentation/devicetree/bindings/dma/atmel-dma.txt for details.
+- dma-names: Must be "tx", "rx".
+
+Examples:
+- PDC transfer:
 ssc0: ssc@fffbc000 {
 	compatible = "atmel,at91rm9200-ssc";
 	reg = <0xfffbc000 0x4000>;
 	interrupts = <14 4 5>;
 };
+
+- DMA transfer:
+ssc0: ssc@f0010000 {
+      compatible = "atmel,at91sam9g45-ssc";
+      reg = <0xf0010000 0x4000>;
+      interrupts = <28 4 5>;
+      dmas = <&dma0 1 13>,
+	     <&dma0 1 14>;
+      dma-names = "tx", "rx";
+      pinctrl-names = "default";
+      pinctrl-0 = <&pinctrl_ssc0_tx &pinctrl_ssc0_rx>;
+      status = "disabled";
+};

Documentation/devicetree/bindings/serial/mrvl,pxa-ssp.txt

@@ -0,0 +1,65 @@
+Device tree bindings for Marvell PXA SSP ports
+
+Required properties:
+
+	- compatible:	Must be one of
+				mrvl,pxa25x-ssp
+				mvrl,pxa25x-nssp
+				mrvl,pxa27x-ssp
+				mrvl,pxa3xx-ssp
+				mvrl,pxa168-ssp
+				mrvl,pxa910-ssp
+				mrvl,ce4100-ssp
+				mrvl,lpss-ssp
+
+	- reg:		The memory base
+	- dmas:		Two dma phandles, one for rx, one for tx
+	- dma-names:	Must be "rx", "tx"
+
+
+Example for PXA3xx:
+
+	ssp0: ssp@41000000 {
+		compatible = "mrvl,pxa3xx-ssp";
+		reg = <0x41000000 0x40>;
+		ssp-id = <1>;
+		interrupts = <24>;
+		clock-names = "pxa27x-ssp.0";
+		dmas = <&dma 13
+			&dma 14>;
+		dma-names = "rx", "tx";
+	};
+
+	ssp1: ssp@41700000 {
+		compatible = "mrvl,pxa3xx-ssp";
+		reg = <0x41700000 0x40>;
+		ssp-id = <2>;
+		interrupts = <16>;
+		clock-names = "pxa27x-ssp.1";
+		dmas = <&dma 15
+			&dma 16>;
+		dma-names = "rx", "tx";
+	};
+
+	ssp2: ssp@41900000 {
+		compatibl3 = "mrvl,pxa3xx-ssp";
+		reg = <0x41900000 0x40>;
+		ssp-id = <3>;
+		interrupts = <0>;
+		clock-names = "pxa27x-ssp.2";
+		dmas = <&dma 66
+			&dma 67>;
+		dma-names = "rx", "tx";
+	};
+
+	ssp3: ssp@41a00000 {
+		compatible = "mrvl,pxa3xx-ssp";
+		reg = <0x41a00000 0x40>;
+		ssp-id = <4>;
+		interrupts = <13>;
+		clock-names = "pxa27x-ssp.3";
+		dmas = <&dma 2
+			&dma 3>;
+		dma-names = "rx", "tx";
+	};
+

Documentation/devicetree/bindings/sound/ak4554.c

@@ -0,0 +1,11 @@
+AK4554 ADC/DAC
+
+Required properties:
+
+  - compatible : "asahi-kasei,ak4554"
+
+Example:
+
+ak4554-adc-dac {
+	compatible = "asahi-kasei,ak4554";
+};

Documentation/devicetree/bindings/sound/alc5632.txt

@@ -13,6 +13,25 @@ Required properties:
   - #gpio-cells : Should be two. The first cell is the pin number and the
     second cell is used to specify optional parameters (currently unused).
 
+Pins on the device (for linking into audio routes):
+
+  * SPK_OUTP
+  * SPK_OUTN
+  * HP_OUT_L
+  * HP_OUT_R
+  * AUX_OUT_P
+  * AUX_OUT_N
+  * LINE_IN_L
+  * LINE_IN_R
+  * PHONE_P
+  * PHONE_N
+  * MIC1_P
+  * MIC1_N
+  * MIC2_P
+  * MIC2_N
+  * MICBIAS1
+  * DMICDAT
+
 Example:
 
 alc5632: alc5632@1e {

Documentation/devicetree/bindings/sound/atmel-sam9x5-wm8731-audio.txt

@@ -0,0 +1,35 @@
+* Atmel at91sam9x5ek wm8731 audio complex
+
+Required properties:
+  - compatible: "atmel,sam9x5-wm8731-audio"
+  - atmel,model: The user-visible name of this sound complex.
+  - atmel,ssc-controller: The phandle of the SSC controller
+  - atmel,audio-codec: The phandle of the WM8731 audio codec
+  - atmel,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.
+
+Available audio endpoints for the audio-routing table:
+
+Board connectors:
+ * Headphone Jack
+ * Line In Jack
+
+wm8731 pins:
+cf Documentation/devicetree/bindings/sound/wm8731.txt
+
+Example:
+sound {
+	compatible = "atmel,sam9x5-wm8731-audio";
+
+	atmel,model = "wm8731 @ AT91SAM9X5EK";
+
+	atmel,audio-routing =
+		"Headphone Jack", "RHPOUT",
+		"Headphone Jack", "LHPOUT",
+		"LLINEIN", "Line In Jack",
+		"RLINEIN", "Line In Jack";
+
+	atmel,ssc-controller = <&ssc0>;
+	atmel,audio-codec = <&wm8731>;
+};

Documentation/devicetree/bindings/sound/atmel-wm8904.txt

@@ -0,0 +1,55 @@
+Atmel ASoC driver with wm8904 audio codec complex
+
+Required properties:
+  - compatible: "atmel,asoc-wm8904"
+  - atmel,model: The user-visible name of this sound complex.
+  - atmel,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 WM8904's pins, and the jacks on the board:
+
+    WM8904 pins:
+
+    * IN1L
+    * IN1R
+    * IN2L
+    * IN2R
+    * IN3L
+    * IN3R
+    * HPOUTL
+    * HPOUTR
+    * LINEOUTL
+    * LINEOUTR
+    * MICBIAS
+
+    Board connectors:
+
+    * Headphone Jack
+    * Line In Jack
+    * Mic
+
+  - atmel,ssc-controller: The phandle of the SSC controller
+  - atmel,audio-codec: The phandle of the WM8904 audio codec
+
+Optional properties:
+  - pinctrl-names, pinctrl-0: Please refer to pinctrl-bindings.txt
+
+Example:
+sound {
+	compatible = "atmel,asoc-wm8904";
+	pinctrl-names = "default";
+	pinctrl-0 = <&pinctrl_pck0_as_mck>;
+
+	atmel,model = "wm8904 @ AT91SAM9N12EK";
+
+	atmel,audio-routing =
+		"Headphone Jack", "HPOUTL",
+		"Headphone Jack", "HPOUTR",
+		"IN2L", "Line In Jack",
+		"IN2R", "Line In Jack",
+		"Mic", "MICBIAS",
+		"IN1L", "Mic";
+
+	atmel,ssc-controller = <&ssc0>;
+	atmel,audio-codec = <&wm8904>;
+};

Documentation/devicetree/bindings/sound/fsl,spdif.txt

@@ -0,0 +1,54 @@
+Freescale Sony/Philips Digital Interface Format (S/PDIF) Controller
+
+The Freescale S/PDIF audio block is a stereo transceiver that allows the
+processor to receive and transmit digital audio via an coaxial cable or
+a fibre cable.
+
+Required properties:
+
+  - compatible : Compatible list, must contain "fsl,imx35-spdif".
+
+  - reg : Offset and length of the register set for the device.
+
+  - interrupts : Contains the spdif interrupt.
+
+  - dmas : Generic dma devicetree binding as described in
+  Documentation/devicetree/bindings/dma/dma.txt.
+
+  - dma-names : Two dmas have to be defined, "tx" and "rx".
+
+  - clocks : Contains an entry for each entry in clock-names.
+
+  - clock-names : Includes the following entries:
+	"core"		The core clock of spdif controller
+	"rxtx<0-7>"	Clock source list for tx and rx clock.
+			This clock list should be identical to
+			the source list connecting to the spdif
+			clock mux in "SPDIF Transceiver Clock
+			Diagram" of SoC reference manual. It
+			can also be referred to TxClk_Source
+			bit of register SPDIF_STC.
+
+Example:
+
+spdif: spdif@02004000 {
+	compatible = "fsl,imx35-spdif";
+	reg = <0x02004000 0x4000>;
+	interrupts = <0 52 0x04>;
+	dmas = <&sdma 14 18 0>,
+	       <&sdma 15 18 0>;
+	dma-names = "rx", "tx";
+
+	clocks = <&clks 197>, <&clks 3>,
+	       <&clks 197>, <&clks 107>,
+	       <&clks 0>, <&clks 118>,
+	       <&clks 62>, <&clks 139>,
+	       <&clks 0>;
+	clock-names = "core", "rxtx0",
+		"rxtx1", "rxtx2",
+		"rxtx3", "rxtx4",
+		"rxtx5", "rxtx6",
+		"rxtx7";
+
+	status = "okay";
+};

Documentation/devicetree/bindings/powerpc/fsl/ssi.txt → Documentation/devicetree/bindings/sound/fsl,ssi.txt

@@ -43,10 +43,22 @@ Required properties:
                     together.  This would still allow different sample sizes,
                     but not different sample rates.
 
+Required are also ac97 link bindings if ac97 is used. See
+Documentation/devicetree/bindings/sound/soc-ac97link.txt for the necessary
+bindings.
+
 Optional properties:
 - codec-handle:     Phandle to a 'codec' node that defines an audio
                     codec connected to this SSI.  This node is typically
                     a child of an I2C or other control node.
+- fsl,fiq-stream-filter: Bool property. Disabled DMA and use FIQ instead to
+		    filter the codec stream. This is necessary for some boards
+		    where an incompatible codec is connected to this SSI, e.g.
+		    on pca100 and pcm043.
+- dmas:		    Generic dma devicetree binding as described in
+		    Documentation/devicetree/bindings/dma/dma.txt.
+- dma-names:	    Two dmas have to be defined, "tx" and "rx", if fsl,imx-fiq
+		    is not defined.
 
 Child 'codec' node required properties:
 - compatible:       Compatible list, contains the name of the codec

Documentation/devicetree/bindings/sound/imx-audio-spdif.txt

@@ -0,0 +1,34 @@
+Freescale i.MX audio complex with S/PDIF transceiver
+
+Required properties:
+
+  - compatible : "fsl,imx-audio-spdif"
+
+  - model : The user-visible name of this sound complex
+
+  - spdif-controller : The phandle of the i.MX S/PDIF controller
+
+
+Optional properties:
+
+  - spdif-out : This is a boolean property. If present, the transmitting
+    function of S/PDIF will be enabled, indicating there's a physical
+    S/PDIF out connector/jack on the board or it's connecting to some
+    other IP block, such as an HDMI encoder/display-controller.
+
+  - spdif-in : This is a boolean property. If present, the receiving
+    function of S/PDIF will be enabled, indicating there's a physical
+    S/PDIF in connector/jack on the board.
+
+* Note: At least one of these two properties should be set in the DT binding.
+
+
+Example:
+
+sound-spdif {
+	compatible = "fsl,imx-audio-spdif";
+	model = "imx-spdif";
+	spdif-controller = <&spdif>;
+	spdif-out;
+	spdif-in;
+};

Documentation/devicetree/bindings/sound/imx-audmux.txt

@@ -5,6 +5,15 @@ Required properties:
   or "fsl,imx31-audmux" for the version firstly used on i.MX31.
 - reg : Should contain AUDMUX registers location and length
 
+An initial configuration can be setup using child nodes.
+
+Required properties of optional child nodes:
+- fsl,audmux-port : Integer of the audmux port that is configured by this
+  child node.
+- fsl,port-config : List of configuration options for the specific port. For
+  imx31-audmux and above, it is a list of tuples <ptcr pdcr>. For
+  imx21-audmux it is a list of pcr values.
+
 Example:
 
 audmux@021d8000 {

Documentation/devicetree/bindings/sound/mrvl,pxa-ssp.txt

@@ -0,0 +1,28 @@
+Marvell PXA SSP CPU DAI bindings
+
+Required properties:
+
+	compatible	Must be "mrvl,pxa-ssp-dai"
+	port		A phandle reference to a PXA ssp upstream device
+
+Example:
+
+	/* upstream device */
+
+	ssp0: ssp@41000000 {
+		compatible = "mrvl,pxa3xx-ssp";
+		reg = <0x41000000 0x40>;
+		interrupts = <24>;
+		clock-names = "pxa27x-ssp.0";
+		dmas = <&dma 13
+			&dma 14>;
+		dma-names = "rx", "tx";
+	};
+
+	/* DAI as user */
+
+	ssp_dai0: ssp_dai@0 {
+		compatible = "mrvl,pxa-ssp-dai";
+		port = <&ssp0>;
+	};
+

Documentation/devicetree/bindings/sound/mrvl,pxa2xx-pcm.txt

@@ -0,0 +1,15 @@
+DT bindings for ARM PXA2xx PCM platform driver
+
+This is just a dummy driver that registers the PXA ASoC platform driver.
+It does not have any resources assigned.
+
+Required properties:
+
+	- compatible		'mrvl,pxa-pcm-audio'
+
+Example:
+
+	pxa_pcm_audio: snd_soc_pxa_audio {
+		compatible = "mrvl,pxa-pcm-audio";
+	};
+

Documentation/devicetree/bindings/sound/mvebu-audio.txt

@@ -0,0 +1,29 @@
+* mvebu (Kirkwood, Dove, Armada 370) audio controller
+
+Required properties:
+
+- compatible: "marvell,mvebu-audio"
+
+- reg: physical base address of the controller and length of memory mapped
+  region.
+
+- interrupts: list of two irq numbers.
+  The first irq is used for data flow and the second one is used for errors.
+
+- clocks: one or two phandles.
+  The first one is mandatory and defines the internal clock.
+  The second one is optional and defines an external clock.
+
+- clock-names: names associated to the clocks:
+	"internal" for the internal clock
+	"extclk" for the external clock
+
+Example:
+
+i2s1: audio-controller@b4000 {
+	compatible = "marvell,mvebu-audio";
+	reg = <0xb4000 0x2210>;
+	interrupts = <21>, <22>;
+	clocks = <&gate_clk 13>;
+	clock-names = "internal";
+};

Documentation/devicetree/bindings/sound/nvidia,tegra-audio-alc5632.txt

@@ -11,28 +11,8 @@ Required properties:
 - 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 ALC5632's pins:
-
-  ALC5632 pins:
-
-  * SPK_OUTP
-  * SPK_OUTN
-  * HP_OUT_L
-  * HP_OUT_R
-  * AUX_OUT_P
-  * AUX_OUT_N
-  * LINE_IN_L
-  * LINE_IN_R
-  * PHONE_P
-  * PHONE_N
-  * MIC1_P
-  * MIC1_N
-  * MIC2_P
-  * MIC2_N
-  * MICBIAS1
-  * DMICDAT
-
-  Board connectors:
+  sinks are the ALC5632's pins as documented in the binding for the device
+  and:
 
   * Headset Stereophone
   * Int Spk

Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5640.txt

@@ -11,32 +11,12 @@ Required properties:
 - 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 RT5640's pins, and the jacks on the board:
-
-  RT5640 pins:
-
-  * DMIC1
-  * DMIC2
-  * MICBIAS1
-  * IN1P
-  * IN1R
-  * IN2P
-  * IN2R
-  * HPOL
-  * HPOR
-  * LOUTL
-  * LOUTR
-  * MONOP
-  * MONON
-  * SPOLP
-  * SPOLN
-  * SPORP
-  * SPORN
-
-  Board connectors:
+  sinks are the RT5640's pins (as documented in its binding), and the jacks
+  on the board:
 
   * Headphones
   * Speakers
+  * Mic Jack
 
 - nvidia,i2s-controller : The phandle of the Tegra I2S controller that's
   connected to the CODEC.

Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm8753.txt

@@ -11,31 +11,8 @@ Required properties:
 - 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:
+  sinks are the WM8753's pins as documented in the binding for the WM8753,
+  and the jacks on the board:
 
   * Headphone Jack
   * Mic Jack

Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm8903.txt

@@ -11,28 +11,8 @@ Required properties:
 - 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 WM8903's pins, and the jacks on the board:
-
-  WM8903 pins:
-
-  * IN1L
-  * IN1R
-  * IN2L
-  * IN2R
-  * IN3L
-  * IN3R
-  * DMICDAT
-  * HPOUTL
-  * HPOUTR
-  * LINEOUTL
-  * LINEOUTR
-  * LOP
-  * LON
-  * ROP
-  * RON
-  * MICBIAS
-
-  Board connectors:
+  sinks are the WM8903's pins (documented in the WM8903 binding document),
+  and the jacks on the board:
 
   * Headphone Jack
   * Int Spk

Documentation/devicetree/bindings/sound/pcm1792a.txt

@@ -0,0 +1,18 @@
+Texas Instruments pcm1792a DT bindings
+
+This driver supports the SPI bus.
+
+Required properties:
+
+ - compatible: "ti,pcm1792a"
+
+For required properties on SPI, please consult
+Documentation/devicetree/bindings/spi/spi-bus.txt
+
+Examples:
+
+	codec_spi: 1792a@0 {
+		compatible = "ti,pcm1792a";
+		spi-max-frequency = <600000>;
+	};
+

Documentation/devicetree/bindings/sound/rt5640.txt

@@ -18,6 +18,26 @@ Optional properties:
 
 - realtek,ldo1-en-gpios : The GPIO that controls the CODEC's LDO1_EN pin.
 
+Pins on the device (for linking into audio routes):
+
+  * DMIC1
+  * DMIC2
+  * MICBIAS1
+  * IN1P
+  * IN1R
+  * IN2P
+  * IN2R
+  * HPOL
+  * HPOR
+  * LOUTL
+  * LOUTR
+  * MONOP
+  * MONON
+  * SPOLP
+  * SPOLN
+  * SPORP
+  * SPORN
+
 Example:
 
 rt5640 {

Documentation/devicetree/bindings/sound/samsung-i2s.txt

@@ -2,7 +2,15 @@
 
 Required SoC Specific Properties:
 
-- compatible : "samsung,i2s-v5"
+- compatible : should be one of the following.
+   - samsung,s3c6410-i2s: for 8/16/24bit stereo I2S.
+   - samsung,s5pv210-i2s: for 8/16/24bit multichannel(5.1) I2S with
+     secondary fifo, s/w reset control and internal mux for root clk src.
+   - samsung,exynos5420-i2s: for 8/16/24bit multichannel(7.1) I2S with
+     secondary fifo, s/w reset control, internal mux for root clk src and
+     TDM support. TDM (Time division multiplexing) is to allow transfer of
+     multiple channel audio data on single data line.
+
 - reg: physical base address of the controller and length of memory mapped
   region.
 - dmas: list of DMA controller phandle and DMA request line ordered pairs.
@@ -21,13 +29,6 @@ Required SoC Specific Properties:
 
 Optional SoC Specific Properties:
 
-- samsung,supports-6ch: If the I2S Primary sound source has 5.1 Channel
-  support, this flag is enabled.
-- samsung,supports-rstclr: This flag should be set if I2S software reset bit
-  control is required. When this flag is set I2S software reset bit will be
-  enabled or disabled based on need.
-- samsung,supports-secdai:If I2S block has a secondary FIFO and internal DMA,
-  then this flag is enabled.
 - samsung,idma-addr: Internal DMA register base address of the audio
   sub system(used in secondary sound source).
 - pinctrl-0: Should specify pin control groups used for this controller.
@@ -36,7 +37,7 @@ Optional SoC Specific Properties:
 Example:
 
 i2s0: i2s@03830000 {
-	compatible = "samsung,i2s-v5";
+	compatible = "samsung,s5pv210-i2s";
 	reg = <0x03830000 0x100>;
 	dmas = <&pdma0 10
 		&pdma0 9
@@ -46,9 +47,6 @@ i2s0: i2s@03830000 {
 		<&clock_audss EXYNOS_I2S_BUS>,
 		<&clock_audss EXYNOS_SCLK_I2S>;
 	clock-names = "iis", "i2s_opclk0", "i2s_opclk1";
-	samsung,supports-6ch;
-	samsung,supports-rstclr;
-	samsung,supports-secdai;
 	samsung,idma-addr = <0x03000000>;
 	pinctrl-names = "default";
 	pinctrl-0 = <&i2s0_bus>;

Documentation/devicetree/bindings/sound/soc-ac97link.txt

@@ -0,0 +1,28 @@
+AC97 link bindings
+
+These bindings can be included within any other device node.
+
+Required properties:
+ - pinctrl-names: Has to contain following states to setup the correct
+   pinmuxing for the used gpios:
+	"ac97-running": AC97-link is active
+	"ac97-reset": AC97-link reset state
+	"ac97-warm-reset": AC97-link warm reset state
+ - ac97-gpios: List of gpio phandles with args in the order ac97-sync,
+   ac97-sdata, ac97-reset
+
+
+Example:
+
+ssi {
+	...
+
+	pinctrl-names = "default", "ac97-running", "ac97-reset", "ac97-warm-reset";
+	pinctrl-0 = <&ac97link_running>;
+	pinctrl-1 = <&ac97link_running>;
+	pinctrl-2 = <&ac97link_reset>;
+	pinctrl-3 = <&ac97link_warm_reset>;
+	ac97-gpios = <&gpio3 20 0 &gpio3 22 0 &gpio3 28 0>;
+
+	...
+};

Documentation/devicetree/bindings/sound/ti,pcm1681.txt

@@ -0,0 +1,15 @@
+Texas Instruments PCM1681 8-channel PWM Processor
+
+Required properties:
+
+ - compatible:		Should contain "ti,pcm1681".
+ - reg:			The i2c address. Should contain <0x4c>.
+
+Examples:
+
+	i2c_bus {
+		pcm1681@4c {
+			compatible = "ti,pcm1681";
+			reg = <0x4c>;
+		};
+	};

Documentation/devicetree/bindings/sound/tlv320aic3x.txt

@@ -3,7 +3,14 @@ Texas Instruments - tlv320aic3x Codec module
 The tlv320aic3x serial control bus communicates through I2C protocols
 
 Required properties:
-- compatible - "string" -  "ti,tlv320aic3x"
+
+- compatible - "string" - One of:
+    "ti,tlv320aic3x" - Generic TLV320AIC3x device
+    "ti,tlv320aic33" - TLV320AIC33
+    "ti,tlv320aic3007" - TLV320AIC3007
+    "ti,tlv320aic3106" - TLV320AIC3106
+
+
 - reg - <int> -  I2C slave address
 
 

Documentation/devicetree/bindings/sound/wm8731.txt

@@ -16,3 +16,12 @@ codec: wm8731@1a {
 	compatible = "wlf,wm8731";
 	reg = <0x1a>;
 };
+
+Available audio endpoints for an audio-routing table:
+ * LOUT: Left Channel Line Output
+ * ROUT: Right Channel Line Output
+ * LHPOUT: Left Channel Headphone Output
+ * RHPOUT: Right Channel Headphone Output
+ * LLINEIN: Left Channel Line Input
+ * RLINEIN: Right Channel Line Input
+ * MICIN: Microphone Input

Documentation/devicetree/bindings/sound/wm8753.txt

@@ -10,9 +10,31 @@ Required properties:
   - reg : the I2C address of the device for I2C, the chip select
           number for SPI.
 
+Pins on the device (for linking into audio routes):
+
+  * LOUT1
+  * LOUT2
+  * ROUT1
+  * ROUT2
+  * MONO1
+  * MONO2
+  * OUT3
+  * OUT4
+  * LINE1
+  * LINE2
+  * RXP
+  * RXN
+  * ACIN
+  * ACOP
+  * MIC1N
+  * MIC1
+  * MIC2N
+  * MIC2
+  * Mic Bias
+
 Example:
 
-codec: wm8737@1a {
+codec: wm8753@1a {
 	compatible = "wlf,wm8753";
 	reg = <0x1a>;
 };

Documentation/devicetree/bindings/sound/wm8903.txt

@@ -28,6 +28,25 @@ Optional properties:
     performed. If any entry has the value 0xffffffff, that GPIO's
     configuration will not be modified.
 
+Pins on the device (for linking into audio routes):
+
+  * IN1L
+  * IN1R
+  * IN2L
+  * IN2R
+  * IN3L
+  * IN3R
+  * DMICDAT
+  * HPOUTL
+  * HPOUTR
+  * LINEOUTL
+  * LINEOUTR
+  * LOP
+  * LON
+  * ROP
+  * RON
+  * MICBIAS
+
 Example:
 
 codec: wm8903@1a {

Documentation/devicetree/bindings/sound/wm8994.txt

@@ -32,6 +32,10 @@ Optional properties:
     The second cell is the flags, encoded as the trigger masks from
     Documentation/devicetree/bindings/interrupts.txt
 
+  - clocks : A list of up to two phandle and clock specifier pairs
+  - clock-names : A list of clock names sorted in the same order as clocks.
+                  Valid clock names are "MCLK1" and "MCLK2".
+
   - wlf,gpio-cfg : A list of GPIO configuration register values. If absent,
     no configuration of these registers is performed. If any value is
     over 0xffff then the register will be left as default. If present 11

Documentation/sound/alsa/HD-Audio-Models.txt

@@ -244,6 +244,7 @@ STAC9227/9228/9229/927x
   5stack-no-fp	D965 5stack without front panel
   dell-3stack	Dell Dimension E520
   dell-bios	Fixes with Dell BIOS setup
+  dell-bios-amic Fixes with Dell BIOS setup including analog mic
   volknob	Fixes with volume-knob widget 0x24
   auto		BIOS setup (default)
 

Documentation/sound/alsa/HD-Audio.txt

@@ -454,6 +454,8 @@ The generic parser supports the following hints:
 - need_dac_fix (bool): limits the DACs depending on the channel count
 - primary_hp (bool): probe headphone jacks as the primary outputs;
   default true
+- multi_io (bool): try probing multi-I/O config (e.g. shared
+  line-in/surround, mic/clfe jacks)
 - multi_cap_vol (bool): provide multiple capture volumes
 - inv_dmic_split (bool): provide split internal mic volume/switch for
   phase-inverted digital mics

MAINTAINERS

@@ -595,6 +595,7 @@ S:	Supported
 F:	sound/soc/codecs/adau*
 F:	sound/soc/codecs/adav*
 F:	sound/soc/codecs/ad1*
+F:	sound/soc/codecs/ad7*
 F:	sound/soc/codecs/ssm*
 F:	sound/soc/codecs/sigmadsp.*
 
@@ -7694,6 +7695,17 @@ F:	include/sound/
 F:	include/uapi/sound/
 F:	sound/
 
+SOUND - COMPRESSED AUDIO
+M:	Vinod Koul <vinod.koul@intel.com>
+L:	alsa-devel@alsa-project.org (moderated for non-subscribers)
+T:	git git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git
+S:	Supported
+F:	Documentation/sound/alsa/compress_offload.txt
+F:	include/sound/compress_driver.h
+F:	include/uapi/sound/compress_*
+F:	sound/core/compress_offload.c
+F:	sound/soc/soc-compress.c
+
 SOUND - SOC LAYER / DYNAMIC AUDIO POWER MANAGEMENT (ASoC)
 M:	Liam Girdwood <lgirdwood@gmail.com>
 M:	Mark Brown <broonie@kernel.org>
@@ -7701,6 +7713,7 @@ T:	git git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
 L:	alsa-devel@alsa-project.org (moderated for non-subscribers)
 W:	http://alsa-project.org/main/index.php/ASoC
 S:	Supported
+F:	Documentation/sound/alsa/soc/
 F:	sound/soc/
 F:	include/sound/soc*
 

arch/arm/boot/dts/exynos5250.dtsi

@@ -405,7 +405,7 @@
 	};
 
 	i2s0: i2s@03830000 {
-		compatible = "samsung,i2s-v5";
+		compatible = "samsung,s5pv210-i2s";
 		reg = <0x03830000 0x100>;
 		dmas = <&pdma0 10
 			&pdma0 9
@@ -415,16 +415,13 @@
 			<&clock_audss EXYNOS_I2S_BUS>,
 			<&clock_audss EXYNOS_SCLK_I2S>;
 		clock-names = "iis", "i2s_opclk0", "i2s_opclk1";
-		samsung,supports-6ch;
-		samsung,supports-rstclr;
-		samsung,supports-secdai;
 		samsung,idma-addr = <0x03000000>;
 		pinctrl-names = "default";
 		pinctrl-0 = <&i2s0_bus>;
 	};
 
 	i2s1: i2s@12D60000 {
-		compatible = "samsung,i2s-v5";
+		compatible = "samsung,s3c6410-i2s";
 		reg = <0x12D60000 0x100>;
 		dmas = <&pdma1 12
 			&pdma1 11>;
@@ -436,7 +433,7 @@
 	};
 
 	i2s2: i2s@12D70000 {
-		compatible = "samsung,i2s-v5";
+		compatible = "samsung,s3c6410-i2s";
 		reg = <0x12D70000 0x100>;
 		dmas = <&pdma0 12
 			&pdma0 11>;

arch/arm/mach-dove/common.c

@@ -108,8 +108,8 @@ static void __init dove_clk_init(void)
 	orion_clkdev_add(NULL, "sdhci-dove.1", sdio1);
 	orion_clkdev_add(NULL, "orion_nand", nand);
 	orion_clkdev_add(NULL, "cafe1000-ccic.0", camera);
-	orion_clkdev_add(NULL, "kirkwood-i2s.0", i2s0);
-	orion_clkdev_add(NULL, "kirkwood-i2s.1", i2s1);
+	orion_clkdev_add(NULL, "mvebu-audio.0", i2s0);
+	orion_clkdev_add(NULL, "mvebu-audio.1", i2s1);
 	orion_clkdev_add(NULL, "mv_crypto", crypto);
 	orion_clkdev_add(NULL, "dove-ac97", ac97);
 	orion_clkdev_add(NULL, "dove-pdma", pdma);

arch/arm/mach-kirkwood/common.c

@@ -264,7 +264,7 @@ void __init kirkwood_clk_init(void)
 	orion_clkdev_add(NULL, MV_XOR_NAME ".1", xor1);
 	orion_clkdev_add("0", "pcie", pex0);
 	orion_clkdev_add("1", "pcie", pex1);
-	orion_clkdev_add(NULL, "kirkwood-i2s", audio);
+	orion_clkdev_add(NULL, "mvebu-audio", audio);
 	orion_clkdev_add(NULL, MV64XXX_I2C_CTLR_NAME ".0", runit);
 	orion_clkdev_add(NULL, MV64XXX_I2C_CTLR_NAME ".1", runit);
 
@@ -560,7 +560,7 @@ void __init kirkwood_timer_init(void)
 /*****************************************************************************
  * Audio
  ****************************************************************************/
-static struct resource kirkwood_i2s_resources[] = {
+static struct resource kirkwood_audio_resources[] = {
 	[0] = {
 		.start  = AUDIO_PHYS_BASE,
 		.end    = AUDIO_PHYS_BASE + SZ_16K - 1,
@@ -573,29 +573,23 @@ static struct resource kirkwood_i2s_resources[] = {
 	},
 };
 
-static struct kirkwood_asoc_platform_data kirkwood_i2s_data = {
+static struct kirkwood_asoc_platform_data kirkwood_audio_data = {
 	.burst       = 128,
 };
 
-static struct platform_device kirkwood_i2s_device = {
-	.name		= "kirkwood-i2s",
+static struct platform_device kirkwood_audio_device = {
+	.name		= "mvebu-audio",
 	.id		= -1,
-	.num_resources	= ARRAY_SIZE(kirkwood_i2s_resources),
-	.resource	= kirkwood_i2s_resources,
+	.num_resources	= ARRAY_SIZE(kirkwood_audio_resources),
+	.resource	= kirkwood_audio_resources,
 	.dev		= {
-		.platform_data	= &kirkwood_i2s_data,
+		.platform_data	= &kirkwood_audio_data,
 	},
 };
 
-static struct platform_device kirkwood_pcm_device = {
-	.name		= "kirkwood-pcm-audio",
-	.id		= -1,
-};
-
 void __init kirkwood_audio_init(void)
 {
-	platform_device_register(&kirkwood_i2s_device);
-	platform_device_register(&kirkwood_pcm_device);
+	platform_device_register(&kirkwood_audio_device);
 }
 
 /*****************************************************************************

arch/arm/plat-pxa/ssp.c

@@ -30,6 +30,8 @@
 #include <linux/platform_device.h>
 #include <linux/spi/pxa2xx_spi.h>
 #include <linux/io.h>
+#include <linux/of.h>
+#include <linux/of_device.h>
 
 #include <asm/irq.h>
 #include <mach/hardware.h>
@@ -60,6 +62,30 @@ struct ssp_device *pxa_ssp_request(int port, const char *label)
 }
 EXPORT_SYMBOL(pxa_ssp_request);
 
+struct ssp_device *pxa_ssp_request_of(const struct device_node *of_node,
+				      const char *label)
+{
+	struct ssp_device *ssp = NULL;
+
+	mutex_lock(&ssp_lock);
+
+	list_for_each_entry(ssp, &ssp_list, node) {
+		if (ssp->of_node == of_node && ssp->use_count == 0) {
+			ssp->use_count++;
+			ssp->label = label;
+			break;
+		}
+	}
+
+	mutex_unlock(&ssp_lock);
+
+	if (&ssp->node == &ssp_list)
+		return NULL;
+
+	return ssp;
+}
+EXPORT_SYMBOL(pxa_ssp_request_of);
+
 void pxa_ssp_free(struct ssp_device *ssp)
 {
 	mutex_lock(&ssp_lock);
@@ -72,96 +98,126 @@ void pxa_ssp_free(struct ssp_device *ssp)
 }
 EXPORT_SYMBOL(pxa_ssp_free);
 
+#ifdef CONFIG_OF
+static const struct of_device_id pxa_ssp_of_ids[] = {
+	{ .compatible = "mrvl,pxa25x-ssp",	.data = (void *) PXA25x_SSP },
+	{ .compatible = "mvrl,pxa25x-nssp",	.data = (void *) PXA25x_NSSP },
+	{ .compatible = "mrvl,pxa27x-ssp",	.data = (void *) PXA27x_SSP },
+	{ .compatible = "mrvl,pxa3xx-ssp",	.data = (void *) PXA3xx_SSP },
+	{ .compatible = "mvrl,pxa168-ssp",	.data = (void *) PXA168_SSP },
+	{ .compatible = "mrvl,pxa910-ssp",	.data = (void *) PXA910_SSP },
+	{ .compatible = "mrvl,ce4100-ssp",	.data = (void *) CE4100_SSP },
+	{ .compatible = "mrvl,lpss-ssp",	.data = (void *) LPSS_SSP },
+	{ },
+};
+MODULE_DEVICE_TABLE(of, pxa_ssp_of_ids);
+#endif
+
 static int pxa_ssp_probe(struct platform_device *pdev)
 {
-	const struct platform_device_id *id = platform_get_device_id(pdev);
 	struct resource *res;
 	struct ssp_device *ssp;
-	int ret = 0;
+	struct device *dev = &pdev->dev;
 
-	ssp = kzalloc(sizeof(struct ssp_device), GFP_KERNEL);
-	if (ssp == NULL) {
-		dev_err(&pdev->dev, "failed to allocate memory");
+	ssp = devm_kzalloc(dev, sizeof(struct ssp_device), GFP_KERNEL);
+	if (ssp == NULL)
 		return -ENOMEM;
-	}
-	ssp->pdev = pdev;
 
-	ssp->clk = clk_get(&pdev->dev, NULL);
-	if (IS_ERR(ssp->clk)) {
-		ret = PTR_ERR(ssp->clk);
-		goto err_free;
-	}
+	ssp->pdev = pdev;
 
-	res = platform_get_resource(pdev, IORESOURCE_DMA, 0);
-	if (res == NULL) {
-		dev_err(&pdev->dev, "no SSP RX DRCMR defined\n");
-		ret = -ENODEV;
-		goto err_free_clk;
-	}
-	ssp->drcmr_rx = res->start;
+	ssp->clk = devm_clk_get(dev, NULL);
+	if (IS_ERR(ssp->clk))
+		return PTR_ERR(ssp->clk);
+
+	if (dev->of_node) {
+		struct of_phandle_args dma_spec;
+		struct device_node *np = dev->of_node;
+
+		/*
+		 * FIXME: we should allocate the DMA channel from this
+		 * context and pass the channel down to the ssp users.
+		 * For now, we lookup the rx and tx indices manually
+		 */
+
+		/* rx */
+		of_parse_phandle_with_args(np, "dmas", "#dma-cells",
+					   0, &dma_spec);
+		ssp->drcmr_rx = dma_spec.args[0];
+		of_node_put(dma_spec.np);
+
+		/* tx */
+		of_parse_phandle_with_args(np, "dmas", "#dma-cells",
+					   1, &dma_spec);
+		ssp->drcmr_tx = dma_spec.args[0];
+		of_node_put(dma_spec.np);
+	} else {
+		res = platform_get_resource(pdev, IORESOURCE_DMA, 0);
+		if (res == NULL) {
+			dev_err(dev, "no SSP RX DRCMR defined\n");
+			return -ENODEV;
+		}
+		ssp->drcmr_rx = res->start;
 
-	res = platform_get_resource(pdev, IORESOURCE_DMA, 1);
-	if (res == NULL) {
-		dev_err(&pdev->dev, "no SSP TX DRCMR defined\n");
-		ret = -ENODEV;
-		goto err_free_clk;
+		res = platform_get_resource(pdev, IORESOURCE_DMA, 1);
+		if (res == NULL) {
+			dev_err(dev, "no SSP TX DRCMR defined\n");
+			return -ENODEV;
+		}
+		ssp->drcmr_tx = res->start;
 	}
-	ssp->drcmr_tx = res->start;
 
 	res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
 	if (res == NULL) {
-		dev_err(&pdev->dev, "no memory resource defined\n");
-		ret = -ENODEV;
-		goto err_free_clk;
+		dev_err(dev, "no memory resource defined\n");
+		return -ENODEV;
 	}
 
-	res = request_mem_region(res->start, resource_size(res),
-			pdev->name);
+	res = devm_request_mem_region(dev, res->start, resource_size(res),
+				      pdev->name);
 	if (res == NULL) {
-		dev_err(&pdev->dev, "failed to request memory resource\n");
-		ret = -EBUSY;
-		goto err_free_clk;
+		dev_err(dev, "failed to request memory resource\n");
+		return -EBUSY;
 	}
 
 	ssp->phys_base = res->start;
 
-	ssp->mmio_base = ioremap(res->start, resource_size(res));
+	ssp->mmio_base = devm_ioremap(dev, res->start, resource_size(res));
 	if (ssp->mmio_base == NULL) {
-		dev_err(&pdev->dev, "failed to ioremap() registers\n");
-		ret = -ENODEV;
-		goto err_free_mem;
+		dev_err(dev, "failed to ioremap() registers\n");
+		return -ENODEV;
 	}
 
 	ssp->irq = platform_get_irq(pdev, 0);
 	if (ssp->irq < 0) {
-		dev_err(&pdev->dev, "no IRQ resource defined\n");
-		ret = -ENODEV;
-		goto err_free_io;
+		dev_err(dev, "no IRQ resource defined\n");
+		return -ENODEV;
+	}
+
+	if (dev->of_node) {
+		const struct of_device_id *id =
+			of_match_device(of_match_ptr(pxa_ssp_of_ids), dev);
+		ssp->type = (int) id->data;
+	} else {
+		const struct platform_device_id *id =
+			platform_get_device_id(pdev);
+		ssp->type = (int) id->driver_data;
+
+		/* PXA2xx/3xx SSP ports starts from 1 and the internal pdev->id
+		 * starts from 0, do a translation here
+		 */
+		ssp->port_id = pdev->id + 1;
 	}
 
-	/* PXA2xx/3xx SSP ports starts from 1 and the internal pdev->id
-	 * starts from 0, do a translation here
-	 */
-	ssp->port_id = pdev->id + 1;
 	ssp->use_count = 0;
-	ssp->type = (int)id->driver_data;
+	ssp->of_node = dev->of_node;
 
 	mutex_lock(&ssp_lock);
 	list_add(&ssp->node, &ssp_list);
 	mutex_unlock(&ssp_lock);
 
 	platform_set_drvdata(pdev, ssp);
-	return 0;
 
-err_free_io:
-	iounmap(ssp->mmio_base);
-err_free_mem:
-	release_mem_region(res->start, resource_size(res));
-err_free_clk:
-	clk_put(ssp->clk);
-err_free:
-	kfree(ssp);
-	return ret;
+	return 0;
 }
 
 static int pxa_ssp_remove(struct platform_device *pdev)
@@ -201,8 +257,9 @@ static struct platform_driver pxa_ssp_driver = {
 	.probe		= pxa_ssp_probe,
 	.remove		= pxa_ssp_remove,
 	.driver		= {
-		.owner	= THIS_MODULE,
-		.name	= "pxa2xx-ssp",
+		.owner		= THIS_MODULE,
+		.name		= "pxa2xx-ssp",
+		.of_match_table	= of_match_ptr(pxa_ssp_of_ids),
 	},
 	.id_table	= ssp_id_table,
 };

arch/arm/plat-samsung/s3c-dma-ops.c

@@ -82,7 +82,8 @@ static int s3c_dma_config(unsigned ch, struct samsung_dma_config *param)
 static int s3c_dma_prepare(unsigned ch, struct samsung_dma_prep *param)
 {
 	struct cb_data *data;
-	int len = (param->cap == DMA_CYCLIC) ? param->period : param->len;
+	dma_addr_t pos = param->buf;
+	dma_addr_t end = param->buf + param->len;
 
 	list_for_each_entry(data, &dma_list, node)
 		if (data->ch == ch)
@@ -94,7 +95,15 @@ static int s3c_dma_prepare(unsigned ch, struct samsung_dma_prep *param)
 		data->fp_param = param->fp_param;
 	}
 
-	s3c2410_dma_enqueue(ch, (void *)data, param->buf, len);
+	if (param->cap != DMA_CYCLIC) {
+		s3c2410_dma_enqueue(ch, (void *)data, param->buf, param->len);
+		return 0;
+	}
+
+	while (pos < end) {
+		s3c2410_dma_enqueue(ch, (void *)data, pos, param->period);
+		pos += param->period;
+	}
 
 	return 0;
 }

include/dt-bindings/sound/fsl-imx-audmux.h

@@ -0,0 +1,56 @@
+#ifndef __DT_FSL_IMX_AUDMUX_H
+#define __DT_FSL_IMX_AUDMUX_H
+
+#define MX27_AUDMUX_HPCR1_SSI0		0
+#define MX27_AUDMUX_HPCR2_SSI1		1
+#define MX27_AUDMUX_HPCR3_SSI_PINS_4	2
+#define MX27_AUDMUX_PPCR1_SSI_PINS_1	3
+#define MX27_AUDMUX_PPCR2_SSI_PINS_2	4
+#define MX27_AUDMUX_PPCR3_SSI_PINS_3	5
+
+#define MX31_AUDMUX_PORT1_SSI0		0
+#define MX31_AUDMUX_PORT2_SSI1		1
+#define MX31_AUDMUX_PORT3_SSI_PINS_3	2
+#define MX31_AUDMUX_PORT4_SSI_PINS_4	3
+#define MX31_AUDMUX_PORT5_SSI_PINS_5	4
+#define MX31_AUDMUX_PORT6_SSI_PINS_6	5
+#define MX31_AUDMUX_PORT7_SSI_PINS_7	6
+
+#define MX51_AUDMUX_PORT1_SSI0		0
+#define MX51_AUDMUX_PORT2_SSI1		1
+#define MX51_AUDMUX_PORT3		2
+#define MX51_AUDMUX_PORT4		3
+#define MX51_AUDMUX_PORT5		4
+#define MX51_AUDMUX_PORT6		5
+#define MX51_AUDMUX_PORT7		6
+
+/* Register definitions for the i.MX21/27 Digital Audio Multiplexer */
+#define IMX_AUDMUX_V1_PCR_INMMASK(x)	((x) & 0xff)
+#define IMX_AUDMUX_V1_PCR_INMEN		(1 << 8)
+#define IMX_AUDMUX_V1_PCR_TXRXEN	(1 << 10)
+#define IMX_AUDMUX_V1_PCR_SYN		(1 << 12)
+#define IMX_AUDMUX_V1_PCR_RXDSEL(x)	(((x) & 0x7) << 13)
+#define IMX_AUDMUX_V1_PCR_RFCSEL(x)	(((x) & 0xf) << 20)
+#define IMX_AUDMUX_V1_PCR_RCLKDIR	(1 << 24)
+#define IMX_AUDMUX_V1_PCR_RFSDIR	(1 << 25)
+#define IMX_AUDMUX_V1_PCR_TFCSEL(x)	(((x) & 0xf) << 26)
+#define IMX_AUDMUX_V1_PCR_TCLKDIR	(1 << 30)
+#define IMX_AUDMUX_V1_PCR_TFSDIR	(1 << 31)
+
+/* Register definitions for the i.MX25/31/35/51 Digital Audio Multiplexer */
+#define IMX_AUDMUX_V2_PTCR_TFSDIR	(1 << 31)
+#define IMX_AUDMUX_V2_PTCR_TFSEL(x)	(((x) & 0xf) << 27)
+#define IMX_AUDMUX_V2_PTCR_TCLKDIR	(1 << 26)
+#define IMX_AUDMUX_V2_PTCR_TCSEL(x)	(((x) & 0xf) << 22)
+#define IMX_AUDMUX_V2_PTCR_RFSDIR	(1 << 21)
+#define IMX_AUDMUX_V2_PTCR_RFSEL(x)	(((x) & 0xf) << 17)
+#define IMX_AUDMUX_V2_PTCR_RCLKDIR	(1 << 16)
+#define IMX_AUDMUX_V2_PTCR_RCSEL(x)	(((x) & 0xf) << 12)
+#define IMX_AUDMUX_V2_PTCR_SYN		(1 << 11)
+
+#define IMX_AUDMUX_V2_PDCR_RXDSEL(x)	(((x) & 0x7) << 13)
+#define IMX_AUDMUX_V2_PDCR_TXRXEN	(1 << 12)
+#define IMX_AUDMUX_V2_PDCR_MODE(x)	(((x) & 0x3) << 8)
+#define IMX_AUDMUX_V2_PDCR_INMMASK(x)	((x) & 0xff)
+
+#endif /* __DT_FSL_IMX_AUDMUX_H */

include/linux/atmel-ssc.h

@@ -11,7 +11,7 @@ struct atmel_ssc_platform_data {
 
 struct ssc_device {
 	struct list_head	list;
-	resource_size_t		phybase;
+	dma_addr_t		phybase;
 	void __iomem		*regs;
 	struct platform_device	*pdev;
 	struct atmel_ssc_platform_data *pdata;

include/linux/mfd/arizona/gpio.h

@@ -0,0 +1,96 @@
+/*
+ * GPIO configuration for Arizona devices
+ *
+ * Copyright 2013 Wolfson Microelectronics. PLC.
+ *
+ * Author: Charles Keepax <ckeepax@opensource.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.
+ */
+
+#ifndef _ARIZONA_GPIO_H
+#define _ARIZONA_GPIO_H
+
+#define ARIZONA_GP_FN_TXLRCLK                    0x00
+#define ARIZONA_GP_FN_GPIO                       0x01
+#define ARIZONA_GP_FN_IRQ1                       0x02
+#define ARIZONA_GP_FN_IRQ2                       0x03
+#define ARIZONA_GP_FN_OPCLK                      0x04
+#define ARIZONA_GP_FN_FLL1_OUT                   0x05
+#define ARIZONA_GP_FN_FLL2_OUT                   0x06
+#define ARIZONA_GP_FN_PWM1                       0x08
+#define ARIZONA_GP_FN_PWM2                       0x09
+#define ARIZONA_GP_FN_SYSCLK_UNDERCLOCKED        0x0A
+#define ARIZONA_GP_FN_ASYNCCLK_UNDERCLOCKED      0x0B
+#define ARIZONA_GP_FN_FLL1_LOCK                  0x0C
+#define ARIZONA_GP_FN_FLL2_LOCK                  0x0D
+#define ARIZONA_GP_FN_FLL1_CLOCK_OK              0x0F
+#define ARIZONA_GP_FN_FLL2_CLOCK_OK              0x10
+#define ARIZONA_GP_FN_HEADPHONE_DET              0x12
+#define ARIZONA_GP_FN_MIC_DET                    0x13
+#define ARIZONA_GP_FN_WSEQ_STATUS                0x15
+#define ARIZONA_GP_FN_CIF_ADDRESS_ERROR          0x16
+#define ARIZONA_GP_FN_ASRC1_LOCK                 0x1A
+#define ARIZONA_GP_FN_ASRC2_LOCK                 0x1B
+#define ARIZONA_GP_FN_ASRC_CONFIG_ERROR          0x1C
+#define ARIZONA_GP_FN_DRC1_SIGNAL_DETECT         0x1D
+#define ARIZONA_GP_FN_DRC1_ANTICLIP              0x1E
+#define ARIZONA_GP_FN_DRC1_DECAY                 0x1F
+#define ARIZONA_GP_FN_DRC1_NOISE                 0x20
+#define ARIZONA_GP_FN_DRC1_QUICK_RELEASE         0x21
+#define ARIZONA_GP_FN_DRC2_SIGNAL_DETECT         0x22
+#define ARIZONA_GP_FN_DRC2_ANTICLIP              0x23
+#define ARIZONA_GP_FN_DRC2_DECAY                 0x24
+#define ARIZONA_GP_FN_DRC2_NOISE                 0x25
+#define ARIZONA_GP_FN_DRC2_QUICK_RELEASE         0x26
+#define ARIZONA_GP_FN_MIXER_DROPPED_SAMPLE       0x27
+#define ARIZONA_GP_FN_AIF1_CONFIG_ERROR          0x28
+#define ARIZONA_GP_FN_AIF2_CONFIG_ERROR          0x29
+#define ARIZONA_GP_FN_AIF3_CONFIG_ERROR          0x2A
+#define ARIZONA_GP_FN_SPK_TEMP_SHUTDOWN          0x2B
+#define ARIZONA_GP_FN_SPK_TEMP_WARNING           0x2C
+#define ARIZONA_GP_FN_UNDERCLOCKED               0x2D
+#define ARIZONA_GP_FN_OVERCLOCKED                0x2E
+#define ARIZONA_GP_FN_DSP_IRQ1                   0x35
+#define ARIZONA_GP_FN_DSP_IRQ2                   0x36
+#define ARIZONA_GP_FN_ASYNC_OPCLK                0x3D
+#define ARIZONA_GP_FN_BOOT_DONE                  0x44
+#define ARIZONA_GP_FN_DSP1_RAM_READY             0x45
+#define ARIZONA_GP_FN_SYSCLK_ENA_STATUS          0x4B
+#define ARIZONA_GP_FN_ASYNCCLK_ENA_STATUS        0x4C
+
+#define ARIZONA_GPN_DIR                          0x8000  /* GPN_DIR */
+#define ARIZONA_GPN_DIR_MASK                     0x8000  /* GPN_DIR */
+#define ARIZONA_GPN_DIR_SHIFT                        15  /* GPN_DIR */
+#define ARIZONA_GPN_DIR_WIDTH                         1  /* GPN_DIR */
+#define ARIZONA_GPN_PU                           0x4000  /* GPN_PU */
+#define ARIZONA_GPN_PU_MASK                      0x4000  /* GPN_PU */
+#define ARIZONA_GPN_PU_SHIFT                         14  /* GPN_PU */
+#define ARIZONA_GPN_PU_WIDTH                          1  /* GPN_PU */
+#define ARIZONA_GPN_PD                           0x2000  /* GPN_PD */
+#define ARIZONA_GPN_PD_MASK                      0x2000  /* GPN_PD */
+#define ARIZONA_GPN_PD_SHIFT                         13  /* GPN_PD */
+#define ARIZONA_GPN_PD_WIDTH                          1  /* GPN_PD */
+#define ARIZONA_GPN_LVL                          0x0800  /* GPN_LVL */
+#define ARIZONA_GPN_LVL_MASK                     0x0800  /* GPN_LVL */
+#define ARIZONA_GPN_LVL_SHIFT                        11  /* GPN_LVL */
+#define ARIZONA_GPN_LVL_WIDTH                         1  /* GPN_LVL */
+#define ARIZONA_GPN_POL                          0x0400  /* GPN_POL */
+#define ARIZONA_GPN_POL_MASK                     0x0400  /* GPN_POL */
+#define ARIZONA_GPN_POL_SHIFT                        10  /* GPN_POL */
+#define ARIZONA_GPN_POL_WIDTH                         1  /* GPN_POL */
+#define ARIZONA_GPN_OP_CFG                       0x0200  /* GPN_OP_CFG */
+#define ARIZONA_GPN_OP_CFG_MASK                  0x0200  /* GPN_OP_CFG */
+#define ARIZONA_GPN_OP_CFG_SHIFT                      9  /* GPN_OP_CFG */
+#define ARIZONA_GPN_OP_CFG_WIDTH                      1  /* GPN_OP_CFG */
+#define ARIZONA_GPN_DB                           0x0100  /* GPN_DB */
+#define ARIZONA_GPN_DB_MASK                      0x0100  /* GPN_DB */
+#define ARIZONA_GPN_DB_SHIFT                          8  /* GPN_DB */
+#define ARIZONA_GPN_DB_WIDTH                          1  /* GPN_DB */
+#define ARIZONA_GPN_FN_MASK                      0x007F  /* GPN_DB */
+#define ARIZONA_GPN_FN_SHIFT                          0  /* GPN_DB */
+#define ARIZONA_GPN_FN_WIDTH                          7  /* GPN_DB */
+
+#endif

include/linux/platform_data/asoc-s3c.h

@@ -36,6 +36,7 @@ struct samsung_i2s {
  */
 #define QUIRK_NO_MUXPSR		(1 << 2)
 #define QUIRK_NEED_RSTCLR	(1 << 3)
+#define QUIRK_SUPPORTS_TDM	(1 << 4)
 	/* Quirks of the I2S controller */
 	u32 quirks;
 	dma_addr_t idma_addr;

include/linux/platform_data/omap-abe-twl6040.h

@@ -1,49 +0,0 @@
-/**
- * omap-abe-twl6040.h - ASoC machine driver OMAP4+ devices, header.
- *
- * Copyright (C) 2011 Texas Instruments Incorporated - http://www.ti.com
- * All rights reserved.
- *
- * Author: Peter Ujfalusi <peter.ujfalusi@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
- */
-
-#ifndef _OMAP_ABE_TWL6040_H_
-#define _OMAP_ABE_TWL6040_H_
-
-/* To select if only one channel is connected in a stereo port */
-#define ABE_TWL6040_LEFT	(1 << 0)
-#define ABE_TWL6040_RIGHT	(1 << 1)
-
-struct omap_abe_twl6040_data {
-	char *card_name;
-	/* Feature flags for connected audio pins */
-	u8	has_hs;
-	u8	has_hf;
-	bool	has_ep;
-	u8	has_aux;
-	u8	has_vibra;
-	bool	has_dmic;
-	bool	has_hsmic;
-	bool	has_mainmic;
-	bool	has_submic;
-	u8	has_afm;
-	/* Other features */
-	bool	jack_detection;	/* board can detect jack events */
-	int	mclk_freq;	/* MCLK frequency speed for twl6040 */
-};
-
-#endif /* _OMAP_ABE_TWL6040_H_ */

include/linux/pxa2xx_ssp.h

@@ -21,6 +21,8 @@
 
 #include <linux/list.h>
 #include <linux/io.h>
+#include <linux/of.h>
+
 
 /*
  * SSP Serial Port Registers
@@ -190,6 +192,8 @@ struct ssp_device {
 	int		irq;
 	int		drcmr_rx;
 	int		drcmr_tx;
+
+	struct device_node	*of_node;
 };
 
 /**
@@ -218,11 +222,18 @@ static inline u32 pxa_ssp_read_reg(struct ssp_device *dev, u32 reg)
 #ifdef CONFIG_ARCH_PXA
 struct ssp_device *pxa_ssp_request(int port, const char *label);
 void pxa_ssp_free(struct ssp_device *);
+struct ssp_device *pxa_ssp_request_of(const struct device_node *of_node,
+				      const char *label);
 #else
 static inline struct ssp_device *pxa_ssp_request(int port, const char *label)
 {
 	return NULL;
 }
+static inline struct ssp_device *pxa_ssp_request_of(const struct device_node *n,
+						    const char *name)
+{
+	return NULL;
+}
 static inline void pxa_ssp_free(struct ssp_device *ssp) {}
 #endif
 

include/sound/core.h

@@ -27,6 +27,7 @@
 #include <linux/rwsem.h>		/* struct rw_semaphore */
 #include <linux/pm.h>			/* pm_message_t */
 #include <linux/stringify.h>
+#include <linux/printk.h>
 
 /* number of supported soundcards */
 #ifdef CONFIG_SND_DYNAMIC_MINORS
@@ -375,6 +376,11 @@ void __snd_printk(unsigned int level, const char *file, int line,
  */
 #define snd_BUG()		WARN(1, "BUG?\n")
 
+/**
+ * Suppress high rates of output when CONFIG_SND_DEBUG is enabled.
+ */
+#define snd_printd_ratelimit() printk_ratelimit()
+
 /**
  * snd_BUG_ON - debugging check macro
  * @cond: condition to evaluate
@@ -398,6 +404,8 @@ static inline void _snd_printd(int level, const char *format, ...) {}
 	unlikely(__ret_warn_on); \
 })
 
+static inline bool snd_printd_ratelimit(void) { return false; }
+
 #endif /* CONFIG_SND_DEBUG */
 
 #ifdef CONFIG_SND_DEBUG_VERBOSE

include/sound/pxa2xx-lib.h

@@ -6,13 +6,6 @@
 
 /* PCM */
 
-struct pxa2xx_pcm_dma_params {
-	char *name;			/* stream identifier */
-	u32 dcmd;			/* DMA descriptor dcmd field */
-	volatile u32 *drcmr;		/* the DMA request channel to use */
-	u32 dev_addr;			/* device physical address for DMA */
-};
-
 extern int __pxa2xx_pcm_hw_params(struct snd_pcm_substream *substream,
 				struct snd_pcm_hw_params *params);
 extern int __pxa2xx_pcm_hw_free(struct snd_pcm_substream *substream);

include/sound/rcar_snd.h

@@ -0,0 +1,84 @@
+/*
+ * Renesas R-Car SRU/SCU/SSIU/SSI support
+ *
+ * Copyright (C) 2013 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 RCAR_SND_H
+#define RCAR_SND_H
+
+#include <linux/sh_clk.h>
+
+#define RSND_GEN1_SRU	0
+#define RSND_GEN1_ADG	1
+#define RSND_GEN1_SSI	2
+
+#define RSND_GEN2_SRU	0
+#define RSND_GEN2_ADG	1
+#define RSND_GEN2_SSIU	2
+#define RSND_GEN2_SSI	3
+
+#define RSND_BASE_MAX	4
+
+/*
+ * flags
+ *
+ * 0xAB000000
+ *
+ * A : clock sharing settings
+ * B : SSI direction
+ */
+#define RSND_SSI_CLK_PIN_SHARE		(1 << 31)
+#define RSND_SSI_CLK_FROM_ADG		(1 << 30) /* clock parent is master */
+#define RSND_SSI_SYNC			(1 << 29) /* SSI34_sync etc */
+#define RSND_SSI_DEPENDENT		(1 << 28) /* SSI needs SRU/SCU */
+
+#define RSND_SSI_PLAY			(1 << 24)
+
+#define RSND_SSI_SET(_dai_id, _dma_id, _pio_irq, _flags)	\
+{ .dai_id = _dai_id, .dma_id = _dma_id, .pio_irq = _pio_irq, .flags = _flags }
+#define RSND_SSI_UNUSED \
+{ .dai_id = -1, .dma_id = -1, .pio_irq = -1, .flags = 0 }
+
+struct rsnd_ssi_platform_info {
+	int dai_id;
+	int dma_id;
+	int pio_irq;
+	u32 flags;
+};
+
+/*
+ * flags
+ */
+#define RSND_SCU_USB_HPBIF		(1 << 31) /* it needs RSND_SSI_DEPENDENT */
+
+struct rsnd_scu_platform_info {
+	u32 flags;
+};
+
+/*
+ * flags
+ *
+ * 0x0000000A
+ *
+ * A : generation
+ */
+#define RSND_GEN1	(1 << 0) /* fixme */
+#define RSND_GEN2	(2 << 0) /* fixme */
+
+struct rcar_snd_info {
+	u32 flags;
+	struct rsnd_ssi_platform_info *ssi_info;
+	int ssi_info_nr;
+	struct rsnd_scu_platform_info *scu_info;
+	int scu_info_nr;
+	int (*start)(int id);
+	int (*stop)(int id);
+};
+
+#endif

include/sound/soc-dapm.h

@@ -70,121 +70,144 @@ struct device;
 	.num_kcontrols = 0, .reg = SND_SOC_NOPM, .event = wevent, \
 	.event_flags = SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD}
 
+#define SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert) \
+	.reg = wreg, .mask = 1, .shift = wshift, \
+	.on_val = winvert ? 0 : 1, .off_val = winvert ? 1 : 0
+
 /* path domain */
 #define SND_SOC_DAPM_PGA(wname, wreg, wshift, winvert,\
 	 wcontrols, wncontrols) \
-{	.id = snd_soc_dapm_pga, .name = wname, .reg = wreg, .shift = wshift, \
-	.invert = winvert, .kcontrol_news = wcontrols, .num_kcontrols = wncontrols}
+{	.id = snd_soc_dapm_pga, .name = wname, \
+	SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \
+	.kcontrol_news = wcontrols, .num_kcontrols = wncontrols}
 #define SND_SOC_DAPM_OUT_DRV(wname, wreg, wshift, winvert,\
 	 wcontrols, wncontrols) \
-{	.id = snd_soc_dapm_out_drv, .name = wname, .reg = wreg, .shift = wshift, \
-	.invert = winvert, .kcontrol_news = wcontrols, .num_kcontrols = wncontrols}
+{	.id = snd_soc_dapm_out_drv, .name = wname, \
+	SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \
+	.kcontrol_news = wcontrols, .num_kcontrols = wncontrols}
 #define SND_SOC_DAPM_MIXER(wname, wreg, wshift, winvert, \
 	 wcontrols, wncontrols)\
-{	.id = snd_soc_dapm_mixer, .name = wname, .reg = wreg, .shift = wshift, \
-	.invert = winvert, .kcontrol_news = wcontrols, .num_kcontrols = wncontrols}
+{	.id = snd_soc_dapm_mixer, .name = wname, \
+	SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \
+	.kcontrol_news = wcontrols, .num_kcontrols = wncontrols}
 #define SND_SOC_DAPM_MIXER_NAMED_CTL(wname, wreg, wshift, winvert, \
 	 wcontrols, wncontrols)\
-{       .id = snd_soc_dapm_mixer_named_ctl, .name = wname, .reg = wreg, \
-	.shift = wshift, .invert = winvert, .kcontrol_news = wcontrols, \
-	.num_kcontrols = wncontrols}
+{       .id = snd_soc_dapm_mixer_named_ctl, .name = wname, \
+	SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \
+	.kcontrol_news = wcontrols, .num_kcontrols = wncontrols}
 #define SND_SOC_DAPM_MICBIAS(wname, wreg, wshift, winvert) \
-{	.id = snd_soc_dapm_micbias, .name = wname, .reg = wreg, .shift = wshift, \
-	.invert = winvert, .kcontrol_news = NULL, .num_kcontrols = 0}
+{	.id = snd_soc_dapm_micbias, .name = wname, \
+	SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \
+	.kcontrol_news = NULL, .num_kcontrols = 0}
 #define SND_SOC_DAPM_SWITCH(wname, wreg, wshift, winvert, wcontrols) \
-{	.id = snd_soc_dapm_switch, .name = wname, .reg = wreg, .shift = wshift, \
-	.invert = winvert, .kcontrol_news = wcontrols, .num_kcontrols = 1}
+{	.id = snd_soc_dapm_switch, .name = wname, \
+	SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \
+	.kcontrol_news = wcontrols, .num_kcontrols = 1}
 #define SND_SOC_DAPM_MUX(wname, wreg, wshift, winvert, wcontrols) \
-{	.id = snd_soc_dapm_mux, .name = wname, .reg = wreg, .shift = wshift, \
-	.invert = winvert, .kcontrol_news = wcontrols, .num_kcontrols = 1}
+{	.id = snd_soc_dapm_mux, .name = wname, .reg = wreg, \
+	.kcontrol_news = wcontrols, .num_kcontrols = 1}
 #define SND_SOC_DAPM_VIRT_MUX(wname, wreg, wshift, winvert, wcontrols) \
-{	.id = snd_soc_dapm_virt_mux, .name = wname, .reg = wreg, .shift = wshift, \
-	.invert = winvert, .kcontrol_news = wcontrols, .num_kcontrols = 1}
+{	.id = snd_soc_dapm_virt_mux, .name = wname, \
+	SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \
+	.kcontrol_news = wcontrols, .num_kcontrols = 1}
 #define SND_SOC_DAPM_VALUE_MUX(wname, wreg, wshift, winvert, wcontrols) \
-{	.id = snd_soc_dapm_value_mux, .name = wname, .reg = wreg, \
-	.shift = wshift, .invert = winvert, .kcontrol_news = wcontrols, \
-	.num_kcontrols = 1}
+{	.id = snd_soc_dapm_value_mux, .name = wname, \
+	SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \
+	.kcontrol_news = wcontrols, .num_kcontrols = 1}
 
 /* Simplified versions of above macros, assuming wncontrols = ARRAY_SIZE(wcontrols) */
 #define SOC_PGA_ARRAY(wname, wreg, wshift, winvert,\
 	 wcontrols) \
-{	.id = snd_soc_dapm_pga, .name = wname, .reg = wreg, .shift = wshift, \
-	.invert = winvert, .kcontrol_news = wcontrols, .num_kcontrols = ARRAY_SIZE(wcontrols)}
+{	.id = snd_soc_dapm_pga, .name = wname, \
+	SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \
+	.kcontrol_news = wcontrols, .num_kcontrols = ARRAY_SIZE(wcontrols)}
 #define SOC_MIXER_ARRAY(wname, wreg, wshift, winvert, \
 	 wcontrols)\
-{	.id = snd_soc_dapm_mixer, .name = wname, .reg = wreg, .shift = wshift, \
-	.invert = winvert, .kcontrol_news = wcontrols, .num_kcontrols = ARRAY_SIZE(wcontrols)}
+{	.id = snd_soc_dapm_mixer, .name = wname, \
+	SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \
+	.kcontrol_news = wcontrols, .num_kcontrols = ARRAY_SIZE(wcontrols)}
 #define SOC_MIXER_NAMED_CTL_ARRAY(wname, wreg, wshift, winvert, \
 	 wcontrols)\
-{       .id = snd_soc_dapm_mixer_named_ctl, .name = wname, .reg = wreg, \
-	.shift = wshift, .invert = winvert, .kcontrol_news = wcontrols, \
-	.num_kcontrols = ARRAY_SIZE(wcontrols)}
+{       .id = snd_soc_dapm_mixer_named_ctl, .name = wname, \
+	SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \
+	.kcontrol_news = wcontrols, .num_kcontrols = ARRAY_SIZE(wcontrols)}
 
 /* path domain with event - event handler must return 0 for success */
 #define SND_SOC_DAPM_PGA_E(wname, wreg, wshift, winvert, wcontrols, \
 	wncontrols, wevent, wflags) \
-{	.id = snd_soc_dapm_pga, .name = wname, .reg = wreg, .shift = wshift, \
-	.invert = winvert, .kcontrol_news = wcontrols, .num_kcontrols = wncontrols, \
+{	.id = snd_soc_dapm_pga, .name = wname, \
+	SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \
+	.kcontrol_news = wcontrols, .num_kcontrols = wncontrols, \
 	.event = wevent, .event_flags = wflags}
 #define SND_SOC_DAPM_OUT_DRV_E(wname, wreg, wshift, winvert, wcontrols, \
 	wncontrols, wevent, wflags) \
-{	.id = snd_soc_dapm_out_drv, .name = wname, .reg = wreg, .shift = wshift, \
-	.invert = winvert, .kcontrol_news = wcontrols, .num_kcontrols = wncontrols, \
+{	.id = snd_soc_dapm_out_drv, .name = wname, \
+	SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \
+	.kcontrol_news = wcontrols, .num_kcontrols = wncontrols, \
 	.event = wevent, .event_flags = wflags}
 #define SND_SOC_DAPM_MIXER_E(wname, wreg, wshift, winvert, wcontrols, \
 	wncontrols, wevent, wflags) \
-{	.id = snd_soc_dapm_mixer, .name = wname, .reg = wreg, .shift = wshift, \
-	.invert = winvert, .kcontrol_news = wcontrols, .num_kcontrols = wncontrols, \
+{	.id = snd_soc_dapm_mixer, .name = wname, \
+	SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \
+	.kcontrol_news = wcontrols, .num_kcontrols = wncontrols, \
 	.event = wevent, .event_flags = wflags}
 #define SND_SOC_DAPM_MIXER_NAMED_CTL_E(wname, wreg, wshift, winvert, \
 	wcontrols, wncontrols, wevent, wflags) \
-{       .id = snd_soc_dapm_mixer, .name = wname, .reg = wreg, .shift = wshift, \
-	.invert = winvert, .kcontrol_news = wcontrols, \
+{       .id = snd_soc_dapm_mixer, .name = wname, \
+	SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \
+	.kcontrol_news = wcontrols, \
 	.num_kcontrols = wncontrols, .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, \
-	.invert = winvert, .kcontrol_news = wcontrols, .num_kcontrols = 1, \
+{	.id = snd_soc_dapm_switch, .name = wname, \
+	SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \
+	.kcontrol_news = wcontrols, .num_kcontrols = 1, \
 	.event = wevent, .event_flags = wflags}
 #define SND_SOC_DAPM_MUX_E(wname, wreg, wshift, winvert, wcontrols, \
 	wevent, wflags) \
-{	.id = snd_soc_dapm_mux, .name = wname, .reg = wreg, .shift = wshift, \
-	.invert = winvert, .kcontrol_news = wcontrols, .num_kcontrols = 1, \
+{	.id = snd_soc_dapm_mux, .name = wname, \
+	SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \
+	.kcontrol_news = wcontrols, .num_kcontrols = 1, \
 	.event = wevent, .event_flags = wflags}
 #define SND_SOC_DAPM_VIRT_MUX_E(wname, wreg, wshift, winvert, wcontrols, \
 	wevent, wflags) \
-{	.id = snd_soc_dapm_virt_mux, .name = wname, .reg = wreg, .shift = wshift, \
-	.invert = winvert, .kcontrol_news = wcontrols, .num_kcontrols = 1, \
+{	.id = snd_soc_dapm_virt_mux, .name = wname, \
+	SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \
+	.kcontrol_news = wcontrols, .num_kcontrols = 1, \
 	.event = wevent, .event_flags = wflags}
 
 /* additional sequencing control within an event type */
 #define SND_SOC_DAPM_PGA_S(wname, wsubseq, wreg, wshift, winvert, \
 	wevent, wflags) \
-{	.id = snd_soc_dapm_pga, .name = wname, .reg = wreg, .shift = wshift, \
-	.invert = winvert, .event = wevent, .event_flags = wflags, \
+{	.id = snd_soc_dapm_pga, .name = wname, \
+	SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \
+	.event = wevent, .event_flags = wflags, \
 	.subseq = wsubseq}
 #define SND_SOC_DAPM_SUPPLY_S(wname, wsubseq, wreg, wshift, winvert, wevent, \
 	wflags)	\
-{	.id = snd_soc_dapm_supply, .name = wname, .reg = wreg,	\
-	.shift = wshift, .invert = winvert, .event = wevent, \
-	.event_flags = wflags, .subseq = wsubseq}
+{	.id = snd_soc_dapm_supply, .name = wname, \
+	SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \
+	.event = wevent, .event_flags = wflags, .subseq = wsubseq}
 
 /* Simplified versions of above macros, assuming wncontrols = ARRAY_SIZE(wcontrols) */
 #define SOC_PGA_E_ARRAY(wname, wreg, wshift, winvert, wcontrols, \
 	wevent, wflags) \
-{	.id = snd_soc_dapm_pga, .name = wname, .reg = wreg, .shift = wshift, \
-	.invert = winvert, .kcontrol_news = wcontrols, .num_kcontrols = ARRAY_SIZE(wcontrols), \
+{	.id = snd_soc_dapm_pga, .name = wname, \
+	SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \
+	.kcontrol_news = wcontrols, .num_kcontrols = ARRAY_SIZE(wcontrols), \
 	.event = wevent, .event_flags = wflags}
 #define SOC_MIXER_E_ARRAY(wname, wreg, wshift, winvert, wcontrols, \
 	wevent, wflags) \
-{	.id = snd_soc_dapm_mixer, .name = wname, .reg = wreg, .shift = wshift, \
-	.invert = winvert, .kcontrol_news = wcontrols, .num_kcontrols = ARRAY_SIZE(wcontrols), \
+{	.id = snd_soc_dapm_mixer, .name = wname, \
+	SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \
+	.kcontrol_news = wcontrols, .num_kcontrols = ARRAY_SIZE(wcontrols), \
 	.event = wevent, .event_flags = wflags}
 #define SOC_MIXER_NAMED_CTL_E_ARRAY(wname, wreg, wshift, winvert, \
 	wcontrols, wevent, wflags) \
-{       .id = snd_soc_dapm_mixer, .name = wname, .reg = wreg, .shift = wshift, \
-	.invert = winvert, .kcontrol_news = wcontrols, \
-	.num_kcontrols = ARRAY_SIZE(wcontrols), .event = wevent, .event_flags = wflags}
+{       .id = snd_soc_dapm_mixer, .name = wname, \
+	SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \
+	.kcontrol_news = wcontrols, .num_kcontrols = ARRAY_SIZE(wcontrols), \
+	.event = wevent, .event_flags = wflags}
 
 /* events that are pre and post DAPM */
 #define SND_SOC_DAPM_PRE(wname, wevent) \
@@ -199,35 +222,36 @@ struct device;
 /* stream domain */
 #define SND_SOC_DAPM_AIF_IN(wname, stname, wslot, wreg, wshift, winvert) \
 {	.id = snd_soc_dapm_aif_in, .name = wname, .sname = stname, \
-	.reg = wreg, .shift = wshift, .invert = winvert }
+	SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), }
 #define SND_SOC_DAPM_AIF_IN_E(wname, stname, wslot, wreg, wshift, winvert, \
 			      wevent, wflags)				\
 {	.id = snd_soc_dapm_aif_in, .name = wname, .sname = stname, \
-	.reg = wreg, .shift = wshift, .invert = winvert, \
+	SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \
 	.event = wevent, .event_flags = wflags }
 #define SND_SOC_DAPM_AIF_OUT(wname, stname, wslot, wreg, wshift, winvert) \
 {	.id = snd_soc_dapm_aif_out, .name = wname, .sname = stname, \
-	.reg = wreg, .shift = wshift, .invert = winvert }
+	SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), }
 #define SND_SOC_DAPM_AIF_OUT_E(wname, stname, wslot, wreg, wshift, winvert, \
 			     wevent, wflags)				\
 {	.id = snd_soc_dapm_aif_out, .name = wname, .sname = stname, \
-	.reg = wreg, .shift = wshift, .invert = winvert, \
+	SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \
 	.event = wevent, .event_flags = wflags }
 #define SND_SOC_DAPM_DAC(wname, stname, wreg, wshift, winvert) \
-{	.id = snd_soc_dapm_dac, .name = wname, .sname = stname, .reg = wreg, \
-	.shift = wshift, .invert = winvert}
+{	.id = snd_soc_dapm_dac, .name = wname, .sname = stname, \
+	SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert) }
 #define SND_SOC_DAPM_DAC_E(wname, stname, wreg, wshift, winvert, \
 			   wevent, wflags)				\
-{	.id = snd_soc_dapm_dac, .name = wname, .sname = stname, .reg = wreg, \
-	.shift = wshift, .invert = winvert, \
+{	.id = snd_soc_dapm_dac, .name = wname, .sname = stname, \
+	SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \
 	.event = wevent, .event_flags = wflags}
+
 #define SND_SOC_DAPM_ADC(wname, stname, wreg, wshift, winvert) \
-{	.id = snd_soc_dapm_adc, .name = wname, .sname = stname, .reg = wreg, \
-	.shift = wshift, .invert = winvert}
+{	.id = snd_soc_dapm_adc, .name = wname, .sname = stname, \
+	SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), }
 #define SND_SOC_DAPM_ADC_E(wname, stname, wreg, wshift, winvert, \
 			   wevent, wflags)				\
-{	.id = snd_soc_dapm_adc, .name = wname, .sname = stname, .reg = wreg, \
-	.shift = wshift, .invert = winvert, \
+{	.id = snd_soc_dapm_adc, .name = wname, .sname = stname, \
+	SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \
 	.event = wevent, .event_flags = wflags}
 #define SND_SOC_DAPM_CLOCK_SUPPLY(wname) \
 {	.id = snd_soc_dapm_clock_supply, .name = wname, \
@@ -241,14 +265,14 @@ struct device;
 	.on_val = won_val, .off_val = woff_val, .event = dapm_reg_event, \
 	.event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD}
 #define SND_SOC_DAPM_SUPPLY(wname, wreg, wshift, winvert, wevent, wflags) \
-{	.id = snd_soc_dapm_supply, .name = wname, .reg = wreg,	\
-	.shift = wshift, .invert = winvert, .event = wevent, \
-	.event_flags = wflags}
+{	.id = snd_soc_dapm_supply, .name = wname, \
+	SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \
+	.event = wevent, .event_flags = wflags}
 #define SND_SOC_DAPM_REGULATOR_SUPPLY(wname, wdelay, wflags)	    \
 {	.id = snd_soc_dapm_regulator_supply, .name = wname, \
 	.reg = SND_SOC_NOPM, .shift = wdelay, .event = dapm_regulator_event, \
 	.event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD, \
-	.invert = wflags}
+	.on_val = wflags}
 
 
 /* dapm kcontrol types */
@@ -256,14 +280,26 @@ struct device;
 {	.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
 	.info = snd_soc_info_volsw, \
 	.get = snd_soc_dapm_get_volsw, .put = snd_soc_dapm_put_volsw, \
-	.private_value =  SOC_SINGLE_VALUE(reg, shift, max, invert) }
+	.private_value = SOC_SINGLE_VALUE(reg, shift, max, invert, 0) }
+#define SOC_DAPM_SINGLE_AUTODISABLE(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 = snd_soc_dapm_put_volsw, \
+	.private_value = SOC_SINGLE_VALUE(reg, shift, max, invert, 1) }
 #define SOC_DAPM_SINGLE_TLV(xname, reg, shift, max, invert, tlv_array) \
 {	.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
 	.info = snd_soc_info_volsw, \
 	.access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | SNDRV_CTL_ELEM_ACCESS_READWRITE,\
 	.tlv.p = (tlv_array), \
 	.get = snd_soc_dapm_get_volsw, .put = snd_soc_dapm_put_volsw, \
-	.private_value =  SOC_SINGLE_VALUE(reg, shift, max, invert) }
+	.private_value = SOC_SINGLE_VALUE(reg, shift, max, invert, 0) }
+#define SOC_DAPM_SINGLE_TLV_AUTODISABLE(xname, reg, shift, max, invert, tlv_array) \
+{	.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
+	.info = snd_soc_info_volsw, \
+	.access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | SNDRV_CTL_ELEM_ACCESS_READWRITE,\
+	.tlv.p = (tlv_array), \
+	.get = snd_soc_dapm_get_volsw, .put = snd_soc_dapm_put_volsw, \
+	.private_value = SOC_SINGLE_VALUE(reg, shift, max, invert, 0) }
 #define SOC_DAPM_ENUM(xname, xenum) \
 {	.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
 	.info = snd_soc_info_enum_double, \
@@ -333,6 +369,7 @@ struct snd_soc_dapm_route;
 struct snd_soc_dapm_context;
 struct regulator;
 struct snd_soc_dapm_widget_list;
+struct snd_soc_dapm_update;
 
 int dapm_reg_event(struct snd_soc_dapm_widget *w,
 		   struct snd_kcontrol *kcontrol, int event);
@@ -376,7 +413,7 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card,
 			 struct snd_soc_dapm_widget *sink);
 
 /* dapm path setup */
-int snd_soc_dapm_new_widgets(struct snd_soc_dapm_context *dapm);
+int snd_soc_dapm_new_widgets(struct snd_soc_card *card);
 void snd_soc_dapm_free(struct snd_soc_dapm_context *dapm);
 int snd_soc_dapm_add_routes(struct snd_soc_dapm_context *dapm,
 			    const struct snd_soc_dapm_route *route, int num);
@@ -391,10 +428,12 @@ void snd_soc_dapm_stream_event(struct snd_soc_pcm_runtime *rtd, int stream,
 void snd_soc_dapm_shutdown(struct snd_soc_card *card);
 
 /* external DAPM widget events */
-int snd_soc_dapm_mixer_update_power(struct snd_soc_dapm_widget *widget,
-		struct snd_kcontrol *kcontrol, int connect);
-int snd_soc_dapm_mux_update_power(struct snd_soc_dapm_widget *widget,
-				 struct snd_kcontrol *kcontrol, int mux, struct soc_enum *e);
+int snd_soc_dapm_mixer_update_power(struct snd_soc_dapm_context *dapm,
+		struct snd_kcontrol *kcontrol, int connect,
+		struct snd_soc_dapm_update *update);
+int snd_soc_dapm_mux_update_power(struct snd_soc_dapm_context *dapm,
+		struct snd_kcontrol *kcontrol, int mux, struct soc_enum *e,
+		struct snd_soc_dapm_update *update);
 
 /* dapm sys fs - used by the core */
 int snd_soc_dapm_sys_add(struct device *dev);
@@ -424,6 +463,8 @@ void dapm_mark_io_dirty(struct snd_soc_dapm_context *dapm);
 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_codec *snd_soc_dapm_kcontrol_codec(struct snd_kcontrol *kcontrol);
+
 /* dapm widget types */
 enum snd_soc_dapm_type {
 	snd_soc_dapm_input = 0,		/* input pin */
@@ -455,6 +496,7 @@ enum snd_soc_dapm_type {
 	snd_soc_dapm_dai_in,		/* link to DAI structure */
 	snd_soc_dapm_dai_out,
 	snd_soc_dapm_dai_link,		/* link between two DAI structures */
+	snd_soc_dapm_kcontrol,		/* Auto-disabled kcontrol */
 };
 
 enum snd_soc_dapm_subclass {
@@ -485,7 +527,6 @@ struct snd_soc_dapm_path {
 	/* source (input) and sink (output) widgets */
 	struct snd_soc_dapm_widget *source;
 	struct snd_soc_dapm_widget *sink;
-	struct snd_kcontrol *kcontrol;
 
 	/* status */
 	u32 connect:1;	/* source and sink widgets are connected */
@@ -498,6 +539,7 @@ struct snd_soc_dapm_path {
 
 	struct list_head list_source;
 	struct list_head list_sink;
+	struct list_head list_kcontrol;
 	struct list_head list;
 };
 
@@ -518,12 +560,10 @@ struct snd_soc_dapm_widget {
 	/* dapm control */
 	int reg;				/* negative reg = no direct dapm */
 	unsigned char shift;			/* bits to shift */
-	unsigned int value;				/* widget current value */
 	unsigned int mask;			/* non-shifted mask */
 	unsigned int on_val;			/* on state value */
 	unsigned int off_val;			/* off state value */
 	unsigned char power:1;			/* block power status */
-	unsigned char invert:1;			/* invert the power bit */
 	unsigned char active:1;			/* active stream on DAC, ADC's */
 	unsigned char connected:1;		/* connected codec pin */
 	unsigned char new:1;			/* cnew complete */
@@ -559,7 +599,6 @@ struct snd_soc_dapm_widget {
 };
 
 struct snd_soc_dapm_update {
-	struct snd_soc_dapm_widget *widget;
 	struct snd_kcontrol *kcontrol;
 	int reg;
 	int mask;
@@ -573,8 +612,6 @@ struct snd_soc_dapm_context {
 	struct delayed_work delayed_work;
 	unsigned int idle_bias_off:1; /* Use BIAS_OFF instead of STANDBY */
 
-	struct snd_soc_dapm_update *update;
-
 	void (*seq_notifier)(struct snd_soc_dapm_context *,
 			     enum snd_soc_dapm_type, int);
 

include/sound/soc-dpcm.h

@@ -133,6 +133,6 @@ void snd_soc_dpcm_be_set_state(struct snd_soc_pcm_runtime *be, int stream,
 /* 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 *);
+int soc_dpcm_runtime_update(struct snd_soc_card *);
 
 #endif

include/sound/soc.h

@@ -30,13 +30,13 @@
 /*
  * Convenience kcontrol builders
  */
-#define SOC_DOUBLE_VALUE(xreg, shift_left, shift_right, xmax, xinvert) \
+#define SOC_DOUBLE_VALUE(xreg, shift_left, shift_right, xmax, xinvert, xautodisable) \
 	((unsigned long)&(struct soc_mixer_control) \
 	{.reg = xreg, .rreg = xreg, .shift = shift_left, \
 	.rshift = shift_right, .max = xmax, .platform_max = xmax, \
-	.invert = xinvert})
-#define SOC_SINGLE_VALUE(xreg, xshift, xmax, xinvert) \
-	SOC_DOUBLE_VALUE(xreg, xshift, xshift, xmax, xinvert)
+	.invert = xinvert, .autodisable = xautodisable})
+#define SOC_SINGLE_VALUE(xreg, xshift, xmax, xinvert, xautodisable) \
+	SOC_DOUBLE_VALUE(xreg, xshift, xshift, xmax, xinvert, xautodisable)
 #define SOC_SINGLE_VALUE_EXT(xreg, xmax, xinvert) \
 	((unsigned long)&(struct soc_mixer_control) \
 	{.reg = xreg, .max = xmax, .platform_max = xmax, .invert = xinvert})
@@ -52,7 +52,7 @@
 {	.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
 	.info = snd_soc_info_volsw, .get = snd_soc_get_volsw,\
 	.put = snd_soc_put_volsw, \
-	.private_value =  SOC_SINGLE_VALUE(reg, shift, max, invert) }
+	.private_value = SOC_SINGLE_VALUE(reg, shift, max, invert, 0) }
 #define SOC_SINGLE_RANGE(xname, xreg, xshift, xmin, xmax, xinvert) \
 {	.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\
 	.info = snd_soc_info_volsw_range, .get = snd_soc_get_volsw_range, \
@@ -68,7 +68,7 @@
 	.tlv.p = (tlv_array), \
 	.info = snd_soc_info_volsw, .get = snd_soc_get_volsw,\
 	.put = snd_soc_put_volsw, \
-	.private_value =  SOC_SINGLE_VALUE(reg, shift, max, invert) }
+	.private_value = SOC_SINGLE_VALUE(reg, shift, max, invert, 0) }
 #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 | \
@@ -97,7 +97,7 @@
 	.info = snd_soc_info_volsw, .get = snd_soc_get_volsw, \
 	.put = snd_soc_put_volsw, \
 	.private_value = SOC_DOUBLE_VALUE(reg, shift_left, shift_right, \
-					  max, invert) }
+					  max, invert, 0) }
 #define SOC_DOUBLE_R(xname, reg_left, reg_right, xshift, xmax, xinvert) \
 {	.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \
 	.info = snd_soc_info_volsw, \
@@ -119,7 +119,7 @@
 	.info = snd_soc_info_volsw, .get = snd_soc_get_volsw, \
 	.put = snd_soc_put_volsw, \
 	.private_value = SOC_DOUBLE_VALUE(reg, shift_left, shift_right, \
-					  max, invert) }
+					  max, invert, 0) }
 #define SOC_DOUBLE_R_TLV(xname, reg_left, reg_right, xshift, xmax, xinvert, tlv_array) \
 {	.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\
 	.access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\
@@ -190,14 +190,14 @@
 {	.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
 	.info = snd_soc_info_volsw, \
 	.get = xhandler_get, .put = xhandler_put, \
-	.private_value = SOC_SINGLE_VALUE(xreg, xshift, xmax, xinvert) }
+	.private_value = SOC_SINGLE_VALUE(xreg, xshift, xmax, xinvert, 0) }
 #define SOC_DOUBLE_EXT(xname, reg, shift_left, shift_right, max, invert,\
 	 xhandler_get, xhandler_put) \
 {	.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\
 	.info = snd_soc_info_volsw, \
 	.get = xhandler_get, .put = xhandler_put, \
 	.private_value = \
-		SOC_DOUBLE_VALUE(reg, shift_left, shift_right, max, invert) }
+		SOC_DOUBLE_VALUE(reg, shift_left, shift_right, max, invert, 0) }
 #define SOC_SINGLE_EXT_TLV(xname, xreg, xshift, xmax, xinvert,\
 	 xhandler_get, xhandler_put, tlv_array) \
 {	.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
@@ -206,7 +206,7 @@
 	.tlv.p = (tlv_array), \
 	.info = snd_soc_info_volsw, \
 	.get = xhandler_get, .put = xhandler_put, \
-	.private_value = SOC_SINGLE_VALUE(xreg, xshift, xmax, xinvert) }
+	.private_value = SOC_SINGLE_VALUE(xreg, xshift, xmax, xinvert, 0) }
 #define SOC_DOUBLE_EXT_TLV(xname, xreg, shift_left, shift_right, xmax, xinvert,\
 	 xhandler_get, xhandler_put, tlv_array) \
 {	.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \
@@ -216,7 +216,7 @@
 	.info = snd_soc_info_volsw, \
 	.get = xhandler_get, .put = xhandler_put, \
 	.private_value = SOC_DOUBLE_VALUE(xreg, shift_left, shift_right, \
-					  xmax, xinvert) }
+					  xmax, xinvert, 0) }
 #define SOC_DOUBLE_R_EXT_TLV(xname, reg_left, reg_right, xshift, xmax, xinvert,\
 	 xhandler_get, xhandler_put, tlv_array) \
 {	.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \
@@ -234,7 +234,7 @@
 	.private_value = xdata }
 #define SOC_ENUM_EXT(xname, xenum, xhandler_get, xhandler_put) \
 {	.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
-	.info = snd_soc_info_enum_ext, \
+	.info = snd_soc_info_enum_double, \
 	.get = xhandler_get, .put = xhandler_put, \
 	.private_value = (unsigned long)&xenum }
 
@@ -468,6 +468,8 @@ int snd_soc_new_ac97_codec(struct snd_soc_codec *codec,
 void snd_soc_free_ac97_codec(struct snd_soc_codec *codec);
 
 int snd_soc_set_ac97_ops(struct snd_ac97_bus_ops *ops);
+int snd_soc_set_ac97_ops_of_reset(struct snd_ac97_bus_ops *ops,
+		struct platform_device *pdev);
 
 /*
  *Controls
@@ -475,6 +477,8 @@ int snd_soc_set_ac97_ops(struct snd_ac97_bus_ops *ops);
 struct snd_kcontrol *snd_soc_cnew(const struct snd_kcontrol_new *_template,
 				  void *data, const char *long_name,
 				  const char *prefix);
+struct snd_kcontrol *snd_soc_card_get_kcontrol(struct snd_soc_card *soc_card,
+					       const char *name);
 int snd_soc_add_codec_controls(struct snd_soc_codec *codec,
 	const struct snd_kcontrol_new *controls, int num_controls);
 int snd_soc_add_platform_controls(struct snd_soc_platform *platform,
@@ -485,8 +489,6 @@ int snd_soc_add_dai_controls(struct snd_soc_dai *dai,
 	const struct snd_kcontrol_new *controls, int num_controls);
 int snd_soc_info_enum_double(struct snd_kcontrol *kcontrol,
 	struct snd_ctl_elem_info *uinfo);
-int snd_soc_info_enum_ext(struct snd_kcontrol *kcontrol,
-	struct snd_ctl_elem_info *uinfo);
 int snd_soc_get_enum_double(struct snd_kcontrol *kcontrol,
 	struct snd_ctl_elem_value *ucontrol);
 int snd_soc_put_enum_double(struct snd_kcontrol *kcontrol,
@@ -497,8 +499,6 @@ int snd_soc_put_value_enum_double(struct snd_kcontrol *kcontrol,
 	struct snd_ctl_elem_value *ucontrol);
 int snd_soc_info_volsw(struct snd_kcontrol *kcontrol,
 	struct snd_ctl_elem_info *uinfo);
-int snd_soc_info_volsw_ext(struct snd_kcontrol *kcontrol,
-	struct snd_ctl_elem_info *uinfo);
 #define snd_soc_info_bool_ext		snd_ctl_boolean_mono_info
 int snd_soc_get_volsw(struct snd_kcontrol *kcontrol,
 	struct snd_ctl_elem_value *ucontrol);
@@ -697,7 +697,6 @@ struct snd_soc_codec {
 	unsigned int probed:1; /* Codec has been probed */
 	unsigned int ac97_registered:1; /* Codec has been AC97 registered */
 	unsigned int ac97_created:1; /* Codec has been created by SoC */
-	unsigned int sysfs_registered:1; /* codec has been sysfs registered */
 	unsigned int cache_init:1; /* codec cache has been initialized */
 	unsigned int using_regmap:1; /* using regmap access */
 	u32 cache_only;  /* Suppress writes to hardware */
@@ -705,7 +704,6 @@ struct snd_soc_codec {
 
 	/* codec IO */
 	void *control_data; /* codec control (i2c/3wire) data */
-	enum snd_soc_control_type control_type;
 	hw_write_t hw_write;
 	unsigned int (*hw_read)(struct snd_soc_codec *, unsigned int);
 	unsigned int (*read)(struct snd_soc_codec *, unsigned int);
@@ -724,7 +722,6 @@ struct snd_soc_codec {
 #ifdef CONFIG_DEBUG_FS
 	struct dentry *debugfs_codec_root;
 	struct dentry *debugfs_reg;
-	struct dentry *debugfs_dapm;
 #endif
 };
 
@@ -849,7 +846,6 @@ struct snd_soc_platform {
 
 #ifdef CONFIG_DEBUG_FS
 	struct dentry *debugfs_platform_root;
-	struct dentry *debugfs_dapm;
 #endif
 };
 
@@ -934,6 +930,10 @@ struct snd_soc_dai_link {
 	/* machine stream operations */
 	const struct snd_soc_ops *ops;
 	const struct snd_soc_compr_ops *compr_ops;
+
+	/* For unidirectional dai links */
+	bool playback_only;
+	bool capture_only;
 };
 
 struct snd_soc_codec_conf {
@@ -1042,6 +1042,7 @@ struct snd_soc_card {
 	/* Generic DAPM context for the card */
 	struct snd_soc_dapm_context dapm;
 	struct snd_soc_dapm_stats dapm_stats;
+	struct snd_soc_dapm_update *update;
 
 #ifdef CONFIG_DEBUG_FS
 	struct dentry *debugfs_card_root;
@@ -1087,7 +1088,9 @@ struct snd_soc_pcm_runtime {
 /* mixer control */
 struct soc_mixer_control {
 	int min, max, platform_max;
-	unsigned int reg, rreg, shift, rshift, invert;
+	unsigned int reg, rreg, shift, rshift;
+	unsigned int invert:1;
+	unsigned int autodisable:1;
 };
 
 struct soc_bytes {

include/uapi/sound/hdspm.h

@@ -111,7 +111,7 @@ struct hdspm_ltc {
 	enum hdspm_ltc_input_format input_format;
 };
 
-#define SNDRV_HDSPM_IOCTL_GET_LTC _IOR('H', 0x46, struct hdspm_mixer_ioctl)
+#define SNDRV_HDSPM_IOCTL_GET_LTC _IOR('H', 0x46, struct hdspm_ltc)
 
 /**
  * The status data reflects the device's current state

sound/arm/pxa2xx-ac97.c

@@ -14,12 +14,14 @@
 #include <linux/io.h>
 #include <linux/module.h>
 #include <linux/platform_device.h>
+#include <linux/dmaengine.h>
 
 #include <sound/core.h>
 #include <sound/pcm.h>
 #include <sound/ac97_codec.h>
 #include <sound/initval.h>
 #include <sound/pxa2xx-lib.h>
+#include <sound/dmaengine_pcm.h>
 
 #include <mach/regs-ac97.h>
 #include <mach/audio.h>
@@ -41,20 +43,20 @@ static struct snd_ac97_bus_ops pxa2xx_ac97_ops = {
 	.reset	= pxa2xx_ac97_reset,
 };
 
-static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_out = {
-	.name			= "AC97 PCM out",
-	.dev_addr		= __PREG(PCDR),
-	.drcmr			= &DRCMR(12),
-	.dcmd			= DCMD_INCSRCADDR | DCMD_FLOWTRG |
-				  DCMD_BURST32 | DCMD_WIDTH4,
+static unsigned long pxa2xx_ac97_pcm_out_req = 12;
+static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_out = {
+	.addr		= __PREG(PCDR),
+	.addr_width	= DMA_SLAVE_BUSWIDTH_4_BYTES,
+	.maxburst	= 32,
+	.filter_data	= &pxa2xx_ac97_pcm_out_req,
 };
 
-static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_in = {
-	.name			= "AC97 PCM in",
-	.dev_addr		= __PREG(PCDR),
-	.drcmr			= &DRCMR(11),
-	.dcmd			= DCMD_INCTRGADDR | DCMD_FLOWSRC |
-				  DCMD_BURST32 | DCMD_WIDTH4,
+static unsigned long pxa2xx_ac97_pcm_in_req = 11;
+static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_in = {
+	.addr		= __PREG(PCDR),
+	.addr_width	= DMA_SLAVE_BUSWIDTH_4_BYTES,
+	.maxburst	= 32,
+	.filter_data	= &pxa2xx_ac97_pcm_in_req,
 };
 
 static struct snd_pcm *pxa2xx_ac97_pcm;

sound/arm/pxa2xx-pcm-lib.c

@@ -7,11 +7,13 @@
 #include <linux/slab.h>
 #include <linux/module.h>
 #include <linux/dma-mapping.h>
+#include <linux/dmaengine.h>
 
 #include <sound/core.h>
 #include <sound/pcm.h>
 #include <sound/pcm_params.h>
 #include <sound/pxa2xx-lib.h>
+#include <sound/dmaengine_pcm.h>
 
 #include <mach/dma.h>
 
@@ -43,6 +45,35 @@ int __pxa2xx_pcm_hw_params(struct snd_pcm_substream *substream,
 	size_t period = params_period_bytes(params);
 	pxa_dma_desc *dma_desc;
 	dma_addr_t dma_buff_phys, next_desc_phys;
+	u32 dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG;
+
+	/* temporary transition hack */
+	switch (rtd->params->addr_width) {
+	case DMA_SLAVE_BUSWIDTH_1_BYTE:
+		dcmd |= DCMD_WIDTH1;
+		break;
+	case DMA_SLAVE_BUSWIDTH_2_BYTES:
+		dcmd |= DCMD_WIDTH2;
+		break;
+	case DMA_SLAVE_BUSWIDTH_4_BYTES:
+		dcmd |= DCMD_WIDTH4;
+		break;
+	default:
+		/* can't happen */
+		break;
+	}
+
+	switch (rtd->params->maxburst) {
+	case 8:
+		dcmd |= DCMD_BURST8;
+		break;
+	case 16:
+		dcmd |= DCMD_BURST16;
+		break;
+	case 32:
+		dcmd |= DCMD_BURST32;
+		break;
+	}
 
 	snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
 	runtime->dma_bytes = totsize;
@@ -55,14 +86,14 @@ int __pxa2xx_pcm_hw_params(struct snd_pcm_substream *substream,
 		dma_desc->ddadr = next_desc_phys;
 		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
 			dma_desc->dsadr = dma_buff_phys;
-			dma_desc->dtadr = rtd->params->dev_addr;
+			dma_desc->dtadr = rtd->params->addr;
 		} else {
-			dma_desc->dsadr = rtd->params->dev_addr;
+			dma_desc->dsadr = rtd->params->addr;
 			dma_desc->dtadr = dma_buff_phys;
 		}
 		if (period > totsize)
 			period = totsize;
-		dma_desc->dcmd = rtd->params->dcmd | period | DCMD_ENDIRQEN;
+		dma_desc->dcmd = dcmd | period | DCMD_ENDIRQEN;
 		dma_desc++;
 		dma_buff_phys += period;
 	} while (totsize -= period);
@@ -76,8 +107,10 @@ int __pxa2xx_pcm_hw_free(struct snd_pcm_substream *substream)
 {
 	struct pxa2xx_runtime_data *rtd = substream->runtime->private_data;
 
-	if (rtd && rtd->params && rtd->params->drcmr)
-		*rtd->params->drcmr = 0;
+	if (rtd && rtd->params && rtd->params->filter_data) {
+		unsigned long req = *(unsigned long *) rtd->params->filter_data;
+		DRCMR(req) = 0;
+	}
 
 	snd_pcm_set_runtime_buffer(substream, NULL);
 	return 0;
@@ -136,6 +169,7 @@ EXPORT_SYMBOL(pxa2xx_pcm_pointer);
 int __pxa2xx_pcm_prepare(struct snd_pcm_substream *substream)
 {
 	struct pxa2xx_runtime_data *prtd = substream->runtime->private_data;
+	unsigned long req;
 
 	if (!prtd || !prtd->params)
 		return 0;
@@ -146,7 +180,8 @@ int __pxa2xx_pcm_prepare(struct snd_pcm_substream *substream)
 	DCSR(prtd->dma_ch) &= ~DCSR_RUN;
 	DCSR(prtd->dma_ch) = 0;
 	DCMD(prtd->dma_ch) = 0;
-	*prtd->params->drcmr = prtd->dma_ch | DRCMR_MAPVLD;
+	req = *(unsigned long *) prtd->params->filter_data;
+	DRCMR(req) = prtd->dma_ch | DRCMR_MAPVLD;
 
 	return 0;
 }
@@ -155,7 +190,6 @@ EXPORT_SYMBOL(__pxa2xx_pcm_prepare);
 void pxa2xx_pcm_dma_irq(int dma_ch, void *dev_id)
 {
 	struct snd_pcm_substream *substream = dev_id;
-	struct pxa2xx_runtime_data *rtd = substream->runtime->private_data;
 	int dcsr;
 
 	dcsr = DCSR(dma_ch);
@@ -164,8 +198,8 @@ void pxa2xx_pcm_dma_irq(int dma_ch, void *dev_id)
 	if (dcsr & DCSR_ENDINTR) {
 		snd_pcm_period_elapsed(substream);
 	} else {
-		printk(KERN_ERR "%s: DMA error on channel %d (DCSR=%#x)\n",
-			rtd->params->name, dma_ch, dcsr);
+		printk(KERN_ERR "DMA error on channel %d (DCSR=%#x)\n",
+			dma_ch, dcsr);
 		snd_pcm_stream_lock(substream);
 		snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN);
 		snd_pcm_stream_unlock(substream);

sound/arm/pxa2xx-pcm.c

@@ -11,8 +11,11 @@
  */
 
 #include <linux/module.h>
+#include <linux/dmaengine.h>
+
 #include <sound/core.h>
 #include <sound/pxa2xx-lib.h>
+#include <sound/dmaengine_pcm.h>
 
 #include "pxa2xx-pcm.h"
 
@@ -40,7 +43,7 @@ static int pxa2xx_pcm_open(struct snd_pcm_substream *substream)
 
 	rtd->params = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ?
 		      client->playback_params : client->capture_params;
-	ret = pxa_request_dma(rtd->params->name, DMA_PRIO_LOW,
+	ret = pxa_request_dma("dma", DMA_PRIO_LOW,
 			      pxa2xx_pcm_dma_irq, substream);
 	if (ret < 0)
 		goto err2;

sound/arm/pxa2xx-pcm.h

@@ -13,14 +13,14 @@
 
 struct pxa2xx_runtime_data {
 	int dma_ch;
-	struct pxa2xx_pcm_dma_params *params;
+	struct snd_dmaengine_dai_dma_data *params;
 	pxa_dma_desc *dma_desc_array;
 	dma_addr_t dma_desc_array_phys;
 };
 
 struct pxa2xx_pcm_client {
-	struct pxa2xx_pcm_dma_params *playback_params;
-	struct pxa2xx_pcm_dma_params *capture_params;
+	struct snd_dmaengine_dai_dma_data *playback_params;
+	struct snd_dmaengine_dai_dma_data *capture_params;
 	int (*startup)(struct snd_pcm_substream *);
 	void (*shutdown)(struct snd_pcm_substream *);
 	int (*prepare)(struct snd_pcm_substream *);

sound/core/Kconfig

@@ -6,6 +6,9 @@ config SND_PCM
 	tristate
 	select SND_TIMER
 
+config SND_DMAENGINE_PCM
+	tristate
+
 config SND_HWDEP
 	tristate
 

sound/core/Makefile

@@ -13,6 +13,8 @@ snd-$(CONFIG_SND_JACK)	  += jack.o
 snd-pcm-objs := pcm.o pcm_native.o pcm_lib.o pcm_timer.o pcm_misc.o \
 		pcm_memory.o
 
+snd-pcm-dmaengine-objs := pcm_dmaengine.o
+
 snd-page-alloc-y := memalloc.o
 snd-page-alloc-$(CONFIG_SND_DMA_SGBUF) += sgbuf.o
 
@@ -30,6 +32,7 @@ obj-$(CONFIG_SND_TIMER)		+= snd-timer.o
 obj-$(CONFIG_SND_HRTIMER)	+= snd-hrtimer.o
 obj-$(CONFIG_SND_RTCTIMER)	+= snd-rtctimer.o
 obj-$(CONFIG_SND_PCM)		+= snd-pcm.o snd-page-alloc.o
+obj-$(CONFIG_SND_DMAENGINE_PCM)	+= snd-pcm-dmaengine.o
 obj-$(CONFIG_SND_RAWMIDI)	+= snd-rawmidi.o
 
 obj-$(CONFIG_SND_OSSEMUL)	+= oss/

sound/core/pcm_lib.c

@@ -184,7 +184,7 @@ static void xrun(struct snd_pcm_substream *substream)
 	do {								\
 		if (xrun_debug(substream, XRUN_DEBUG_BASIC)) {		\
 			xrun_log_show(substream);			\
-			if (printk_ratelimit()) {			\
+			if (snd_printd_ratelimit()) {			\
 				snd_printd("PCM: " fmt, ##args);	\
 			}						\
 			dump_stack_on_xrun(substream);			\
@@ -342,7 +342,7 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream,
 		return -EPIPE;
 	}
 	if (pos >= runtime->buffer_size) {
-		if (printk_ratelimit()) {
+		if (snd_printd_ratelimit()) {
 			char name[16];
 			snd_pcm_debug_name(substream, name, sizeof(name));
 			xrun_log_show(substream);

sound/drivers/dummy.c

@@ -1022,7 +1022,7 @@ static void dummy_proc_write(struct snd_info_entry *entry,
 		if (i >= ARRAY_SIZE(fields))
 			continue;
 		snd_info_get_str(item, ptr, sizeof(item));
-		if (strict_strtoull(item, 0, &val))
+		if (kstrtoull(item, 0, &val))
 			continue;
 		if (fields[i].size == sizeof(int))
 			*get_dummy_int_ptr(dummy, fields[i].offset) = val;

sound/firewire/speakers.c

@@ -49,7 +49,6 @@ struct fwspk {
 	struct snd_card *card;
 	struct fw_unit *unit;
 	const struct device_info *device_info;
-	struct snd_pcm_substream *pcm;
 	struct mutex mutex;
 	struct cmp_connection connection;
 	struct amdtp_out_stream stream;
@@ -363,8 +362,7 @@ static int fwspk_create_pcm(struct fwspk *fwspk)
 		return err;
 	pcm->private_data = fwspk;
 	strcpy(pcm->name, fwspk->device_info->short_name);
-	fwspk->pcm = pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream;
-	fwspk->pcm->ops = &ops;
+	snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &ops);
 	return 0;
 }
 

sound/isa/gus/interwave.c

@@ -443,8 +443,7 @@ static void snd_interwave_detect_memory(struct snd_gus_card *gus)
 		for (i = 0; i < 8; ++i)
 			iwave[i] = snd_gf1_peek(gus, bank_pos + i);
 #ifdef CONFIG_SND_DEBUG_ROM
-		printk(KERN_DEBUG "ROM at 0x%06x = %*phC\n", bank_pos,
-				  8, iwave);
+		printk(KERN_DEBUG "ROM at 0x%06x = %8phC\n", bank_pos, iwave);
 #endif
 		if (strncmp(iwave, "INTRWAVE", 8))
 			continue;	/* first check */

sound/oss/dmabuf.c

@@ -557,7 +557,6 @@ int DMAbuf_getrdbuffer(int dev, char **buf, int *len, int dontblock)
 	unsigned long flags;
 	int err = 0, n = 0;
 	struct dma_buffparms *dmap = adev->dmap_in;
-	int go;
 
 	if (!(adev->open_mode & OPEN_READ))
 		return -EIO;
@@ -584,7 +583,7 @@ int DMAbuf_getrdbuffer(int dev, char **buf, int *len, int dontblock)
 			spin_unlock_irqrestore(&dmap->lock,flags);
 			return -EAGAIN;
 		}
-		if ((go = adev->go))
+		if (adev->go)
 			timeout = dmabuf_timeout(dmap);
 
 		spin_unlock_irqrestore(&dmap->lock,flags);

sound/pci/hda/Kconfig

@@ -152,14 +152,9 @@ config SND_HDA_CODEC_HDMI
 	  This module is automatically loaded at probing.
 
 config SND_HDA_I915
-	bool "Build Display HD-audio controller/codec power well support for i915 cards"
+	bool
+	default y
 	depends on DRM_I915
-	help
-	  Say Y here to include full HDMI and DisplayPort HD-audio controller/codec
-	  power-well support for Intel Haswell graphics cards based on the i915 driver.
-
-	  Note that this option must be enabled for Intel Haswell C+ stepping machines, otherwise
-	  the GPU audio controller/codecs will not be initialized or damaged when exit from S3 mode.
 
 config SND_HDA_CODEC_CIRRUS
 	bool "Build Cirrus Logic codec support"

sound/pci/hda/hda_codec.c

@@ -666,6 +666,64 @@ int snd_hda_get_conn_index(struct hda_codec *codec, hda_nid_t mux,
 }
 EXPORT_SYMBOL_HDA(snd_hda_get_conn_index);
 
+
+/* return DEVLIST_LEN parameter of the given widget */
+static unsigned int get_num_devices(struct hda_codec *codec, hda_nid_t nid)
+{
+	unsigned int wcaps = get_wcaps(codec, nid);
+	unsigned int parm;
+
+	if (!codec->dp_mst || !(wcaps & AC_WCAP_DIGITAL) ||
+	    get_wcaps_type(wcaps) != AC_WID_PIN)
+		return 0;
+
+	parm = snd_hda_param_read(codec, nid, AC_PAR_DEVLIST_LEN);
+	if (parm == -1 && codec->bus->rirb_error)
+		parm = 0;
+	return parm & AC_DEV_LIST_LEN_MASK;
+}
+
+/**
+ * snd_hda_get_devices - copy device list without cache
+ * @codec: the HDA codec
+ * @nid: NID of the pin to parse
+ * @dev_list: device list array
+ * @max_devices: max. number of devices to store
+ *
+ * Copy the device list. This info is dynamic and so not cached.
+ * Currently called only from hda_proc.c, so not exported.
+ */
+int snd_hda_get_devices(struct hda_codec *codec, hda_nid_t nid,
+			u8 *dev_list, int max_devices)
+{
+	unsigned int parm;
+	int i, dev_len, devices;
+
+	parm = get_num_devices(codec, nid);
+	if (!parm)	/* not multi-stream capable */
+		return 0;
+
+	dev_len = parm + 1;
+	dev_len = dev_len < max_devices ? dev_len : max_devices;
+
+	devices = 0;
+	while (devices < dev_len) {
+		parm = snd_hda_codec_read(codec, nid, 0,
+					  AC_VERB_GET_DEVICE_LIST, devices);
+		if (parm == -1 && codec->bus->rirb_error)
+			break;
+
+		for (i = 0; i < 8; i++) {
+			dev_list[devices] = (u8)parm;
+			parm >>= 4;
+			devices++;
+			if (devices >= dev_len)
+				break;
+		}
+	}
+	return devices;
+}
+
 /**
  * snd_hda_queue_unsol_event - add an unsolicited event to queue
  * @bus: the BUS
@@ -1216,11 +1274,13 @@ static void hda_jackpoll_work(struct work_struct *work)
 {
 	struct hda_codec *codec =
 		container_of(work, struct hda_codec, jackpoll_work.work);
-	if (!codec->jackpoll_interval)
-		return;
 
 	snd_hda_jack_set_dirty_all(codec);
 	snd_hda_jack_poll_all(codec);
+
+	if (!codec->jackpoll_interval)
+		return;
+
 	queue_delayed_work(codec->bus->workq, &codec->jackpoll_work,
 			   codec->jackpoll_interval);
 }

sound/pci/hda/hda_codec.h

@@ -94,6 +94,8 @@ enum {
 #define AC_VERB_GET_HDMI_DIP_XMIT		0x0f32
 #define AC_VERB_GET_HDMI_CP_CTRL		0x0f33
 #define AC_VERB_GET_HDMI_CHAN_SLOT		0x0f34
+#define AC_VERB_GET_DEVICE_SEL			0xf35
+#define AC_VERB_GET_DEVICE_LIST			0xf36
 
 /*
  * SET verbs
@@ -133,6 +135,7 @@ enum {
 #define AC_VERB_SET_HDMI_DIP_XMIT		0x732
 #define AC_VERB_SET_HDMI_CP_CTRL		0x733
 #define AC_VERB_SET_HDMI_CHAN_SLOT		0x734
+#define AC_VERB_SET_DEVICE_SEL			0x735
 
 /*
  * Parameter IDs
@@ -154,6 +157,7 @@ enum {
 #define AC_PAR_GPIO_CAP			0x11
 #define AC_PAR_AMP_OUT_CAP		0x12
 #define AC_PAR_VOL_KNB_CAP		0x13
+#define AC_PAR_DEVLIST_LEN		0x15
 #define AC_PAR_HDMI_LPCM_CAP		0x20
 
 /*
@@ -251,6 +255,11 @@ enum {
 #define AC_UNSOL_RES_TAG_SHIFT		26
 #define AC_UNSOL_RES_SUBTAG		(0x1f<<21)
 #define AC_UNSOL_RES_SUBTAG_SHIFT	21
+#define AC_UNSOL_RES_DE			(0x3f<<15)  /* Device Entry
+						     * (for DP1.2 MST)
+						     */
+#define AC_UNSOL_RES_DE_SHIFT		15
+#define AC_UNSOL_RES_IA			(1<<2)	/* Inactive (for DP1.2 MST) */
 #define AC_UNSOL_RES_ELDV		(1<<1)	/* ELD Data valid (for HDMI) */
 #define AC_UNSOL_RES_PD			(1<<0)	/* pinsense detect */
 #define AC_UNSOL_RES_CP_STATE		(1<<1)	/* content protection */
@@ -352,6 +361,10 @@ enum {
 #define AC_LPCMCAP_44K			(1<<30)	/* 44.1kHz support */
 #define AC_LPCMCAP_44K_MS		(1<<31)	/* 44.1kHz-multiplies support */
 
+/* Display pin's device list length */
+#define AC_DEV_LIST_LEN_MASK		0x3f
+#define AC_MAX_DEV_LIST_LEN		64
+
 /*
  * Control Parameters
  */
@@ -460,6 +473,11 @@ enum {
 #define AC_DEFCFG_PORT_CONN		(0x3<<30)
 #define AC_DEFCFG_PORT_CONN_SHIFT	30
 
+/* Display pin's device list entry */
+#define AC_DE_PD			(1<<0)
+#define AC_DE_ELDV			(1<<1)
+#define AC_DE_IA			(1<<2)
+
 /* device device types (0x0-0xf) */
 enum {
 	AC_JACK_LINE_OUT,
@@ -885,6 +903,7 @@ struct hda_codec {
 	unsigned int pcm_format_first:1; /* PCM format must be set first */
 	unsigned int epss:1;		/* supporting EPSS? */
 	unsigned int cached_write:1;	/* write only to caches */
+	unsigned int dp_mst:1; /* support DP1.2 Multi-stream transport */
 #ifdef CONFIG_PM
 	unsigned int power_on :1;	/* current (global) power-state */
 	unsigned int d3_stop_clk:1;	/* support D3 operation without BCLK */
@@ -972,6 +991,8 @@ 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,
 			   hda_nid_t nid, int recursive);
+int snd_hda_get_devices(struct hda_codec *codec, hda_nid_t nid,
+			u8 *dev_list, int max_devices);
 int snd_hda_query_supported_pcm(struct hda_codec *codec, hda_nid_t nid,
 				u32 *ratesp, u64 *formatsp, unsigned int *bpsp);
 

sound/pci/hda/hda_generic.c

@@ -142,6 +142,9 @@ static void parse_user_hints(struct hda_codec *codec)
 	val = snd_hda_get_bool_hint(codec, "primary_hp");
 	if (val >= 0)
 		spec->no_primary_hp = !val;
+	val = snd_hda_get_bool_hint(codec, "multi_io");
+	if (val >= 0)
+		spec->no_multi_io = !val;
 	val = snd_hda_get_bool_hint(codec, "multi_cap_vol");
 	if (val >= 0)
 		spec->multi_cap_vol = !!val;
@@ -813,6 +816,8 @@ static void resume_path_from_idx(struct hda_codec *codec, int path_idx)
 
 static int hda_gen_mixer_mute_put(struct snd_kcontrol *kcontrol,
 				  struct snd_ctl_elem_value *ucontrol);
+static int hda_gen_bind_mute_put(struct snd_kcontrol *kcontrol,
+				 struct snd_ctl_elem_value *ucontrol);
 
 enum {
 	HDA_CTL_WIDGET_VOL,
@@ -830,7 +835,13 @@ static const struct snd_kcontrol_new control_templates[] = {
 		.put = hda_gen_mixer_mute_put, /* replaced */
 		.private_value = HDA_COMPOSE_AMP_VAL(0, 3, 0, 0),
 	},
-	HDA_BIND_MUTE(NULL, 0, 0, 0),
+	{
+		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+		.info = snd_hda_mixer_amp_switch_info,
+		.get = snd_hda_mixer_bind_switch_get,
+		.put = hda_gen_bind_mute_put, /* replaced */
+		.private_value = HDA_COMPOSE_AMP_VAL(0, 3, 0, 0),
+	},
 };
 
 /* add dynamic controls from template */
@@ -937,8 +948,8 @@ static int add_stereo_sw(struct hda_codec *codec, const char *pfx,
 }
 
 /* playback mute control with the software mute bit check */
-static int hda_gen_mixer_mute_put(struct snd_kcontrol *kcontrol,
-				  struct snd_ctl_elem_value *ucontrol)
+static void sync_auto_mute_bits(struct snd_kcontrol *kcontrol,
+				struct snd_ctl_elem_value *ucontrol)
 {
 	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
 	struct hda_gen_spec *spec = codec->spec;
@@ -949,10 +960,22 @@ static int hda_gen_mixer_mute_put(struct snd_kcontrol *kcontrol,
 		ucontrol->value.integer.value[0] &= enabled;
 		ucontrol->value.integer.value[1] &= enabled;
 	}
+}
 
+static int hda_gen_mixer_mute_put(struct snd_kcontrol *kcontrol,
+				  struct snd_ctl_elem_value *ucontrol)
+{
+	sync_auto_mute_bits(kcontrol, ucontrol);
 	return snd_hda_mixer_amp_switch_put(kcontrol, ucontrol);
 }
 
+static int hda_gen_bind_mute_put(struct snd_kcontrol *kcontrol,
+				 struct snd_ctl_elem_value *ucontrol)
+{
+	sync_auto_mute_bits(kcontrol, ucontrol);
+	return snd_hda_mixer_bind_switch_put(kcontrol, ucontrol);
+}
+
 /* any ctl assigned to the path with the given index? */
 static bool path_has_mixer(struct hda_codec *codec, int path_idx, int ctl_type)
 {
@@ -1541,7 +1564,8 @@ static int fill_and_eval_dacs(struct hda_codec *codec,
 					      cfg->speaker_pins,
 					      spec->multiout.extra_out_nid,
 					      spec->speaker_paths);
-			if (fill_mio_first && cfg->line_outs == 1 &&
+			if (!spec->no_multi_io &&
+			    fill_mio_first && cfg->line_outs == 1 &&
 			    cfg->line_out_type != AUTO_PIN_SPEAKER_OUT) {
 				err = fill_multi_ios(codec, cfg->line_out_pins[0], true);
 				if (!err)
@@ -1554,7 +1578,7 @@ static int fill_and_eval_dacs(struct hda_codec *codec,
 				   spec->private_dac_nids, spec->out_paths,
 				   spec->main_out_badness);
 
-	if (fill_mio_first &&
+	if (!spec->no_multi_io && fill_mio_first &&
 	    cfg->line_outs == 1 && cfg->line_out_type != AUTO_PIN_SPEAKER_OUT) {
 		/* try to fill multi-io first */
 		err = fill_multi_ios(codec, cfg->line_out_pins[0], false);
@@ -1582,7 +1606,8 @@ static int fill_and_eval_dacs(struct hda_codec *codec,
 			return err;
 		badness += err;
 	}
-	if (cfg->line_outs == 1 && cfg->line_out_type != AUTO_PIN_SPEAKER_OUT) {
+	if (!spec->no_multi_io &&
+	    cfg->line_outs == 1 && cfg->line_out_type != AUTO_PIN_SPEAKER_OUT) {
 		err = fill_multi_ios(codec, cfg->line_out_pins[0], false);
 		if (err < 0)
 			return err;
@@ -1600,7 +1625,8 @@ static int fill_and_eval_dacs(struct hda_codec *codec,
 				check_aamix_out_path(codec, spec->speaker_paths[0]);
 	}
 
-	if (cfg->hp_outs && cfg->line_out_type == AUTO_PIN_SPEAKER_OUT)
+	if (!spec->no_multi_io &&
+	    cfg->hp_outs && cfg->line_out_type == AUTO_PIN_SPEAKER_OUT)
 		if (count_multiio_pins(codec, cfg->hp_pins[0]) >= 2)
 			spec->multi_ios = 1; /* give badness */
 
@@ -3724,7 +3750,8 @@ static int mux_select(struct hda_codec *codec, unsigned int adc_idx,
 /* check each pin in the given array; returns true if any of them is plugged */
 static bool detect_jacks(struct hda_codec *codec, int num_pins, hda_nid_t *pins)
 {
-	int i, present = 0;
+	int i;
+	bool present = false;
 
 	for (i = 0; i < num_pins; i++) {
 		hda_nid_t nid = pins[i];
@@ -3733,14 +3760,15 @@ static bool detect_jacks(struct hda_codec *codec, int num_pins, hda_nid_t *pins)
 		/* don't detect pins retasked as inputs */
 		if (snd_hda_codec_get_pin_target(codec, nid) & AC_PINCTL_IN_EN)
 			continue;
-		present |= snd_hda_jack_detect(codec, nid);
+		if (snd_hda_jack_detect_state(codec, nid) == HDA_JACK_PRESENT)
+			present = true;
 	}
 	return present;
 }
 
 /* standard HP/line-out auto-mute helper */
 static void do_automute(struct hda_codec *codec, int num_pins, hda_nid_t *pins,
-			bool mute)
+			int *paths, bool mute)
 {
 	struct hda_gen_spec *spec = codec->spec;
 	int i;
@@ -3752,10 +3780,19 @@ static void do_automute(struct hda_codec *codec, int num_pins, hda_nid_t *pins,
 			break;
 
 		if (spec->auto_mute_via_amp) {
+			struct nid_path *path;
+			hda_nid_t mute_nid;
+
+			path = snd_hda_get_path_from_idx(codec, paths[i]);
+			if (!path)
+				continue;
+			mute_nid = get_amp_nid_(path->ctls[NID_PATH_MUTE_CTL]);
+			if (!mute_nid)
+				continue;
 			if (mute)
-				spec->mute_bits |= (1ULL << nid);
+				spec->mute_bits |= (1ULL << mute_nid);
 			else
-				spec->mute_bits &= ~(1ULL << nid);
+				spec->mute_bits &= ~(1ULL << mute_nid);
 			set_pin_eapd(codec, nid, !mute);
 			continue;
 		}
@@ -3786,14 +3823,19 @@ static void do_automute(struct hda_codec *codec, int num_pins, hda_nid_t *pins,
 void snd_hda_gen_update_outputs(struct hda_codec *codec)
 {
 	struct hda_gen_spec *spec = codec->spec;
+	int *paths;
 	int on;
 
 	/* Control HP pins/amps depending on master_mute state;
 	 * in general, HP pins/amps control should be enabled in all cases,
 	 * but currently set only for master_mute, just to be safe
 	 */
+	if (spec->autocfg.line_out_type == AUTO_PIN_HP_OUT)
+		paths = spec->out_paths;
+	else
+		paths = spec->hp_paths;
 	do_automute(codec, ARRAY_SIZE(spec->autocfg.hp_pins),
-		    spec->autocfg.hp_pins, spec->master_mute);
+		    spec->autocfg.hp_pins, paths, spec->master_mute);
 
 	if (!spec->automute_speaker)
 		on = 0;
@@ -3801,8 +3843,12 @@ void snd_hda_gen_update_outputs(struct hda_codec *codec)
 		on = spec->hp_jack_present | spec->line_jack_present;
 	on |= spec->master_mute;
 	spec->speaker_muted = on;
+	if (spec->autocfg.line_out_type == AUTO_PIN_SPEAKER_OUT)
+		paths = spec->out_paths;
+	else
+		paths = spec->speaker_paths;
 	do_automute(codec, ARRAY_SIZE(spec->autocfg.speaker_pins),
-		    spec->autocfg.speaker_pins, on);
+		    spec->autocfg.speaker_pins, paths, on);
 
 	/* toggle line-out mutes if needed, too */
 	/* if LO is a copy of either HP or Speaker, don't need to handle it */
@@ -3815,8 +3861,9 @@ void snd_hda_gen_update_outputs(struct hda_codec *codec)
 		on = spec->hp_jack_present;
 	on |= spec->master_mute;
 	spec->line_out_muted = on;
+	paths = spec->out_paths;
 	do_automute(codec, ARRAY_SIZE(spec->autocfg.line_out_pins),
-		    spec->autocfg.line_out_pins, on);
+		    spec->autocfg.line_out_pins, paths, on);
 }
 EXPORT_SYMBOL_HDA(snd_hda_gen_update_outputs);
 
@@ -3887,7 +3934,7 @@ void snd_hda_gen_mic_autoswitch(struct hda_codec *codec, struct hda_jack_tbl *ja
 		/* don't detect pins retasked as outputs */
 		if (snd_hda_codec_get_pin_target(codec, pin) & AC_PINCTL_OUT_EN)
 			continue;
-		if (snd_hda_jack_detect(codec, pin)) {
+		if (snd_hda_jack_detect_state(codec, pin) == HDA_JACK_PRESENT) {
 			mux_select(codec, 0, spec->am_entry[i].idx);
 			return;
 		}

sound/pci/hda/hda_generic.h

@@ -220,6 +220,7 @@ struct hda_gen_spec {
 	unsigned int hp_mic:1; /* Allow HP as a mic-in */
 	unsigned int suppress_hp_mic_detect:1; /* Don't detect HP/mic */
 	unsigned int no_primary_hp:1; /* Don't prefer HP pins to speaker pins */
+	unsigned int no_multi_io:1; /* Don't try multi I/O config */
 	unsigned int multi_cap_vol:1; /* allow multiple capture xxx volumes */
 	unsigned int inv_dmic_split:1; /* inverted dmic w/a for conexant */
 	unsigned int own_eapd_ctl:1; /* set EAPD by own function */

sound/pci/hda/hda_hwdep.c

@@ -295,7 +295,7 @@ static ssize_t type##_store(struct device *dev,			\
 	struct snd_hwdep *hwdep = dev_get_drvdata(dev);		\
 	struct hda_codec *codec = hwdep->private_data;		\
 	unsigned long val;					\
-	int err = strict_strtoul(buf, 0, &val);			\
+	int err = kstrtoul(buf, 0, &val);			\
 	if (err < 0)						\
 		return err;					\
 	codec->type = val;					\
@@ -654,7 +654,7 @@ int snd_hda_get_int_hint(struct hda_codec *codec, const char *key, int *valp)
 	p = snd_hda_get_hint(codec, key);
 	if (!p)
 		ret = -ENOENT;
-	else if (strict_strtoul(p, 0, &val))
+	else if (kstrtoul(p, 0, &val))
 		ret = -EINVAL;
 	else {
 		*valp = val;
@@ -751,7 +751,7 @@ static void parse_##name##_mode(char *buf, struct hda_bus *bus, \
 				 struct hda_codec **codecp) \
 { \
 	unsigned long val; \
-	if (!strict_strtoul(buf, 0, &val)) \
+	if (!kstrtoul(buf, 0, &val)) \
 		(*codecp)->name = val; \
 }
 

sound/pci/hda/hda_intel.c

@@ -1160,7 +1160,7 @@ static int azx_reset(struct azx *chip, int full_reset)
 		goto __skip;
 
 	/* clear STATESTS */
-	azx_writeb(chip, STATESTS, STATESTS_INT_MASK);
+	azx_writew(chip, STATESTS, STATESTS_INT_MASK);
 
 	/* reset controller */
 	azx_enter_link_reset(chip);
@@ -1242,7 +1242,7 @@ static void azx_int_clear(struct azx *chip)
 	}
 
 	/* clear STATESTS */
-	azx_writeb(chip, STATESTS, STATESTS_INT_MASK);
+	azx_writew(chip, STATESTS, STATESTS_INT_MASK);
 
 	/* clear rirb status */
 	azx_writeb(chip, RIRBSTS, RIRB_INT_MASK);
@@ -1451,8 +1451,8 @@ static irqreturn_t azx_interrupt(int irq, void *dev_id)
 
 #if 0
 	/* clear state status int */
-	if (azx_readb(chip, STATESTS) & 0x04)
-		azx_writeb(chip, STATESTS, 0x04);
+	if (azx_readw(chip, STATESTS) & 0x04)
+		azx_writew(chip, STATESTS, 0x04);
 #endif
 	spin_unlock(&chip->reg_lock);
 	
@@ -2971,6 +2971,10 @@ static int azx_runtime_suspend(struct device *dev)
 	struct snd_card *card = dev_get_drvdata(dev);
 	struct azx *chip = card->private_data;
 
+	/* enable controller wake up event */
+	azx_writew(chip, WAKEEN, azx_readw(chip, WAKEEN) |
+		  STATESTS_INT_MASK);
+
 	azx_stop_chip(chip);
 	azx_enter_link_reset(chip);
 	azx_clear_irq_pending(chip);
@@ -2983,11 +2987,31 @@ static int azx_runtime_resume(struct device *dev)
 {
 	struct snd_card *card = dev_get_drvdata(dev);
 	struct azx *chip = card->private_data;
+	struct hda_bus *bus;
+	struct hda_codec *codec;
+	int status;
 
 	if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL)
 		hda_display_power(true);
+
+	/* Read STATESTS before controller reset */
+	status = azx_readw(chip, STATESTS);
+
 	azx_init_pci(chip);
 	azx_init_chip(chip, 1);
+
+	bus = chip->bus;
+	if (status && bus) {
+		list_for_each_entry(codec, &bus->codec_list, list)
+			if (status & (1 << codec->addr))
+				queue_delayed_work(codec->bus->workq,
+						   &codec->jackpoll_work, codec->jackpoll_interval);
+	}
+
+	/* disable controller Wake Up event*/
+	azx_writew(chip, WAKEEN, azx_readw(chip, WAKEEN) &
+			~STATESTS_INT_MASK);
+
 	return 0;
 }
 
@@ -3831,11 +3855,13 @@ static int azx_probe_continue(struct azx *chip)
 
 	/* Request power well for Haswell HDA controller and codec */
 	if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) {
+#ifdef CONFIG_SND_HDA_I915
 		err = hda_i915_init();
 		if (err < 0) {
 			snd_printk(KERN_ERR SFX "Error request power-well from i915\n");
 			goto out_free;
 		}
+#endif
 		hda_display_power(true);
 	}
 

sound/pci/hda/hda_jack.c

@@ -194,18 +194,24 @@ u32 snd_hda_pin_sense(struct hda_codec *codec, hda_nid_t nid)
 EXPORT_SYMBOL_HDA(snd_hda_pin_sense);
 
 /**
- * snd_hda_jack_detect - query pin Presence Detect status
+ * snd_hda_jack_detect_state - query pin Presence Detect status
  * @codec: the CODEC to sense
  * @nid: the pin NID to sense
  *
- * Query and return the pin's Presence Detect status.
+ * Query and return the pin's Presence Detect status, as either
+ * HDA_JACK_NOT_PRESENT, HDA_JACK_PRESENT or HDA_JACK_PHANTOM.
  */
-int snd_hda_jack_detect(struct hda_codec *codec, hda_nid_t nid)
+int snd_hda_jack_detect_state(struct hda_codec *codec, hda_nid_t nid)
 {
-	u32 sense = snd_hda_pin_sense(codec, nid);
-	return get_jack_plug_state(sense);
+	struct hda_jack_tbl *jack = snd_hda_jack_tbl_get(codec, nid);
+	if (jack && jack->phantom_jack)
+		return HDA_JACK_PHANTOM;
+	else if (snd_hda_pin_sense(codec, nid) & AC_PINSENSE_PRESENCE)
+		return HDA_JACK_PRESENT;
+	else
+		return HDA_JACK_NOT_PRESENT;
 }
-EXPORT_SYMBOL_HDA(snd_hda_jack_detect);
+EXPORT_SYMBOL_HDA(snd_hda_jack_detect_state);
 
 /**
  * snd_hda_jack_detect_enable - enable the jack-detection
@@ -247,8 +253,8 @@ EXPORT_SYMBOL_HDA(snd_hda_jack_detect_enable);
 int snd_hda_jack_set_gating_jack(struct hda_codec *codec, hda_nid_t gated_nid,
 				 hda_nid_t gating_nid)
 {
-	struct hda_jack_tbl *gated = snd_hda_jack_tbl_get(codec, gated_nid);
-	struct hda_jack_tbl *gating = snd_hda_jack_tbl_get(codec, gating_nid);
+	struct hda_jack_tbl *gated = snd_hda_jack_tbl_new(codec, gated_nid);
+	struct hda_jack_tbl *gating = snd_hda_jack_tbl_new(codec, gating_nid);
 
 	if (!gated || !gating)
 		return -EINVAL;

sound/pci/hda/hda_jack.h

@@ -75,7 +75,18 @@ int snd_hda_jack_set_gating_jack(struct hda_codec *codec, hda_nid_t gated_nid,
 				 hda_nid_t gating_nid);
 
 u32 snd_hda_pin_sense(struct hda_codec *codec, hda_nid_t nid);
-int snd_hda_jack_detect(struct hda_codec *codec, hda_nid_t nid);
+
+/* the jack state returned from snd_hda_jack_detect_state() */
+enum {
+	HDA_JACK_NOT_PRESENT, HDA_JACK_PRESENT, HDA_JACK_PHANTOM,
+};
+
+int snd_hda_jack_detect_state(struct hda_codec *codec, hda_nid_t nid);
+
+static inline bool snd_hda_jack_detect(struct hda_codec *codec, hda_nid_t nid)
+{
+	return snd_hda_jack_detect_state(codec, nid) != HDA_JACK_NOT_PRESENT;
+}
 
 bool is_jack_detectable(struct hda_codec *codec, hda_nid_t nid);
 

sound/pci/hda/hda_proc.c

@@ -582,6 +582,36 @@ static void print_gpio(struct snd_info_buffer *buffer,
 	print_nid_array(buffer, codec, nid, &codec->nids);
 }
 
+static void print_device_list(struct snd_info_buffer *buffer,
+			    struct hda_codec *codec, hda_nid_t nid)
+{
+	int i, curr = -1;
+	u8 dev_list[AC_MAX_DEV_LIST_LEN];
+	int devlist_len;
+
+	devlist_len = snd_hda_get_devices(codec, nid, dev_list,
+					AC_MAX_DEV_LIST_LEN);
+	snd_iprintf(buffer, "  Devices: %d\n", devlist_len);
+	if (devlist_len <= 0)
+		return;
+
+	curr = snd_hda_codec_read(codec, nid, 0,
+				AC_VERB_GET_DEVICE_SEL, 0);
+
+	for (i = 0; i < devlist_len; i++) {
+		if (i == curr)
+			snd_iprintf(buffer, "    *");
+		else
+			snd_iprintf(buffer, "     ");
+
+		snd_iprintf(buffer,
+			"Dev %02d: PD = %d, ELDV = %d, IA = %d\n", i,
+			!!(dev_list[i] & AC_DE_PD),
+			!!(dev_list[i] & AC_DE_ELDV),
+			!!(dev_list[i] & AC_DE_IA));
+	}
+}
+
 static void print_codec_info(struct snd_info_entry *entry,
 			     struct snd_info_buffer *buffer)
 {
@@ -751,6 +781,9 @@ static void print_codec_info(struct snd_info_entry *entry,
 				    (wid_caps & AC_WCAP_DELAY) >>
 				    AC_WCAP_DELAY_SHIFT);
 
+		if (wid_type == AC_WID_PIN && codec->dp_mst)
+			print_device_list(buffer, codec, nid);
+
 		if (wid_caps & AC_WCAP_CONN_LIST)
 			print_conn_list(buffer, codec, nid, wid_type,
 					conn, conn_len);

sound/pci/hda/patch_analog.c

@@ -32,7 +32,6 @@
 #include "hda_jack.h"
 #include "hda_generic.h"
 
-#define ENABLE_AD_STATIC_QUIRKS
 
 struct ad198x_spec {
 	struct hda_gen_spec gen;
@@ -43,114 +42,8 @@ struct ad198x_spec {
 	hda_nid_t eapd_nid;
 
 	unsigned int beep_amp;	/* beep amp value, set via set_beep_amp() */
-
-#ifdef ENABLE_AD_STATIC_QUIRKS
-	const struct snd_kcontrol_new *mixers[6];
-	int num_mixers;
-	const struct hda_verb *init_verbs[6];	/* initialization verbs
-						 * don't forget NULL termination!
-						 */
-	unsigned int num_init_verbs;
-
-	/* playback */
-	struct hda_multi_out multiout;	/* playback set-up
-					 * max_channels, dacs must be set
-					 * dig_out_nid and hp_nid are optional
-					 */
-	unsigned int cur_eapd;
-	unsigned int need_dac_fix;
-
-	/* capture */
-	unsigned int num_adc_nids;
-	const hda_nid_t *adc_nids;
-	hda_nid_t dig_in_nid;		/* digital-in NID; optional */
-
-	/* capture source */
-	const struct hda_input_mux *input_mux;
-	const hda_nid_t *capsrc_nids;
-	unsigned int cur_mux[3];
-
-	/* channel model */
-	const struct hda_channel_mode *channel_mode;
-	int num_channel_mode;
-
-	/* PCM information */
-	struct hda_pcm pcm_rec[3];	/* used in alc_build_pcms() */
-
-	unsigned int spdif_route;
-
-	unsigned int jack_present: 1;
-	unsigned int inv_jack_detect: 1;/* inverted jack-detection */
-	unsigned int analog_beep: 1;	/* analog beep input present */
-	unsigned int avoid_init_slave_vol:1;
-
-#ifdef CONFIG_PM
-	struct hda_loopback_check loopback;
-#endif
-	/* for virtual master */
-	hda_nid_t vmaster_nid;
-	const char * const *slave_vols;
-	const char * const *slave_sws;
-#endif /* ENABLE_AD_STATIC_QUIRKS */
-};
-
-#ifdef ENABLE_AD_STATIC_QUIRKS
-/*
- * input MUX handling (common part)
- */
-static int ad198x_mux_enum_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
-{
-	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
-	struct ad198x_spec *spec = codec->spec;
-
-	return snd_hda_input_mux_info(spec->input_mux, uinfo);
-}
-
-static int ad198x_mux_enum_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
-{
-	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
-	struct ad198x_spec *spec = codec->spec;
-	unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
-
-	ucontrol->value.enumerated.item[0] = spec->cur_mux[adc_idx];
-	return 0;
-}
-
-static int ad198x_mux_enum_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
-{
-	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
-	struct ad198x_spec *spec = codec->spec;
-	unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
-
-	return snd_hda_input_mux_put(codec, spec->input_mux, ucontrol,
-				     spec->capsrc_nids[adc_idx],
-				     &spec->cur_mux[adc_idx]);
-}
-
-/*
- * initialization (common callbacks)
- */
-static int ad198x_init(struct hda_codec *codec)
-{
-	struct ad198x_spec *spec = codec->spec;
-	int i;
-
-	for (i = 0; i < spec->num_init_verbs; i++)
-		snd_hda_sequence_write(codec, spec->init_verbs[i]);
-	return 0;
-}
-
-static const char * const ad_slave_pfxs[] = {
-	"Front", "Surround", "Center", "LFE", "Side",
-	"Headphone", "Mono", "Speaker", "IEC958",
-	NULL
 };
 
-static const char * const ad1988_6stack_fp_slave_pfxs[] = {
-	"Front", "Surround", "Center", "LFE", "Side", "IEC958",
-	NULL
-};
-#endif /* ENABLE_AD_STATIC_QUIRKS */
 
 #ifdef CONFIG_SND_HDA_INPUT_BEEP
 /* additional beep mixers; the actual parameters are overwritten at build */
@@ -160,12 +53,6 @@ static const struct snd_kcontrol_new ad_beep_mixer[] = {
 	{ } /* end */
 };
 
-static const struct snd_kcontrol_new ad_beep2_mixer[] = {
-	HDA_CODEC_VOLUME("Digital Beep Playback Volume", 0, 0, HDA_OUTPUT),
-	HDA_CODEC_MUTE_BEEP("Digital Beep Playback Switch", 0, 0, HDA_OUTPUT),
-	{ } /* end */
-};
-
 #define set_beep_amp(spec, nid, idx, dir) \
 	((spec)->beep_amp = HDA_COMPOSE_AMP_VAL(nid, 1, idx, dir)) /* mono */
 #else
@@ -181,8 +68,7 @@ static int create_beep_ctls(struct hda_codec *codec)
 	if (!spec->beep_amp)
 		return 0;
 
-	knew = spec->analog_beep ? ad_beep2_mixer : ad_beep_mixer;
-	for ( ; knew->name; knew++) {
+	for (knew = ad_beep_mixer ; knew->name; knew++) {
 		int err;
 		struct snd_kcontrol *kctl;
 		kctl = snd_ctl_new1(knew, codec);
@@ -199,268 +85,6 @@ static int create_beep_ctls(struct hda_codec *codec)
 #define create_beep_ctls(codec)		0
 #endif
 
-#ifdef ENABLE_AD_STATIC_QUIRKS
-static int ad198x_build_controls(struct hda_codec *codec)
-{
-	struct ad198x_spec *spec = codec->spec;
-	struct snd_kcontrol *kctl;
-	unsigned int i;
-	int err;
-
-	for (i = 0; i < spec->num_mixers; i++) {
-		err = snd_hda_add_new_ctls(codec, spec->mixers[i]);
-		if (err < 0)
-			return err;
-	}
-	if (spec->multiout.dig_out_nid) {
-		err = snd_hda_create_spdif_out_ctls(codec,
-						    spec->multiout.dig_out_nid,
-						    spec->multiout.dig_out_nid);
-		if (err < 0)
-			return err;
-		err = snd_hda_create_spdif_share_sw(codec,
-						    &spec->multiout);
-		if (err < 0)
-			return err;
-		spec->multiout.share_spdif = 1;
-	} 
-	if (spec->dig_in_nid) {
-		err = snd_hda_create_spdif_in_ctls(codec, spec->dig_in_nid);
-		if (err < 0)
-			return err;
-	}
-
-	/* create beep controls if needed */
-	err = create_beep_ctls(codec);
-	if (err < 0)
-		return err;
-
-	/* if we have no master control, let's create it */
-	if (!snd_hda_find_mixer_ctl(codec, "Master Playback Volume")) {
-		unsigned int vmaster_tlv[4];
-		snd_hda_set_vmaster_tlv(codec, spec->vmaster_nid,
-					HDA_OUTPUT, vmaster_tlv);
-		err = __snd_hda_add_vmaster(codec, "Master Playback Volume",
-					  vmaster_tlv,
-					  (spec->slave_vols ?
-					   spec->slave_vols : ad_slave_pfxs),
-					  "Playback Volume",
-					  !spec->avoid_init_slave_vol, NULL);
-		if (err < 0)
-			return err;
-	}
-	if (!snd_hda_find_mixer_ctl(codec, "Master Playback Switch")) {
-		err = snd_hda_add_vmaster(codec, "Master Playback Switch",
-					  NULL,
-					  (spec->slave_sws ?
-					   spec->slave_sws : ad_slave_pfxs),
-					  "Playback Switch");
-		if (err < 0)
-			return err;
-	}
-
-	/* assign Capture Source enums to NID */
-	kctl = snd_hda_find_mixer_ctl(codec, "Capture Source");
-	if (!kctl)
-		kctl = snd_hda_find_mixer_ctl(codec, "Input Source");
-	for (i = 0; kctl && i < kctl->count; i++) {
-		err = snd_hda_add_nid(codec, kctl, i, spec->capsrc_nids[i]);
-		if (err < 0)
-			return err;
-	}
-
-	/* assign IEC958 enums to NID */
-	kctl = snd_hda_find_mixer_ctl(codec,
-			SNDRV_CTL_NAME_IEC958("",PLAYBACK,NONE) "Source");
-	if (kctl) {
-		err = snd_hda_add_nid(codec, kctl, 0,
-				      spec->multiout.dig_out_nid);
-		if (err < 0)
-			return err;
-	}
-
-	return 0;
-}
-
-#ifdef CONFIG_PM
-static int ad198x_check_power_status(struct hda_codec *codec, hda_nid_t nid)
-{
-	struct ad198x_spec *spec = codec->spec;
-	return snd_hda_check_amp_list_power(codec, &spec->loopback, nid);
-}
-#endif
-
-/*
- * Analog playback callbacks
- */
-static int ad198x_playback_pcm_open(struct hda_pcm_stream *hinfo,
-				    struct hda_codec *codec,
-				    struct snd_pcm_substream *substream)
-{
-	struct ad198x_spec *spec = codec->spec;
-	return snd_hda_multi_out_analog_open(codec, &spec->multiout, substream,
-					     hinfo);
-}
-
-static int ad198x_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
-				       struct hda_codec *codec,
-				       unsigned int stream_tag,
-				       unsigned int format,
-				       struct snd_pcm_substream *substream)
-{
-	struct ad198x_spec *spec = codec->spec;
-	return snd_hda_multi_out_analog_prepare(codec, &spec->multiout, stream_tag,
-						format, substream);
-}
-
-static int ad198x_playback_pcm_cleanup(struct hda_pcm_stream *hinfo,
-				       struct hda_codec *codec,
-				       struct snd_pcm_substream *substream)
-{
-	struct ad198x_spec *spec = codec->spec;
-	return snd_hda_multi_out_analog_cleanup(codec, &spec->multiout);
-}
-
-/*
- * Digital out
- */
-static int ad198x_dig_playback_pcm_open(struct hda_pcm_stream *hinfo,
-					struct hda_codec *codec,
-					struct snd_pcm_substream *substream)
-{
-	struct ad198x_spec *spec = codec->spec;
-	return snd_hda_multi_out_dig_open(codec, &spec->multiout);
-}
-
-static int ad198x_dig_playback_pcm_close(struct hda_pcm_stream *hinfo,
-					 struct hda_codec *codec,
-					 struct snd_pcm_substream *substream)
-{
-	struct ad198x_spec *spec = codec->spec;
-	return snd_hda_multi_out_dig_close(codec, &spec->multiout);
-}
-
-static int ad198x_dig_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
-					   struct hda_codec *codec,
-					   unsigned int stream_tag,
-					   unsigned int format,
-					   struct snd_pcm_substream *substream)
-{
-	struct ad198x_spec *spec = codec->spec;
-	return snd_hda_multi_out_dig_prepare(codec, &spec->multiout, stream_tag,
-					     format, substream);
-}
-
-static int ad198x_dig_playback_pcm_cleanup(struct hda_pcm_stream *hinfo,
-					   struct hda_codec *codec,
-					   struct snd_pcm_substream *substream)
-{
-	struct ad198x_spec *spec = codec->spec;
-	return snd_hda_multi_out_dig_cleanup(codec, &spec->multiout);
-}
-
-/*
- * Analog capture
- */
-static int ad198x_capture_pcm_prepare(struct hda_pcm_stream *hinfo,
-				      struct hda_codec *codec,
-				      unsigned int stream_tag,
-				      unsigned int format,
-				      struct snd_pcm_substream *substream)
-{
-	struct ad198x_spec *spec = codec->spec;
-	snd_hda_codec_setup_stream(codec, spec->adc_nids[substream->number],
-				   stream_tag, 0, format);
-	return 0;
-}
-
-static int ad198x_capture_pcm_cleanup(struct hda_pcm_stream *hinfo,
-				      struct hda_codec *codec,
-				      struct snd_pcm_substream *substream)
-{
-	struct ad198x_spec *spec = codec->spec;
-	snd_hda_codec_cleanup_stream(codec, spec->adc_nids[substream->number]);
-	return 0;
-}
-
-/*
- */
-static const struct hda_pcm_stream ad198x_pcm_analog_playback = {
-	.substreams = 1,
-	.channels_min = 2,
-	.channels_max = 6, /* changed later */
-	.nid = 0, /* fill later */
-	.ops = {
-		.open = ad198x_playback_pcm_open,
-		.prepare = ad198x_playback_pcm_prepare,
-		.cleanup = ad198x_playback_pcm_cleanup,
-	},
-};
-
-static const struct hda_pcm_stream ad198x_pcm_analog_capture = {
-	.substreams = 1,
-	.channels_min = 2,
-	.channels_max = 2,
-	.nid = 0, /* fill later */
-	.ops = {
-		.prepare = ad198x_capture_pcm_prepare,
-		.cleanup = ad198x_capture_pcm_cleanup
-	},
-};
-
-static const struct hda_pcm_stream ad198x_pcm_digital_playback = {
-	.substreams = 1,
-	.channels_min = 2,
-	.channels_max = 2,
-	.nid = 0, /* fill later */
-	.ops = {
-		.open = ad198x_dig_playback_pcm_open,
-		.close = ad198x_dig_playback_pcm_close,
-		.prepare = ad198x_dig_playback_pcm_prepare,
-		.cleanup = ad198x_dig_playback_pcm_cleanup
-	},
-};
-
-static const struct hda_pcm_stream ad198x_pcm_digital_capture = {
-	.substreams = 1,
-	.channels_min = 2,
-	.channels_max = 2,
-	/* NID is set in alc_build_pcms */
-};
-
-static int ad198x_build_pcms(struct hda_codec *codec)
-{
-	struct ad198x_spec *spec = codec->spec;
-	struct hda_pcm *info = spec->pcm_rec;
-
-	codec->num_pcms = 1;
-	codec->pcm_info = info;
-
-	info->name = "AD198x Analog";
-	info->stream[SNDRV_PCM_STREAM_PLAYBACK] = ad198x_pcm_analog_playback;
-	info->stream[SNDRV_PCM_STREAM_PLAYBACK].channels_max = spec->multiout.max_channels;
-	info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->multiout.dac_nids[0];
-	info->stream[SNDRV_PCM_STREAM_CAPTURE] = ad198x_pcm_analog_capture;
-	info->stream[SNDRV_PCM_STREAM_CAPTURE].substreams = spec->num_adc_nids;
-	info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->adc_nids[0];
-
-	if (spec->multiout.dig_out_nid) {
-		info++;
-		codec->num_pcms++;
-		codec->spdif_status_reset = 1;
-		info->name = "AD198x Digital";
-		info->pcm_type = HDA_PCM_TYPE_SPDIF;
-		info->stream[SNDRV_PCM_STREAM_PLAYBACK] = ad198x_pcm_digital_playback;
-		info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->multiout.dig_out_nid;
-		if (spec->dig_in_nid) {
-			info->stream[SNDRV_PCM_STREAM_CAPTURE] = ad198x_pcm_digital_capture;
-			info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->dig_in_nid;
-		}
-	}
-
-	return 0;
-}
-#endif /* ENABLE_AD_STATIC_QUIRKS */
 
 static void ad198x_power_eapd_write(struct hda_codec *codec, hda_nid_t front,
 				hda_nid_t hp)
@@ -507,18 +131,6 @@ static void ad198x_shutup(struct hda_codec *codec)
 	ad198x_power_eapd(codec);
 }
 
-static void ad198x_free(struct hda_codec *codec)
-{
-	struct ad198x_spec *spec = codec->spec;
-
-	if (!spec)
-		return;
-
-	snd_hda_gen_spec_free(&spec->gen);
-	kfree(spec);
-	snd_hda_detach_beep_device(codec);
-}
-
 #ifdef CONFIG_PM
 static int ad198x_suspend(struct hda_codec *codec)
 {
@@ -527,65 +139,6 @@ static int ad198x_suspend(struct hda_codec *codec)
 }
 #endif
 
-#ifdef ENABLE_AD_STATIC_QUIRKS
-static const struct hda_codec_ops ad198x_patch_ops = {
-	.build_controls = ad198x_build_controls,
-	.build_pcms = ad198x_build_pcms,
-	.init = ad198x_init,
-	.free = ad198x_free,
-#ifdef CONFIG_PM
-	.check_power_status = ad198x_check_power_status,
-	.suspend = ad198x_suspend,
-#endif
-	.reboot_notify = ad198x_shutup,
-};
-
-
-/*
- * EAPD control
- * the private value = nid
- */
-#define ad198x_eapd_info	snd_ctl_boolean_mono_info
-
-static int ad198x_eapd_get(struct snd_kcontrol *kcontrol,
-			   struct snd_ctl_elem_value *ucontrol)
-{
-	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
-	struct ad198x_spec *spec = codec->spec;
-	if (codec->inv_eapd)
-		ucontrol->value.integer.value[0] = ! spec->cur_eapd;
-	else
-		ucontrol->value.integer.value[0] = spec->cur_eapd;
-	return 0;
-}
-
-static int ad198x_eapd_put(struct snd_kcontrol *kcontrol,
-			   struct snd_ctl_elem_value *ucontrol)
-{
-	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
-	struct ad198x_spec *spec = codec->spec;
-	hda_nid_t nid = kcontrol->private_value & 0xff;
-	unsigned int eapd;
-	eapd = !!ucontrol->value.integer.value[0];
-	if (codec->inv_eapd)
-		eapd = !eapd;
-	if (eapd == spec->cur_eapd)
-		return 0;
-	spec->cur_eapd = eapd;
-	snd_hda_codec_write_cache(codec, nid,
-				  0, AC_VERB_SET_EAPD_BTLENABLE,
-				  eapd ? 0x02 : 0x00);
-	return 1;
-}
-
-static int ad198x_ch_mode_info(struct snd_kcontrol *kcontrol,
-			       struct snd_ctl_elem_info *uinfo);
-static int ad198x_ch_mode_get(struct snd_kcontrol *kcontrol,
-			      struct snd_ctl_elem_value *ucontrol);
-static int ad198x_ch_mode_put(struct snd_kcontrol *kcontrol,
-			      struct snd_ctl_elem_value *ucontrol);
-#endif /* ENABLE_AD_STATIC_QUIRKS */
-
 
 /*
  * Automatic parse of I/O pins from the BIOS configuration
@@ -646,580 +199,124 @@ static int ad198x_parse_auto_config(struct hda_codec *codec)
  * AD1986A specific
  */
 
-#ifdef ENABLE_AD_STATIC_QUIRKS
-#define AD1986A_SPDIF_OUT	0x02
-#define AD1986A_FRONT_DAC	0x03
-#define AD1986A_SURR_DAC	0x04
-#define AD1986A_CLFE_DAC	0x05
-#define AD1986A_ADC		0x06
-
-static const hda_nid_t ad1986a_dac_nids[3] = {
-	AD1986A_FRONT_DAC, AD1986A_SURR_DAC, AD1986A_CLFE_DAC
-};
-static const hda_nid_t ad1986a_adc_nids[1] = { AD1986A_ADC };
-static const hda_nid_t ad1986a_capsrc_nids[1] = { 0x12 };
-
-static const struct hda_input_mux ad1986a_capture_source = {
-	.num_items = 7,
-	.items = {
-		{ "Mic", 0x0 },
-		{ "CD", 0x1 },
-		{ "Aux", 0x3 },
-		{ "Line", 0x4 },
-		{ "Mix", 0x5 },
-		{ "Mono", 0x6 },
-		{ "Phone", 0x7 },
-	},
-};
-
-
-static const struct hda_bind_ctls ad1986a_bind_pcm_vol = {
-	.ops = &snd_hda_bind_vol,
-	.values = {
-		HDA_COMPOSE_AMP_VAL(AD1986A_FRONT_DAC, 3, 0, HDA_OUTPUT),
-		HDA_COMPOSE_AMP_VAL(AD1986A_SURR_DAC, 3, 0, HDA_OUTPUT),
-		HDA_COMPOSE_AMP_VAL(AD1986A_CLFE_DAC, 3, 0, HDA_OUTPUT),
-		0
-	},
-};
+static int alloc_ad_spec(struct hda_codec *codec)
+{
+	struct ad198x_spec *spec;
 
-static const struct hda_bind_ctls ad1986a_bind_pcm_sw = {
-	.ops = &snd_hda_bind_sw,
-	.values = {
-		HDA_COMPOSE_AMP_VAL(AD1986A_FRONT_DAC, 3, 0, HDA_OUTPUT),
-		HDA_COMPOSE_AMP_VAL(AD1986A_SURR_DAC, 3, 0, HDA_OUTPUT),
-		HDA_COMPOSE_AMP_VAL(AD1986A_CLFE_DAC, 3, 0, HDA_OUTPUT),
-		0
-	},
-};
+	spec = kzalloc(sizeof(*spec), GFP_KERNEL);
+	if (!spec)
+		return -ENOMEM;
+	codec->spec = spec;
+	snd_hda_gen_spec_init(&spec->gen);
+	return 0;
+}
 
 /*
- * mixers
+ * AD1986A fixup codes
  */
-static const struct snd_kcontrol_new ad1986a_mixers[] = {
-	/*
-	 * bind volumes/mutes of 3 DACs as a single PCM control for simplicity
-	 */
-	HDA_BIND_VOL("PCM Playback Volume", &ad1986a_bind_pcm_vol),
-	HDA_BIND_SW("PCM Playback Switch", &ad1986a_bind_pcm_sw),
-	HDA_CODEC_VOLUME("Front Playback Volume", 0x1b, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Front Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Surround Playback Volume", 0x1c, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Surround Playback Switch", 0x1c, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x1d, 1, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x1d, 2, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x1d, 1, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x1d, 2, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Headphone Playback Volume", 0x1a, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Headphone Playback Switch", 0x1a, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("CD Playback Volume", 0x15, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("CD Playback Switch", 0x15, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Line Playback Volume", 0x17, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Line Playback Switch", 0x17, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Aux Playback Volume", 0x16, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Aux Playback Switch", 0x16, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Mic Playback Volume", 0x13, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Mic Playback Switch", 0x13, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Mic Boost Volume", 0x0f, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Mono Playback Volume", 0x1e, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Mono Playback Switch", 0x1e, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Capture Volume", 0x12, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Capture Switch", 0x12, 0x0, HDA_OUTPUT),
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		.name = "Capture Source",
-		.info = ad198x_mux_enum_info,
-		.get = ad198x_mux_enum_get,
-		.put = ad198x_mux_enum_put,
-	},
-	HDA_CODEC_MUTE("Stereo Downmix Switch", 0x09, 0x0, HDA_OUTPUT),
-	{ } /* end */
-};
-
-/* additional mixers for 3stack mode */
-static const struct snd_kcontrol_new ad1986a_3st_mixers[] = {
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		.name = "Channel Mode",
-		.info = ad198x_ch_mode_info,
-		.get = ad198x_ch_mode_get,
-		.put = ad198x_ch_mode_put,
-	},
-	{ } /* end */
-};
 
-/* laptop model - 2ch only */
-static const hda_nid_t ad1986a_laptop_dac_nids[1] = { AD1986A_FRONT_DAC };
+/* Lenovo N100 seems to report the reversed bit for HP jack-sensing */
+static void ad_fixup_inv_jack_detect(struct hda_codec *codec,
+				     const struct hda_fixup *fix, int action)
+{
+	if (action == HDA_FIXUP_ACT_PRE_PROBE)
+		codec->inv_jack_detect = 1;
+}
 
-/* master controls both pins 0x1a and 0x1b */
-static const struct hda_bind_ctls ad1986a_laptop_master_vol = {
-	.ops = &snd_hda_bind_vol,
-	.values = {
-		HDA_COMPOSE_AMP_VAL(0x1a, 3, 0, HDA_OUTPUT),
-		HDA_COMPOSE_AMP_VAL(0x1b, 3, 0, HDA_OUTPUT),
-		0,
-	},
+enum {
+	AD1986A_FIXUP_INV_JACK_DETECT,
+	AD1986A_FIXUP_ULTRA,
+	AD1986A_FIXUP_SAMSUNG,
+	AD1986A_FIXUP_3STACK,
+	AD1986A_FIXUP_LAPTOP,
+	AD1986A_FIXUP_LAPTOP_IMIC,
 };
 
-static const struct hda_bind_ctls ad1986a_laptop_master_sw = {
-	.ops = &snd_hda_bind_sw,
-	.values = {
-		HDA_COMPOSE_AMP_VAL(0x1a, 3, 0, HDA_OUTPUT),
-		HDA_COMPOSE_AMP_VAL(0x1b, 3, 0, HDA_OUTPUT),
-		0,
+static const struct hda_fixup ad1986a_fixups[] = {
+	[AD1986A_FIXUP_INV_JACK_DETECT] = {
+		.type = HDA_FIXUP_FUNC,
+		.v.func = ad_fixup_inv_jack_detect,
 	},
-};
-
-static const struct snd_kcontrol_new ad1986a_laptop_mixers[] = {
-	HDA_CODEC_VOLUME("PCM Playback Volume", 0x03, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("PCM Playback Switch", 0x03, 0x0, HDA_OUTPUT),
-	HDA_BIND_VOL("Master Playback Volume", &ad1986a_laptop_master_vol),
-	HDA_BIND_SW("Master Playback Switch", &ad1986a_laptop_master_sw),
-	HDA_CODEC_VOLUME("CD Playback Volume", 0x15, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("CD Playback Switch", 0x15, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Line Playback Volume", 0x17, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Line Playback Switch", 0x17, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Aux Playback Volume", 0x16, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Aux Playback Switch", 0x16, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Mic Playback Volume", 0x13, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Mic Playback Switch", 0x13, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Mic Boost Volume", 0x0f, 0x0, HDA_OUTPUT),
-	/* 
-	   HDA_CODEC_VOLUME("Mono Playback Volume", 0x1e, 0x0, HDA_OUTPUT),
-	   HDA_CODEC_MUTE("Mono Playback Switch", 0x1e, 0x0, HDA_OUTPUT), */
-	HDA_CODEC_VOLUME("Capture Volume", 0x12, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Capture Switch", 0x12, 0x0, HDA_OUTPUT),
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		.name = "Capture Source",
-		.info = ad198x_mux_enum_info,
-		.get = ad198x_mux_enum_get,
-		.put = ad198x_mux_enum_put,
+	[AD1986A_FIXUP_ULTRA] = {
+		.type = HDA_FIXUP_PINS,
+		.v.pins = (const struct hda_pintbl[]) {
+			{ 0x1b, 0x90170110 }, /* speaker */
+			{ 0x1d, 0x90a7013e }, /* int mic */
+			{}
+		},
 	},
-	{ } /* end */
-};
-
-/* laptop-eapd model - 2ch only */
-
-static const struct hda_input_mux ad1986a_laptop_eapd_capture_source = {
-	.num_items = 3,
-	.items = {
-		{ "Mic", 0x0 },
-		{ "Internal Mic", 0x4 },
-		{ "Mix", 0x5 },
+	[AD1986A_FIXUP_SAMSUNG] = {
+		.type = HDA_FIXUP_PINS,
+		.v.pins = (const struct hda_pintbl[]) {
+			{ 0x1b, 0x90170110 }, /* speaker */
+			{ 0x1d, 0x90a7013e }, /* int mic */
+			{ 0x20, 0x411111f0 }, /* N/A */
+			{ 0x24, 0x411111f0 }, /* N/A */
+			{}
+		},
 	},
-};
-
-static const struct hda_input_mux ad1986a_automic_capture_source = {
-	.num_items = 2,
-	.items = {
-		{ "Mic", 0x0 },
-		{ "Mix", 0x5 },
+	[AD1986A_FIXUP_3STACK] = {
+		.type = HDA_FIXUP_PINS,
+		.v.pins = (const struct hda_pintbl[]) {
+			{ 0x1a, 0x02214021 }, /* headphone */
+			{ 0x1b, 0x01014011 }, /* front */
+			{ 0x1c, 0x01013012 }, /* surround */
+			{ 0x1d, 0x01019015 }, /* clfe */
+			{ 0x1e, 0x411111f0 }, /* N/A */
+			{ 0x1f, 0x02a190f0 }, /* mic */
+			{ 0x20, 0x018130f0 }, /* line-in */
+			{}
+		},
 	},
-};
-
-static const struct snd_kcontrol_new ad1986a_laptop_master_mixers[] = {
-	HDA_BIND_VOL("Master Playback Volume", &ad1986a_laptop_master_vol),
-	HDA_BIND_SW("Master Playback Switch", &ad1986a_laptop_master_sw),
-	{ } /* end */
-};
-
-static const struct snd_kcontrol_new ad1986a_laptop_eapd_mixers[] = {
-	HDA_CODEC_VOLUME("PCM Playback Volume", 0x03, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("PCM Playback Switch", 0x03, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Mic Playback Volume", 0x13, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Mic Playback Switch", 0x13, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Mic Boost Volume", 0x0f, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Capture Volume", 0x12, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Capture Switch", 0x12, 0x0, HDA_OUTPUT),
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		.name = "Capture Source",
-		.info = ad198x_mux_enum_info,
-		.get = ad198x_mux_enum_get,
-		.put = ad198x_mux_enum_put,
+	[AD1986A_FIXUP_LAPTOP] = {
+		.type = HDA_FIXUP_PINS,
+		.v.pins = (const struct hda_pintbl[]) {
+			{ 0x1a, 0x02214021 }, /* headphone */
+			{ 0x1b, 0x90170110 }, /* speaker */
+			{ 0x1c, 0x411111f0 }, /* N/A */
+			{ 0x1d, 0x411111f0 }, /* N/A */
+			{ 0x1e, 0x411111f0 }, /* N/A */
+			{ 0x1f, 0x02a191f0 }, /* mic */
+			{ 0x20, 0x411111f0 }, /* N/A */
+			{}
+		},
 	},
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		.name = "External Amplifier",
-		.subdevice = HDA_SUBDEV_NID_FLAG | 0x1b,
-		.info = ad198x_eapd_info,
-		.get = ad198x_eapd_get,
-		.put = ad198x_eapd_put,
-		.private_value = 0x1b, /* port-D */
+	[AD1986A_FIXUP_LAPTOP_IMIC] = {
+		.type = HDA_FIXUP_PINS,
+		.v.pins = (const struct hda_pintbl[]) {
+			{ 0x1d, 0x90a7013e }, /* int mic */
+			{}
+		},
+		.chained_before = 1,
+		.chain_id = AD1986A_FIXUP_LAPTOP,
 	},
-	{ } /* end */
 };
 
-static const struct snd_kcontrol_new ad1986a_laptop_intmic_mixers[] = {
-	HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x17, 0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x17, 0, HDA_OUTPUT),
-	{ } /* end */
+static const struct snd_pci_quirk ad1986a_fixup_tbl[] = {
+	SND_PCI_QUIRK(0x103c, 0x30af, "HP B2800", AD1986A_FIXUP_LAPTOP_IMIC),
+	SND_PCI_QUIRK_MASK(0x1043, 0xff00, 0x8100, "ASUS P5", AD1986A_FIXUP_3STACK),
+	SND_PCI_QUIRK_MASK(0x1043, 0xff00, 0x8200, "ASUS M2", AD1986A_FIXUP_3STACK),
+	SND_PCI_QUIRK(0x10de, 0xcb84, "ASUS A8N-VM", AD1986A_FIXUP_3STACK),
+	SND_PCI_QUIRK(0x144d, 0xc01e, "FSC V2060", AD1986A_FIXUP_LAPTOP),
+	SND_PCI_QUIRK_MASK(0x144d, 0xff00, 0xc000, "Samsung", AD1986A_FIXUP_SAMSUNG),
+	SND_PCI_QUIRK(0x144d, 0xc027, "Samsung Q1", AD1986A_FIXUP_ULTRA),
+	SND_PCI_QUIRK(0x17aa, 0x2066, "Lenovo N100", AD1986A_FIXUP_INV_JACK_DETECT),
+	SND_PCI_QUIRK(0x17aa, 0x1011, "Lenovo M55", AD1986A_FIXUP_3STACK),
+	SND_PCI_QUIRK(0x17aa, 0x1017, "Lenovo A60", AD1986A_FIXUP_3STACK),
+	{}
 };
 
-/* re-connect the mic boost input according to the jack sensing */
-static void ad1986a_automic(struct hda_codec *codec)
-{
-	unsigned int present;
-	present = snd_hda_jack_detect(codec, 0x1f);
-	/* 0 = 0x1f, 2 = 0x1d, 4 = mixed */
-	snd_hda_codec_write(codec, 0x0f, 0, AC_VERB_SET_CONNECT_SEL,
-			    present ? 0 : 2);
-}
-
-#define AD1986A_MIC_EVENT		0x36
-
-static void ad1986a_automic_unsol_event(struct hda_codec *codec,
-					    unsigned int res)
-{
-	if ((res >> 26) != AD1986A_MIC_EVENT)
-		return;
-	ad1986a_automic(codec);
-}
-
-static int ad1986a_automic_init(struct hda_codec *codec)
-{
-	ad198x_init(codec);
-	ad1986a_automic(codec);
-	return 0;
-}
-
-/* laptop-automute - 2ch only */
-
-static void ad1986a_update_hp(struct hda_codec *codec)
-{
-	struct ad198x_spec *spec = codec->spec;
-	unsigned int mute;
-
-	if (spec->jack_present)
-		mute = HDA_AMP_MUTE; /* mute internal speaker */
-	else
-		/* unmute internal speaker if necessary */
-		mute = snd_hda_codec_amp_read(codec, 0x1a, 0, HDA_OUTPUT, 0);
-	snd_hda_codec_amp_stereo(codec, 0x1b, HDA_OUTPUT, 0,
-				 HDA_AMP_MUTE, mute);
-}
-
-static void ad1986a_hp_automute(struct hda_codec *codec)
-{
-	struct ad198x_spec *spec = codec->spec;
-
-	spec->jack_present = snd_hda_jack_detect(codec, 0x1a);
-	if (spec->inv_jack_detect)
-		spec->jack_present = !spec->jack_present;
-	ad1986a_update_hp(codec);
-}
-
-#define AD1986A_HP_EVENT		0x37
-
-static void ad1986a_hp_unsol_event(struct hda_codec *codec, unsigned int res)
-{
-	if ((res >> 26) != AD1986A_HP_EVENT)
-		return;
-	ad1986a_hp_automute(codec);
-}
-
-static int ad1986a_hp_init(struct hda_codec *codec)
-{
-	ad198x_init(codec);
-	ad1986a_hp_automute(codec);
-	return 0;
-}
-
-/* bind hp and internal speaker mute (with plug check) */
-static int ad1986a_hp_master_sw_put(struct snd_kcontrol *kcontrol,
-				    struct snd_ctl_elem_value *ucontrol)
-{
-	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
-	int change = snd_hda_mixer_amp_switch_put(kcontrol, ucontrol);
-	if (change)
-		ad1986a_update_hp(codec);
-	return change;
-}
-
-static const struct snd_kcontrol_new ad1986a_automute_master_mixers[] = {
-	HDA_BIND_VOL("Master Playback Volume", &ad1986a_laptop_master_vol),
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		.name = "Master Playback Switch",
-		.subdevice = HDA_SUBDEV_AMP_FLAG,
-		.info = snd_hda_mixer_amp_switch_info,
-		.get = snd_hda_mixer_amp_switch_get,
-		.put = ad1986a_hp_master_sw_put,
-		.private_value = HDA_COMPOSE_AMP_VAL(0x1a, 3, 0, HDA_OUTPUT),
-	},
-	{ } /* end */
-};
-
-
-/*
- * initialization verbs
- */
-static const struct hda_verb ad1986a_init_verbs[] = {
-	/* Front, Surround, CLFE DAC; mute as default */
-	{0x03, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
-	{0x04, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
-	{0x05, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
-	/* Downmix - off */
-	{0x09, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
-	/* HP, Line-Out, Surround, CLFE selectors */
-	{0x0a, AC_VERB_SET_CONNECT_SEL, 0x0},
-	{0x0b, AC_VERB_SET_CONNECT_SEL, 0x0},
-	{0x0c, AC_VERB_SET_CONNECT_SEL, 0x0},
-	{0x0d, AC_VERB_SET_CONNECT_SEL, 0x0},
-	/* Mono selector */
-	{0x0e, AC_VERB_SET_CONNECT_SEL, 0x0},
-	/* Mic selector: Mic 1/2 pin */
-	{0x0f, AC_VERB_SET_CONNECT_SEL, 0x0},
-	/* Line-in selector: Line-in */
-	{0x10, AC_VERB_SET_CONNECT_SEL, 0x0},
-	/* Mic 1/2 swap */
-	{0x11, AC_VERB_SET_CONNECT_SEL, 0x0},
-	/* Record selector: mic */
-	{0x12, AC_VERB_SET_CONNECT_SEL, 0x0},
-	/* Mic, Phone, CD, Aux, Line-In amp; mute as default */
-	{0x13, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
-	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
-	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
-	{0x16, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
-	{0x17, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
-	/* PC beep */
-	{0x18, AC_VERB_SET_CONNECT_SEL, 0x0},
-	/* HP, Line-Out, Surround, CLFE, Mono pins; mute as default */
-	{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
-	{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
-	{0x1c, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
-	{0x1d, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
-	{0x1e, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
-	/* HP Pin */
-	{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0 },
-	/* Front, Surround, CLFE Pins */
-	{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
-	{0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
-	{0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
-	/* Mono Pin */
-	{0x1e, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
-	/* Mic Pin */
-	{0x1f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
-	/* Line, Aux, CD, Beep-In Pin */
-	{0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
-	{0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
-	{0x22, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
-	{0x23, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
-	{0x24, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
-	{ } /* end */
-};
-
-static const struct hda_verb ad1986a_ch2_init[] = {
-	/* Surround out -> Line In */
-	{ 0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },
- 	/* Line-in selectors */
-	{ 0x10, AC_VERB_SET_CONNECT_SEL, 0x1 },
-	/* CLFE -> Mic in */
-	{ 0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
-	/* Mic selector, mix C/LFE (backmic) and Mic (frontmic) */
-	{ 0x0f, AC_VERB_SET_CONNECT_SEL, 0x4 },
-	{ } /* end */
-};
-
-static const struct hda_verb ad1986a_ch4_init[] = {
-	/* Surround out -> Surround */
-	{ 0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
-	{ 0x10, AC_VERB_SET_CONNECT_SEL, 0x0 },
-	/* CLFE -> Mic in */
-	{ 0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
-	{ 0x0f, AC_VERB_SET_CONNECT_SEL, 0x4 },
-	{ } /* end */
-};
-
-static const struct hda_verb ad1986a_ch6_init[] = {
-	/* Surround out -> Surround out */
-	{ 0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
-	{ 0x10, AC_VERB_SET_CONNECT_SEL, 0x0 },
-	/* CLFE -> CLFE */
-	{ 0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
-	{ 0x0f, AC_VERB_SET_CONNECT_SEL, 0x0 },
-	{ } /* end */
-};
-
-static const struct hda_channel_mode ad1986a_modes[3] = {
-	{ 2, ad1986a_ch2_init },
-	{ 4, ad1986a_ch4_init },
-	{ 6, ad1986a_ch6_init },
-};
-
-/* eapd initialization */
-static const struct hda_verb ad1986a_eapd_init_verbs[] = {
-	{0x1b, AC_VERB_SET_EAPD_BTLENABLE, 0x00 },
-	{}
-};
-
-static const struct hda_verb ad1986a_automic_verbs[] = {
-	{0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
-	{0x1f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
-	/*{0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},*/
-	{0x0f, AC_VERB_SET_CONNECT_SEL, 0x0},
-	{0x1f, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1986A_MIC_EVENT},
-	{}
-};
-
-/* Ultra initialization */
-static const struct hda_verb ad1986a_ultra_init[] = {
-	/* eapd initialization */
-	{ 0x1b, AC_VERB_SET_EAPD_BTLENABLE, 0x00 },
-	/* CLFE -> Mic in */
-	{ 0x0f, AC_VERB_SET_CONNECT_SEL, 0x2 },
-	{ 0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
-	{ 0x1d, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080 },
-	{ } /* end */
-};
-
-/* pin sensing on HP jack */
-static const struct hda_verb ad1986a_hp_init_verbs[] = {
-	{0x1a, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1986A_HP_EVENT},
-	{}
-};
-
-static void ad1986a_samsung_p50_unsol_event(struct hda_codec *codec,
-					    unsigned int res)
-{
-	switch (res >> 26) {
-	case AD1986A_HP_EVENT:
-		ad1986a_hp_automute(codec);
-		break;
-	case AD1986A_MIC_EVENT:
-		ad1986a_automic(codec);
-		break;
-	}
-}
-
-static int ad1986a_samsung_p50_init(struct hda_codec *codec)
-{
-	ad198x_init(codec);
-	ad1986a_hp_automute(codec);
-	ad1986a_automic(codec);
-	return 0;
-}
-
-
-/* models */
-enum {
-	AD1986A_AUTO,
-	AD1986A_6STACK,
-	AD1986A_3STACK,
-	AD1986A_LAPTOP,
-	AD1986A_LAPTOP_EAPD,
-	AD1986A_LAPTOP_AUTOMUTE,
-	AD1986A_ULTRA,
-	AD1986A_SAMSUNG,
-	AD1986A_SAMSUNG_P50,
-	AD1986A_MODELS
-};
-
-static const char * const ad1986a_models[AD1986A_MODELS] = {
-	[AD1986A_AUTO]		= "auto",
-	[AD1986A_6STACK]	= "6stack",
-	[AD1986A_3STACK]	= "3stack",
-	[AD1986A_LAPTOP]	= "laptop",
-	[AD1986A_LAPTOP_EAPD]	= "laptop-eapd",
-	[AD1986A_LAPTOP_AUTOMUTE] = "laptop-automute",
-	[AD1986A_ULTRA]		= "ultra",
-	[AD1986A_SAMSUNG]	= "samsung",
-	[AD1986A_SAMSUNG_P50]	= "samsung-p50",
-};
-
-static const struct snd_pci_quirk ad1986a_cfg_tbl[] = {
-	SND_PCI_QUIRK(0x103c, 0x30af, "HP B2800", AD1986A_LAPTOP_EAPD),
-	SND_PCI_QUIRK(0x1043, 0x1153, "ASUS M9", AD1986A_LAPTOP_EAPD),
-	SND_PCI_QUIRK(0x1043, 0x11f7, "ASUS U5A", AD1986A_LAPTOP_EAPD),
-	SND_PCI_QUIRK(0x1043, 0x1213, "ASUS A6J", AD1986A_LAPTOP_EAPD),
-	SND_PCI_QUIRK(0x1043, 0x1263, "ASUS U5F", AD1986A_LAPTOP_EAPD),
-	SND_PCI_QUIRK(0x1043, 0x1297, "ASUS Z62F", AD1986A_LAPTOP_EAPD),
-	SND_PCI_QUIRK(0x1043, 0x12b3, "ASUS V1j", AD1986A_LAPTOP_EAPD),
-	SND_PCI_QUIRK(0x1043, 0x1302, "ASUS W3j", AD1986A_LAPTOP_EAPD),
-	SND_PCI_QUIRK(0x1043, 0x1443, "ASUS VX1", AD1986A_LAPTOP),
-	SND_PCI_QUIRK(0x1043, 0x1447, "ASUS A8J", AD1986A_3STACK),
-	SND_PCI_QUIRK(0x1043, 0x817f, "ASUS P5", AD1986A_3STACK),
-	SND_PCI_QUIRK(0x1043, 0x818f, "ASUS P5", AD1986A_LAPTOP),
-	SND_PCI_QUIRK(0x1043, 0x81b3, "ASUS P5", AD1986A_3STACK),
-	SND_PCI_QUIRK(0x1043, 0x81cb, "ASUS M2N", AD1986A_3STACK),
-	SND_PCI_QUIRK(0x1043, 0x8234, "ASUS M2N", AD1986A_3STACK),
-	SND_PCI_QUIRK(0x10de, 0xcb84, "ASUS A8N-VM", AD1986A_3STACK),
-	SND_PCI_QUIRK(0x1179, 0xff40, "Toshiba Satellite L40-10Q", AD1986A_3STACK),
-	SND_PCI_QUIRK(0x144d, 0xb03c, "Samsung R55", AD1986A_3STACK),
-	SND_PCI_QUIRK(0x144d, 0xc01e, "FSC V2060", AD1986A_LAPTOP),
-	SND_PCI_QUIRK(0x144d, 0xc024, "Samsung P50", AD1986A_SAMSUNG_P50),
-	SND_PCI_QUIRK(0x144d, 0xc027, "Samsung Q1", AD1986A_ULTRA),
-	SND_PCI_QUIRK_MASK(0x144d, 0xff00, 0xc000, "Samsung", AD1986A_SAMSUNG),
-	SND_PCI_QUIRK(0x144d, 0xc504, "Samsung Q35", AD1986A_3STACK),
-	SND_PCI_QUIRK(0x17aa, 0x1011, "Lenovo M55", AD1986A_LAPTOP),
-	SND_PCI_QUIRK(0x17aa, 0x1017, "Lenovo A60", AD1986A_3STACK),
-	SND_PCI_QUIRK(0x17aa, 0x2066, "Lenovo N100", AD1986A_LAPTOP_AUTOMUTE),
-	SND_PCI_QUIRK(0x17c0, 0x2017, "Samsung M50", AD1986A_LAPTOP),
-	{}
-};
-
-#ifdef CONFIG_PM
-static const struct hda_amp_list ad1986a_loopbacks[] = {
-	{ 0x13, HDA_OUTPUT, 0 }, /* Mic */
-	{ 0x14, HDA_OUTPUT, 0 }, /* Phone */
-	{ 0x15, HDA_OUTPUT, 0 }, /* CD */
-	{ 0x16, HDA_OUTPUT, 0 }, /* Aux */
-	{ 0x17, HDA_OUTPUT, 0 }, /* Line */
-	{ } /* end */
-};
-#endif
-
-static int is_jack_available(struct hda_codec *codec, hda_nid_t nid)
-{
-	unsigned int conf = snd_hda_codec_get_pincfg(codec, nid);
-	return get_defcfg_connect(conf) != AC_JACK_PORT_NONE;
-}
-#endif /* ENABLE_AD_STATIC_QUIRKS */
-
-static int alloc_ad_spec(struct hda_codec *codec)
-{
-	struct ad198x_spec *spec;
-
-	spec = kzalloc(sizeof(*spec), GFP_KERNEL);
-	if (!spec)
-		return -ENOMEM;
-	codec->spec = spec;
-	snd_hda_gen_spec_init(&spec->gen);
-	return 0;
-}
-
-/*
- * AD1986A fixup codes
- */
-
-/* Lenovo N100 seems to report the reversed bit for HP jack-sensing */
-static void ad_fixup_inv_jack_detect(struct hda_codec *codec,
-				     const struct hda_fixup *fix, int action)
-{
-	if (action == HDA_FIXUP_ACT_PRE_PROBE)
-		codec->inv_jack_detect = 1;
-}
-
-enum {
-	AD1986A_FIXUP_INV_JACK_DETECT,
-};
-
-static const struct hda_fixup ad1986a_fixups[] = {
-	[AD1986A_FIXUP_INV_JACK_DETECT] = {
-		.type = HDA_FIXUP_FUNC,
-		.v.func = ad_fixup_inv_jack_detect,
-	},
-};
-
-static const struct snd_pci_quirk ad1986a_fixup_tbl[] = {
-	SND_PCI_QUIRK(0x17aa, 0x2066, "Lenovo N100", AD1986A_FIXUP_INV_JACK_DETECT),
+static const struct hda_model_fixup ad1986a_fixup_models[] = {
+	{ .id = AD1986A_FIXUP_3STACK, .name = "3stack" },
+	{ .id = AD1986A_FIXUP_LAPTOP, .name = "laptop" },
+	{ .id = AD1986A_FIXUP_LAPTOP_IMIC, .name = "laptop-imic" },
+	{ .id = AD1986A_FIXUP_LAPTOP_IMIC, .name = "laptop-eapd" }, /* alias */
 	{}
 };
 
 /*
  */
-static int ad1986a_parse_auto_config(struct hda_codec *codec)
+static int patch_ad1986a(struct hda_codec *codec)
 {
 	int err;
 	struct ad198x_spec *spec;
@@ -1244,7 +341,8 @@ static int ad1986a_parse_auto_config(struct hda_codec *codec)
 	 */
 	spec->gen.multiout.no_share_stream = 1;
 
-	snd_hda_pick_fixup(codec, NULL, ad1986a_fixup_tbl, ad1986a_fixups);
+	snd_hda_pick_fixup(codec, ad1986a_fixup_models, ad1986a_fixup_tbl,
+			   ad1986a_fixups);
 	snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE);
 
 	err = ad198x_parse_auto_config(codec);
@@ -1258,330 +356,11 @@ static int ad1986a_parse_auto_config(struct hda_codec *codec)
 	return 0;
 }
 
-#ifdef ENABLE_AD_STATIC_QUIRKS
-static int patch_ad1986a(struct hda_codec *codec)
-{
-	struct ad198x_spec *spec;
-	int err, board_config;
-
-	board_config = snd_hda_check_board_config(codec, AD1986A_MODELS,
-						  ad1986a_models,
-						  ad1986a_cfg_tbl);
-	if (board_config < 0) {
-		printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n",
-		       codec->chip_name);
-		board_config = AD1986A_AUTO;
-	}
-
-	if (board_config == AD1986A_AUTO)
-		return ad1986a_parse_auto_config(codec);
-
-	err = alloc_ad_spec(codec);
-	if (err < 0)
-		return err;
-	spec = codec->spec;
-
-	err = snd_hda_attach_beep_device(codec, 0x19);
-	if (err < 0) {
-		ad198x_free(codec);
-		return err;
-	}
-	set_beep_amp(spec, 0x18, 0, HDA_OUTPUT);
-
-	spec->multiout.max_channels = 6;
-	spec->multiout.num_dacs = ARRAY_SIZE(ad1986a_dac_nids);
-	spec->multiout.dac_nids = ad1986a_dac_nids;
-	spec->multiout.dig_out_nid = AD1986A_SPDIF_OUT;
-	spec->num_adc_nids = 1;
-	spec->adc_nids = ad1986a_adc_nids;
-	spec->capsrc_nids = ad1986a_capsrc_nids;
-	spec->input_mux = &ad1986a_capture_source;
-	spec->num_mixers = 1;
-	spec->mixers[0] = ad1986a_mixers;
-	spec->num_init_verbs = 1;
-	spec->init_verbs[0] = ad1986a_init_verbs;
-#ifdef CONFIG_PM
-	spec->loopback.amplist = ad1986a_loopbacks;
-#endif
-	spec->vmaster_nid = 0x1b;
-	codec->inv_eapd = 1; /* AD1986A has the inverted EAPD implementation */
-
-	codec->patch_ops = ad198x_patch_ops;
-
-	/* override some parameters */
-	switch (board_config) {
-	case AD1986A_3STACK:
-		spec->num_mixers = 2;
-		spec->mixers[1] = ad1986a_3st_mixers;
-		spec->num_init_verbs = 2;
-		spec->init_verbs[1] = ad1986a_ch2_init;
-		spec->channel_mode = ad1986a_modes;
-		spec->num_channel_mode = ARRAY_SIZE(ad1986a_modes);
-		spec->need_dac_fix = 1;
-		spec->multiout.max_channels = 2;
-		spec->multiout.num_dacs = 1;
-		break;
-	case AD1986A_LAPTOP:
-		spec->mixers[0] = ad1986a_laptop_mixers;
-		spec->multiout.max_channels = 2;
-		spec->multiout.num_dacs = 1;
-		spec->multiout.dac_nids = ad1986a_laptop_dac_nids;
-		break;
-	case AD1986A_LAPTOP_EAPD:
-		spec->num_mixers = 3;
-		spec->mixers[0] = ad1986a_laptop_master_mixers;
-		spec->mixers[1] = ad1986a_laptop_eapd_mixers;
-		spec->mixers[2] = ad1986a_laptop_intmic_mixers;
-		spec->num_init_verbs = 2;
-		spec->init_verbs[1] = ad1986a_eapd_init_verbs;
-		spec->multiout.max_channels = 2;
-		spec->multiout.num_dacs = 1;
-		spec->multiout.dac_nids = ad1986a_laptop_dac_nids;
-		if (!is_jack_available(codec, 0x25))
-			spec->multiout.dig_out_nid = 0;
-		spec->input_mux = &ad1986a_laptop_eapd_capture_source;
-		break;
-	case AD1986A_SAMSUNG:
-		spec->num_mixers = 2;
-		spec->mixers[0] = ad1986a_laptop_master_mixers;
-		spec->mixers[1] = ad1986a_laptop_eapd_mixers;
-		spec->num_init_verbs = 3;
-		spec->init_verbs[1] = ad1986a_eapd_init_verbs;
-		spec->init_verbs[2] = ad1986a_automic_verbs;
-		spec->multiout.max_channels = 2;
-		spec->multiout.num_dacs = 1;
-		spec->multiout.dac_nids = ad1986a_laptop_dac_nids;
-		if (!is_jack_available(codec, 0x25))
-			spec->multiout.dig_out_nid = 0;
-		spec->input_mux = &ad1986a_automic_capture_source;
-		codec->patch_ops.unsol_event = ad1986a_automic_unsol_event;
-		codec->patch_ops.init = ad1986a_automic_init;
-		break;
-	case AD1986A_SAMSUNG_P50:
-		spec->num_mixers = 2;
-		spec->mixers[0] = ad1986a_automute_master_mixers;
-		spec->mixers[1] = ad1986a_laptop_eapd_mixers;
-		spec->num_init_verbs = 4;
-		spec->init_verbs[1] = ad1986a_eapd_init_verbs;
-		spec->init_verbs[2] = ad1986a_automic_verbs;
-		spec->init_verbs[3] = ad1986a_hp_init_verbs;
-		spec->multiout.max_channels = 2;
-		spec->multiout.num_dacs = 1;
-		spec->multiout.dac_nids = ad1986a_laptop_dac_nids;
-		if (!is_jack_available(codec, 0x25))
-			spec->multiout.dig_out_nid = 0;
-		spec->input_mux = &ad1986a_automic_capture_source;
-		codec->patch_ops.unsol_event = ad1986a_samsung_p50_unsol_event;
-		codec->patch_ops.init = ad1986a_samsung_p50_init;
-		break;
-	case AD1986A_LAPTOP_AUTOMUTE:
-		spec->num_mixers = 3;
-		spec->mixers[0] = ad1986a_automute_master_mixers;
-		spec->mixers[1] = ad1986a_laptop_eapd_mixers;
-		spec->mixers[2] = ad1986a_laptop_intmic_mixers;
-		spec->num_init_verbs = 3;
-		spec->init_verbs[1] = ad1986a_eapd_init_verbs;
-		spec->init_verbs[2] = ad1986a_hp_init_verbs;
-		spec->multiout.max_channels = 2;
-		spec->multiout.num_dacs = 1;
-		spec->multiout.dac_nids = ad1986a_laptop_dac_nids;
-		if (!is_jack_available(codec, 0x25))
-			spec->multiout.dig_out_nid = 0;
-		spec->input_mux = &ad1986a_laptop_eapd_capture_source;
-		codec->patch_ops.unsol_event = ad1986a_hp_unsol_event;
-		codec->patch_ops.init = ad1986a_hp_init;
-		/* Lenovo N100 seems to report the reversed bit
-		 * for HP jack-sensing
-		 */
-		spec->inv_jack_detect = 1;
-		break;
-	case AD1986A_ULTRA:
-		spec->mixers[0] = ad1986a_laptop_eapd_mixers;
-		spec->num_init_verbs = 2;
-		spec->init_verbs[1] = ad1986a_ultra_init;
-		spec->multiout.max_channels = 2;
-		spec->multiout.num_dacs = 1;
-		spec->multiout.dac_nids = ad1986a_laptop_dac_nids;
-		spec->multiout.dig_out_nid = 0;
-		break;
-	}
-
-	/* AD1986A has a hardware problem that it can't share a stream
-	 * with multiple output pins.  The copy of front to surrounds
-	 * causes noisy or silent outputs at a certain timing, e.g.
-	 * changing the volume.
-	 * So, let's disable the shared stream.
-	 */
-	spec->multiout.no_share_stream = 1;
-
-	codec->no_trigger_sense = 1;
-	codec->no_sticky_stream = 1;
-
-	return 0;
-}
-#else /* ENABLE_AD_STATIC_QUIRKS */
-#define patch_ad1986a	ad1986a_parse_auto_config
-#endif /* ENABLE_AD_STATIC_QUIRKS */
 
 /*
  * AD1983 specific
  */
 
-#ifdef ENABLE_AD_STATIC_QUIRKS
-#define AD1983_SPDIF_OUT	0x02
-#define AD1983_DAC		0x03
-#define AD1983_ADC		0x04
-
-static const hda_nid_t ad1983_dac_nids[1] = { AD1983_DAC };
-static const hda_nid_t ad1983_adc_nids[1] = { AD1983_ADC };
-static const hda_nid_t ad1983_capsrc_nids[1] = { 0x15 };
-
-static const struct hda_input_mux ad1983_capture_source = {
-	.num_items = 4,
-	.items = {
-		{ "Mic", 0x0 },
-		{ "Line", 0x1 },
-		{ "Mix", 0x2 },
-		{ "Mix Mono", 0x3 },
-	},
-};
-
-/*
- * SPDIF playback route
- */
-static int ad1983_spdif_route_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
-{
-	static const char * const texts[] = { "PCM", "ADC" };
-
-	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
-	uinfo->count = 1;
-	uinfo->value.enumerated.items = 2;
-	if (uinfo->value.enumerated.item > 1)
-		uinfo->value.enumerated.item = 1;
-	strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
-	return 0;
-}
-
-static int ad1983_spdif_route_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
-{
-	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
-	struct ad198x_spec *spec = codec->spec;
-
-	ucontrol->value.enumerated.item[0] = spec->spdif_route;
-	return 0;
-}
-
-static int ad1983_spdif_route_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
-{
-	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
-	struct ad198x_spec *spec = codec->spec;
-
-	if (ucontrol->value.enumerated.item[0] > 1)
-		return -EINVAL;
-	if (spec->spdif_route != ucontrol->value.enumerated.item[0]) {
-		spec->spdif_route = ucontrol->value.enumerated.item[0];
-		snd_hda_codec_write_cache(codec, spec->multiout.dig_out_nid, 0,
-					  AC_VERB_SET_CONNECT_SEL,
-					  spec->spdif_route);
-		return 1;
-	}
-	return 0;
-}
-
-static const struct snd_kcontrol_new ad1983_mixers[] = {
-	HDA_CODEC_VOLUME("Front Playback Volume", 0x05, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Front Playback Switch", 0x05, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Headphone Playback Volume", 0x06, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Headphone Playback Switch", 0x06, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x07, 1, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE_MONO("Mono Playback Switch", 0x07, 1, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("PCM Playback Volume", 0x11, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("PCM Playback Switch", 0x11, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Mic Playback Volume", 0x12, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Mic Playback Switch", 0x12, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Line Playback Volume", 0x13, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Line Playback Switch", 0x13, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Mic Boost Volume", 0x0c, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Capture Volume", 0x15, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Capture Switch", 0x15, 0x0, HDA_OUTPUT),
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		.name = "Capture Source",
-		.info = ad198x_mux_enum_info,
-		.get = ad198x_mux_enum_get,
-		.put = ad198x_mux_enum_put,
-	},
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		.name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,NONE) "Source",
-		.info = ad1983_spdif_route_info,
-		.get = ad1983_spdif_route_get,
-		.put = ad1983_spdif_route_put,
-	},
-	{ } /* end */
-};
-
-static const struct hda_verb ad1983_init_verbs[] = {
-	/* Front, HP, Mono; mute as default */
-	{0x05, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
-	{0x06, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
-	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
-	/* Beep, PCM, Mic, Line-In: mute */
-	{0x10, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
-	{0x11, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
-	{0x12, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
-	{0x13, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
-	/* Front, HP selectors; from Mix */
-	{0x05, AC_VERB_SET_CONNECT_SEL, 0x01},
-	{0x06, AC_VERB_SET_CONNECT_SEL, 0x01},
-	/* Mono selector; from Mix */
-	{0x0b, AC_VERB_SET_CONNECT_SEL, 0x03},
-	/* Mic selector; Mic */
-	{0x0c, AC_VERB_SET_CONNECT_SEL, 0x0},
-	/* Line-in selector: Line-in */
-	{0x0d, AC_VERB_SET_CONNECT_SEL, 0x0},
-	/* Mic boost: 0dB */
-	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
-	/* Record selector: mic */
-	{0x15, AC_VERB_SET_CONNECT_SEL, 0x0},
-	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
-	/* SPDIF route: PCM */
-	{0x02, AC_VERB_SET_CONNECT_SEL, 0x0},
-	/* Front Pin */
-	{0x05, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
-	/* HP Pin */
-	{0x06, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0 },
-	/* Mono Pin */
-	{0x07, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
-	/* Mic Pin */
-	{0x08, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
-	/* Line Pin */
-	{0x09, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
-	{ } /* end */
-};
-
-#ifdef CONFIG_PM
-static const struct hda_amp_list ad1983_loopbacks[] = {
-	{ 0x12, HDA_OUTPUT, 0 }, /* Mic */
-	{ 0x13, HDA_OUTPUT, 0 }, /* Line */
-	{ } /* end */
-};
-#endif
-
-/* models */
-enum {
-	AD1983_AUTO,
-	AD1983_BASIC,
-	AD1983_MODELS
-};
-
-static const char * const ad1983_models[AD1983_MODELS] = {
-	[AD1983_AUTO]		= "auto",
-	[AD1983_BASIC]		= "basic",
-};
-#endif /* ENABLE_AD_STATIC_QUIRKS */
-
-
 /*
  * SPDIF mux control for AD1983 auto-parser
  */
@@ -1656,7 +435,7 @@ static int ad1983_add_spdif_mux_ctl(struct hda_codec *codec)
 	return 0;
 }
 
-static int ad1983_parse_auto_config(struct hda_codec *codec)
+static int patch_ad1983(struct hda_codec *codec)
 {
 	struct ad198x_spec *spec;
 	int err;
@@ -1681,432 +460,11 @@ static int ad1983_parse_auto_config(struct hda_codec *codec)
 	return err;
 }
 
-#ifdef ENABLE_AD_STATIC_QUIRKS
-static int patch_ad1983(struct hda_codec *codec)
-{
-	struct ad198x_spec *spec;
-	int board_config;
-	int err;
-
-	board_config = snd_hda_check_board_config(codec, AD1983_MODELS,
-						  ad1983_models, NULL);
-	if (board_config < 0) {
-		printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n",
-		       codec->chip_name);
-		board_config = AD1983_AUTO;
-	}
-
-	if (board_config == AD1983_AUTO)
-		return ad1983_parse_auto_config(codec);
-
-	err = alloc_ad_spec(codec);
-	if (err < 0)
-		return err;
-	spec = codec->spec;
-
-	err = snd_hda_attach_beep_device(codec, 0x10);
-	if (err < 0) {
-		ad198x_free(codec);
-		return err;
-	}
-	set_beep_amp(spec, 0x10, 0, HDA_OUTPUT);
-
-	spec->multiout.max_channels = 2;
-	spec->multiout.num_dacs = ARRAY_SIZE(ad1983_dac_nids);
-	spec->multiout.dac_nids = ad1983_dac_nids;
-	spec->multiout.dig_out_nid = AD1983_SPDIF_OUT;
-	spec->num_adc_nids = 1;
-	spec->adc_nids = ad1983_adc_nids;
-	spec->capsrc_nids = ad1983_capsrc_nids;
-	spec->input_mux = &ad1983_capture_source;
-	spec->num_mixers = 1;
-	spec->mixers[0] = ad1983_mixers;
-	spec->num_init_verbs = 1;
-	spec->init_verbs[0] = ad1983_init_verbs;
-	spec->spdif_route = 0;
-#ifdef CONFIG_PM
-	spec->loopback.amplist = ad1983_loopbacks;
-#endif
-	spec->vmaster_nid = 0x05;
-
-	codec->patch_ops = ad198x_patch_ops;
-
-	codec->no_trigger_sense = 1;
-	codec->no_sticky_stream = 1;
-
-	return 0;
-}
-#else /* ENABLE_AD_STATIC_QUIRKS */
-#define patch_ad1983	ad1983_parse_auto_config
-#endif /* ENABLE_AD_STATIC_QUIRKS */
-
 
 /*
  * AD1981 HD specific
  */
 
-#ifdef ENABLE_AD_STATIC_QUIRKS
-#define AD1981_SPDIF_OUT	0x02
-#define AD1981_DAC		0x03
-#define AD1981_ADC		0x04
-
-static const hda_nid_t ad1981_dac_nids[1] = { AD1981_DAC };
-static const hda_nid_t ad1981_adc_nids[1] = { AD1981_ADC };
-static const hda_nid_t ad1981_capsrc_nids[1] = { 0x15 };
-
-/* 0x0c, 0x09, 0x0e, 0x0f, 0x19, 0x05, 0x18, 0x17 */
-static const struct hda_input_mux ad1981_capture_source = {
-	.num_items = 7,
-	.items = {
-		{ "Front Mic", 0x0 },
-		{ "Line", 0x1 },
-		{ "Mix", 0x2 },
-		{ "Mix Mono", 0x3 },
-		{ "CD", 0x4 },
-		{ "Mic", 0x6 },
-		{ "Aux", 0x7 },
-	},
-};
-
-static const struct snd_kcontrol_new ad1981_mixers[] = {
-	HDA_CODEC_VOLUME("Front Playback Volume", 0x05, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Front Playback Switch", 0x05, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Headphone Playback Volume", 0x06, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Headphone Playback Switch", 0x06, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x07, 1, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE_MONO("Mono Playback Switch", 0x07, 1, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("PCM Playback Volume", 0x11, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("PCM Playback Switch", 0x11, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x12, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Front Mic Playback Switch", 0x12, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Line Playback Volume", 0x13, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Line Playback Switch", 0x13, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Aux Playback Volume", 0x1b, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Aux Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Mic Playback Volume", 0x1c, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Mic Playback Switch", 0x1c, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("CD Playback Volume", 0x1d, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("CD Playback Switch", 0x1d, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x08, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME("Capture Volume", 0x15, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Capture Switch", 0x15, 0x0, HDA_OUTPUT),
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		.name = "Capture Source",
-		.info = ad198x_mux_enum_info,
-		.get = ad198x_mux_enum_get,
-		.put = ad198x_mux_enum_put,
-	},
-	/* identical with AD1983 */
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		.name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,NONE) "Source",
-		.info = ad1983_spdif_route_info,
-		.get = ad1983_spdif_route_get,
-		.put = ad1983_spdif_route_put,
-	},
-	{ } /* end */
-};
-
-static const struct hda_verb ad1981_init_verbs[] = {
-	/* Front, HP, Mono; mute as default */
-	{0x05, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
-	{0x06, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
-	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
-	/* Beep, PCM, Front Mic, Line, Rear Mic, Aux, CD-In: mute */
-	{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
-	{0x11, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
-	{0x12, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
-	{0x13, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
-	{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
-	{0x1c, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
-	{0x1d, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
-	/* Front, HP selectors; from Mix */
-	{0x05, AC_VERB_SET_CONNECT_SEL, 0x01},
-	{0x06, AC_VERB_SET_CONNECT_SEL, 0x01},
-	/* Mono selector; from Mix */
-	{0x0b, AC_VERB_SET_CONNECT_SEL, 0x03},
-	/* Mic Mixer; select Front Mic */
-	{0x1e, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
-	{0x1f, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
-	/* Mic boost: 0dB */
-	{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	/* Record selector: Front mic */
-	{0x15, AC_VERB_SET_CONNECT_SEL, 0x0},
-	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
-	/* SPDIF route: PCM */
-	{0x02, AC_VERB_SET_CONNECT_SEL, 0x0},
-	/* Front Pin */
-	{0x05, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
-	/* HP Pin */
-	{0x06, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0 },
-	/* Mono Pin */
-	{0x07, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
-	/* Front & Rear Mic Pins */
-	{0x08, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
-	{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
-	/* Line Pin */
-	{0x09, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
-	/* Digital Beep */
-	{0x0d, AC_VERB_SET_CONNECT_SEL, 0x00},
-	/* Line-Out as Input: disabled */
-	{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
-	{ } /* end */
-};
-
-#ifdef CONFIG_PM
-static const struct hda_amp_list ad1981_loopbacks[] = {
-	{ 0x12, HDA_OUTPUT, 0 }, /* Front Mic */
-	{ 0x13, HDA_OUTPUT, 0 }, /* Line */
-	{ 0x1b, HDA_OUTPUT, 0 }, /* Aux */
-	{ 0x1c, HDA_OUTPUT, 0 }, /* Mic */
-	{ 0x1d, HDA_OUTPUT, 0 }, /* CD */
-	{ } /* end */
-};
-#endif
-
-/*
- * Patch for HP nx6320
- *
- * nx6320 uses EAPD in the reverse way - EAPD-on means the internal
- * speaker output enabled _and_ mute-LED off.
- */
-
-#define AD1981_HP_EVENT		0x37
-#define AD1981_MIC_EVENT	0x38
-
-static const struct hda_verb ad1981_hp_init_verbs[] = {
-	{0x05, AC_VERB_SET_EAPD_BTLENABLE, 0x00 }, /* default off */
-	/* pin sensing on HP and Mic jacks */
-	{0x06, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1981_HP_EVENT},
-	{0x08, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1981_MIC_EVENT},
-	{}
-};
-
-/* turn on/off EAPD (+ mute HP) as a master switch */
-static int ad1981_hp_master_sw_put(struct snd_kcontrol *kcontrol,
-				   struct snd_ctl_elem_value *ucontrol)
-{
-	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
-	struct ad198x_spec *spec = codec->spec;
-
-	if (! ad198x_eapd_put(kcontrol, ucontrol))
-		return 0;
-	/* change speaker pin appropriately */
-	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,
-				 spec->cur_eapd ? 0 : HDA_AMP_MUTE);
-	return 1;
-}
-
-/* bind volumes of both NID 0x05 and 0x06 */
-static const struct hda_bind_ctls ad1981_hp_bind_master_vol = {
-	.ops = &snd_hda_bind_vol,
-	.values = {
-		HDA_COMPOSE_AMP_VAL(0x05, 3, 0, HDA_OUTPUT),
-		HDA_COMPOSE_AMP_VAL(0x06, 3, 0, HDA_OUTPUT),
-		0
-	},
-};
-
-/* mute internal speaker if HP is plugged */
-static void ad1981_hp_automute(struct hda_codec *codec)
-{
-	unsigned int present;
-
-	present = snd_hda_jack_detect(codec, 0x06);
-	snd_hda_codec_amp_stereo(codec, 0x05, HDA_OUTPUT, 0,
-				 HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
-}
-
-/* toggle input of built-in and mic jack appropriately */
-static void ad1981_hp_automic(struct hda_codec *codec)
-{
-	static const struct hda_verb mic_jack_on[] = {
-		{0x1f, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
-		{0x1e, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
-		{}
-	};
-	static const struct hda_verb mic_jack_off[] = {
-		{0x1e, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
-		{0x1f, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
-		{}
-	};
-	unsigned int present;
-
-	present = snd_hda_jack_detect(codec, 0x08);
-	if (present)
-		snd_hda_sequence_write(codec, mic_jack_on);
-	else
-		snd_hda_sequence_write(codec, mic_jack_off);
-}
-
-/* unsolicited event for HP jack sensing */
-static void ad1981_hp_unsol_event(struct hda_codec *codec,
-				  unsigned int res)
-{
-	res >>= 26;
-	switch (res) {
-	case AD1981_HP_EVENT:
-		ad1981_hp_automute(codec);
-		break;
-	case AD1981_MIC_EVENT:
-		ad1981_hp_automic(codec);
-		break;
-	}
-}
-
-static const struct hda_input_mux ad1981_hp_capture_source = {
-	.num_items = 3,
-	.items = {
-		{ "Mic", 0x0 },
-		{ "Dock Mic", 0x1 },
-		{ "Mix", 0x2 },
-	},
-};
-
-static const struct snd_kcontrol_new ad1981_hp_mixers[] = {
-	HDA_BIND_VOL("Master Playback Volume", &ad1981_hp_bind_master_vol),
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		.subdevice = HDA_SUBDEV_NID_FLAG | 0x05,
-		.name = "Master Playback Switch",
-		.info = ad198x_eapd_info,
-		.get = ad198x_eapd_get,
-		.put = ad1981_hp_master_sw_put,
-		.private_value = 0x05,
-	},
-	HDA_CODEC_VOLUME("PCM Playback Volume", 0x11, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("PCM Playback Switch", 0x11, 0x0, HDA_OUTPUT),
-#if 0
-	/* FIXME: analog mic/line loopback doesn't work with my tests...
-	 *        (although recording is OK)
-	 */
-	HDA_CODEC_VOLUME("Mic Playback Volume", 0x12, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Mic Playback Switch", 0x12, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Dock Mic Playback Volume", 0x13, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Dock Mic Playback Switch", 0x13, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x1c, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x1c, 0x0, HDA_OUTPUT),
-	/* FIXME: does this laptop have analog CD connection? */
-	HDA_CODEC_VOLUME("CD Playback Volume", 0x1d, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("CD Playback Switch", 0x1d, 0x0, HDA_OUTPUT),
-#endif
-	HDA_CODEC_VOLUME("Mic Boost Volume", 0x08, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x18, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME("Capture Volume", 0x15, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Capture Switch", 0x15, 0x0, HDA_OUTPUT),
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		.name = "Capture Source",
-		.info = ad198x_mux_enum_info,
-		.get = ad198x_mux_enum_get,
-		.put = ad198x_mux_enum_put,
-	},
-	{ } /* end */
-};
-
-/* initialize jack-sensing, too */
-static int ad1981_hp_init(struct hda_codec *codec)
-{
-	ad198x_init(codec);
-	ad1981_hp_automute(codec);
-	ad1981_hp_automic(codec);
-	return 0;
-}
-
-/* configuration for Toshiba Laptops */
-static const struct hda_verb ad1981_toshiba_init_verbs[] = {
-	{0x05, AC_VERB_SET_EAPD_BTLENABLE, 0x01 }, /* default on */
-	/* pin sensing on HP and Mic jacks */
-	{0x06, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1981_HP_EVENT},
-	{0x08, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1981_MIC_EVENT},
-	{}
-};
-
-static const struct snd_kcontrol_new ad1981_toshiba_mixers[] = {
-	HDA_CODEC_VOLUME("Amp Volume", 0x1a, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Amp Switch", 0x1a, 0x0, HDA_OUTPUT),
-	{ }
-};
-
-/* configuration for Lenovo Thinkpad T60 */
-static const struct snd_kcontrol_new ad1981_thinkpad_mixers[] = {
-	HDA_CODEC_VOLUME("Master Playback Volume", 0x05, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Master Playback Switch", 0x05, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("PCM Playback Volume", 0x11, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("PCM Playback Switch", 0x11, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Mic Playback Volume", 0x12, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Mic Playback Switch", 0x12, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("CD Playback Volume", 0x1d, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("CD Playback Switch", 0x1d, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Mic Boost Volume", 0x08, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME("Capture Volume", 0x15, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Capture Switch", 0x15, 0x0, HDA_OUTPUT),
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		.name = "Capture Source",
-		.info = ad198x_mux_enum_info,
-		.get = ad198x_mux_enum_get,
-		.put = ad198x_mux_enum_put,
-	},
-	/* identical with AD1983 */
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		.name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,NONE) "Source",
-		.info = ad1983_spdif_route_info,
-		.get = ad1983_spdif_route_get,
-		.put = ad1983_spdif_route_put,
-	},
-	{ } /* end */
-};
-
-static const struct hda_input_mux ad1981_thinkpad_capture_source = {
-	.num_items = 3,
-	.items = {
-		{ "Mic", 0x0 },
-		{ "Mix", 0x2 },
-		{ "CD", 0x4 },
-	},
-};
-
-/* models */
-enum {
-	AD1981_AUTO,
-	AD1981_BASIC,
-	AD1981_HP,
-	AD1981_THINKPAD,
-	AD1981_TOSHIBA,
-	AD1981_MODELS
-};
-
-static const char * const ad1981_models[AD1981_MODELS] = {
-	[AD1981_AUTO]		= "auto",
-	[AD1981_HP]		= "hp",
-	[AD1981_THINKPAD]	= "thinkpad",
-	[AD1981_BASIC]		= "basic",
-	[AD1981_TOSHIBA]	= "toshiba"
-};
-
-static const struct snd_pci_quirk ad1981_cfg_tbl[] = {
-	SND_PCI_QUIRK(0x1014, 0x0597, "Lenovo Z60", AD1981_THINKPAD),
-	SND_PCI_QUIRK(0x1014, 0x05b7, "Lenovo Z60m", AD1981_THINKPAD),
-	/* All HP models */
-	SND_PCI_QUIRK_VENDOR(0x103c, "HP nx", AD1981_HP),
-	SND_PCI_QUIRK(0x1179, 0x0001, "Toshiba U205", AD1981_TOSHIBA),
-	/* Lenovo Thinkpad T60/X60/Z6xx */
-	SND_PCI_QUIRK_VENDOR(0x17aa, "Lenovo Thinkpad", AD1981_THINKPAD),
-	/* HP nx6320 (reversed SSID, H/W bug) */
-	SND_PCI_QUIRK(0x30b0, 0x103c, "HP nx6320", AD1981_HP),
-	{}
-};
-#endif /* ENABLE_AD_STATIC_QUIRKS */
-
-
 /* follow EAPD via vmaster hook */
 static void ad_vmaster_eapd_hook(void *private_data, int enabled)
 {
@@ -2172,7 +530,7 @@ static const struct snd_pci_quirk ad1981_fixup_tbl[] = {
 	{}
 };
 
-static int ad1981_parse_auto_config(struct hda_codec *codec)
+static int patch_ad1981(struct hda_codec *codec)
 {
 	struct ad198x_spec *spec;
 	int err;
@@ -2205,110 +563,6 @@ static int ad1981_parse_auto_config(struct hda_codec *codec)
 	return err;
 }
 
-#ifdef ENABLE_AD_STATIC_QUIRKS
-static int patch_ad1981(struct hda_codec *codec)
-{
-	struct ad198x_spec *spec;
-	int err, board_config;
-
-	board_config = snd_hda_check_board_config(codec, AD1981_MODELS,
-						  ad1981_models,
-						  ad1981_cfg_tbl);
-	if (board_config < 0) {
-		printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n",
-		       codec->chip_name);
-		board_config = AD1981_AUTO;
-	}
-
-	if (board_config == AD1981_AUTO)
-		return ad1981_parse_auto_config(codec);
-
-	err = alloc_ad_spec(codec);
-	if (err < 0)
-		return -ENOMEM;
-	spec = codec->spec;
-
-	err = snd_hda_attach_beep_device(codec, 0x10);
-	if (err < 0) {
-		ad198x_free(codec);
-		return err;
-	}
-	set_beep_amp(spec, 0x0d, 0, HDA_OUTPUT);
-
-	spec->multiout.max_channels = 2;
-	spec->multiout.num_dacs = ARRAY_SIZE(ad1981_dac_nids);
-	spec->multiout.dac_nids = ad1981_dac_nids;
-	spec->multiout.dig_out_nid = AD1981_SPDIF_OUT;
-	spec->num_adc_nids = 1;
-	spec->adc_nids = ad1981_adc_nids;
-	spec->capsrc_nids = ad1981_capsrc_nids;
-	spec->input_mux = &ad1981_capture_source;
-	spec->num_mixers = 1;
-	spec->mixers[0] = ad1981_mixers;
-	spec->num_init_verbs = 1;
-	spec->init_verbs[0] = ad1981_init_verbs;
-	spec->spdif_route = 0;
-#ifdef CONFIG_PM
-	spec->loopback.amplist = ad1981_loopbacks;
-#endif
-	spec->vmaster_nid = 0x05;
-
-	codec->patch_ops = ad198x_patch_ops;
-
-	/* override some parameters */
-	switch (board_config) {
-	case AD1981_HP:
-		spec->mixers[0] = ad1981_hp_mixers;
-		spec->num_init_verbs = 2;
-		spec->init_verbs[1] = ad1981_hp_init_verbs;
-		if (!is_jack_available(codec, 0x0a))
-			spec->multiout.dig_out_nid = 0;
-		spec->input_mux = &ad1981_hp_capture_source;
-
-		codec->patch_ops.init = ad1981_hp_init;
-		codec->patch_ops.unsol_event = ad1981_hp_unsol_event;
-		/* set the upper-limit for mixer amp to 0dB for avoiding the
-		 * possible damage by overloading
-		 */
-		snd_hda_override_amp_caps(codec, 0x11, HDA_INPUT,
-					  (0x17 << AC_AMPCAP_OFFSET_SHIFT) |
-					  (0x17 << AC_AMPCAP_NUM_STEPS_SHIFT) |
-					  (0x05 << AC_AMPCAP_STEP_SIZE_SHIFT) |
-					  (1 << AC_AMPCAP_MUTE_SHIFT));
-		break;
-	case AD1981_THINKPAD:
-		spec->mixers[0] = ad1981_thinkpad_mixers;
-		spec->input_mux = &ad1981_thinkpad_capture_source;
-		/* set the upper-limit for mixer amp to 0dB for avoiding the
-		 * possible damage by overloading
-		 */
-		snd_hda_override_amp_caps(codec, 0x11, HDA_INPUT,
-					  (0x17 << AC_AMPCAP_OFFSET_SHIFT) |
-					  (0x17 << AC_AMPCAP_NUM_STEPS_SHIFT) |
-					  (0x05 << AC_AMPCAP_STEP_SIZE_SHIFT) |
-					  (1 << AC_AMPCAP_MUTE_SHIFT));
-		break;
-	case AD1981_TOSHIBA:
-		spec->mixers[0] = ad1981_hp_mixers;
-		spec->mixers[1] = ad1981_toshiba_mixers;
-		spec->num_init_verbs = 2;
-		spec->init_verbs[1] = ad1981_toshiba_init_verbs;
-		spec->multiout.dig_out_nid = 0;
-		spec->input_mux = &ad1981_hp_capture_source;
-		codec->patch_ops.init = ad1981_hp_init;
-		codec->patch_ops.unsol_event = ad1981_hp_unsol_event;
-		break;
-	}
-
-	codec->no_trigger_sense = 1;
-	codec->no_sticky_stream = 1;
-
-	return 0;
-}
-#else /* ENABLE_AD_STATIC_QUIRKS */
-#define patch_ad1981	ad1981_parse_auto_config
-#endif /* ENABLE_AD_STATIC_QUIRKS */
-
 
 /*
  * AD1988
@@ -2395,90 +649,7 @@ static int patch_ad1981(struct hda_codec *codec)
  *      E/F quad mic array
  */
 
-
 #ifdef ENABLE_AD_STATIC_QUIRKS
-/* models */
-enum {
-	AD1988_AUTO,
-	AD1988_6STACK,
-	AD1988_6STACK_DIG,
-	AD1988_3STACK,
-	AD1988_3STACK_DIG,
-	AD1988_LAPTOP,
-	AD1988_LAPTOP_DIG,
-	AD1988_MODEL_LAST,
-};
-
-/* reivision id to check workarounds */
-#define AD1988A_REV2		0x100200
-
-#define is_rev2(codec) \
-	((codec)->vendor_id == 0x11d41988 && \
-	 (codec)->revision_id == AD1988A_REV2)
-
-/*
- * mixers
- */
-
-static const hda_nid_t ad1988_6stack_dac_nids[4] = {
-	0x04, 0x06, 0x05, 0x0a
-};
-
-static const hda_nid_t ad1988_3stack_dac_nids[3] = {
-	0x04, 0x05, 0x0a
-};
-
-/* for AD1988A revision-2, DAC2-4 are swapped */
-static const hda_nid_t ad1988_6stack_dac_nids_rev2[4] = {
-	0x04, 0x05, 0x0a, 0x06
-};
-
-static const hda_nid_t ad1988_alt_dac_nid[1] = {
-	0x03
-};
-
-static const hda_nid_t ad1988_3stack_dac_nids_rev2[3] = {
-	0x04, 0x0a, 0x06
-};
-
-static const hda_nid_t ad1988_adc_nids[3] = {
-	0x08, 0x09, 0x0f
-};
-
-static const hda_nid_t ad1988_capsrc_nids[3] = {
-	0x0c, 0x0d, 0x0e
-};
-
-#define AD1988_SPDIF_OUT		0x02
-#define AD1988_SPDIF_OUT_HDMI	0x0b
-#define AD1988_SPDIF_IN		0x07
-
-static const hda_nid_t ad1989b_slave_dig_outs[] = {
-	AD1988_SPDIF_OUT, AD1988_SPDIF_OUT_HDMI, 0
-};
-
-static const struct hda_input_mux ad1988_6stack_capture_source = {
-	.num_items = 5,
-	.items = {
-		{ "Front Mic", 0x1 },	/* port-B */
-		{ "Line", 0x2 },	/* port-C */
-		{ "Mic", 0x4 },		/* port-E */
-		{ "CD", 0x5 },
-		{ "Mix", 0x9 },
-	},
-};
-
-static const struct hda_input_mux ad1988_laptop_capture_source = {
-	.num_items = 3,
-	.items = {
-		{ "Mic/Line", 0x1 },	/* port-B */
-		{ "CD", 0x5 },
-		{ "Mix", 0x9 },
-	},
-};
-
-/*
- */
 static int ad198x_ch_mode_info(struct snd_kcontrol *kcontrol,
 			       struct snd_ctl_elem_info *uinfo)
 {
@@ -2509,636 +680,73 @@ static int ad198x_ch_mode_put(struct snd_kcontrol *kcontrol,
 		spec->multiout.num_dacs = spec->multiout.max_channels / 2;
 	return err;
 }
+#endif /* ENABLE_AD_STATIC_QUIRKS */
 
-/* 6-stack mode */
-static const struct snd_kcontrol_new ad1988_6stack_mixers1[] = {
-	HDA_CODEC_VOLUME("Front Playback Volume", 0x04, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Surround Playback Volume", 0x06, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x05, 1, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x05, 2, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Side Playback Volume", 0x0a, 0x0, HDA_OUTPUT),
-	{ } /* end */
-};
-
-static const struct snd_kcontrol_new ad1988_6stack_mixers1_rev2[] = {
-	HDA_CODEC_VOLUME("Front Playback Volume", 0x04, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Surround Playback Volume", 0x05, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0a, 1, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0a, 2, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Side Playback Volume", 0x06, 0x0, HDA_OUTPUT),
-	{ } /* end */
-};
-
-static const struct snd_kcontrol_new ad1988_6stack_mixers2[] = {
-	HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT),
-	HDA_BIND_MUTE("Front Playback Switch", 0x29, 2, HDA_INPUT),
-	HDA_BIND_MUTE("Surround Playback Switch", 0x2a, 2, HDA_INPUT),
-	HDA_BIND_MUTE_MONO("Center Playback Switch", 0x27, 1, 2, HDA_INPUT),
-	HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x27, 2, 2, HDA_INPUT),
-	HDA_BIND_MUTE("Side Playback Switch", 0x28, 2, HDA_INPUT),
-	HDA_BIND_MUTE("Headphone Playback Switch", 0x22, 2, HDA_INPUT),
-	HDA_BIND_MUTE("Mono Playback Switch", 0x1e, 2, HDA_INPUT),
-
-	HDA_CODEC_VOLUME("CD Playback Volume", 0x20, 0x6, HDA_INPUT),
-	HDA_CODEC_MUTE("CD Playback Switch", 0x20, 0x6, HDA_INPUT),
-	HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x20, 0x0, HDA_INPUT),
-	HDA_CODEC_MUTE("Front Mic Playback Switch", 0x20, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME("Line Playback Volume", 0x20, 0x1, HDA_INPUT),
-	HDA_CODEC_MUTE("Line Playback Switch", 0x20, 0x1, HDA_INPUT),
-	HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x4, HDA_INPUT),
-	HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x4, HDA_INPUT),
-
-	HDA_CODEC_VOLUME("Analog Mix Playback Volume", 0x21, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Analog Mix Playback Switch", 0x21, 0x0, HDA_OUTPUT),
-
-	HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x39, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Mic Boost Volume", 0x3c, 0x0, HDA_OUTPUT),
-	{ } /* end */
-};
-
-/* 3-stack mode */
-static const struct snd_kcontrol_new ad1988_3stack_mixers1[] = {
-	HDA_CODEC_VOLUME("Front Playback Volume", 0x04, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Surround Playback Volume", 0x0a, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x05, 1, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x05, 2, 0x0, HDA_OUTPUT),
-	{ } /* end */
-};
-
-static const struct snd_kcontrol_new ad1988_3stack_mixers1_rev2[] = {
-	HDA_CODEC_VOLUME("Front Playback Volume", 0x04, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Surround Playback Volume", 0x0a, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x06, 1, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x06, 2, 0x0, HDA_OUTPUT),
-	{ } /* end */
-};
-
-static const struct snd_kcontrol_new ad1988_3stack_mixers2[] = {
-	HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT),
-	HDA_BIND_MUTE("Front Playback Switch", 0x29, 2, HDA_INPUT),
-	HDA_BIND_MUTE("Surround Playback Switch", 0x2c, 2, HDA_INPUT),
-	HDA_BIND_MUTE_MONO("Center Playback Switch", 0x26, 1, 2, HDA_INPUT),
-	HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x26, 2, 2, HDA_INPUT),
-	HDA_BIND_MUTE("Headphone Playback Switch", 0x22, 2, HDA_INPUT),
-	HDA_BIND_MUTE("Mono Playback Switch", 0x1e, 2, HDA_INPUT),
-
-	HDA_CODEC_VOLUME("CD Playback Volume", 0x20, 0x6, HDA_INPUT),
-	HDA_CODEC_MUTE("CD Playback Switch", 0x20, 0x6, HDA_INPUT),
-	HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x20, 0x0, HDA_INPUT),
-	HDA_CODEC_MUTE("Front Mic Playback Switch", 0x20, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME("Line Playback Volume", 0x20, 0x1, HDA_INPUT),
-	HDA_CODEC_MUTE("Line Playback Switch", 0x20, 0x1, HDA_INPUT),
-	HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x4, HDA_INPUT),
-	HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x4, HDA_INPUT),
-
-	HDA_CODEC_VOLUME("Analog Mix Playback Volume", 0x21, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Analog Mix Playback Switch", 0x21, 0x0, HDA_OUTPUT),
-
-	HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x39, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Mic Boost Volume", 0x3c, 0x0, HDA_OUTPUT),
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		.name = "Channel Mode",
-		.info = ad198x_ch_mode_info,
-		.get = ad198x_ch_mode_get,
-		.put = ad198x_ch_mode_put,
-	},
-
-	{ } /* end */
-};
-
-/* laptop mode */
-static const struct snd_kcontrol_new ad1988_laptop_mixers[] = {
-	HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("PCM Playback Volume", 0x04, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("PCM Playback Switch", 0x29, 0x0, HDA_INPUT),
-	HDA_BIND_MUTE("Mono Playback Switch", 0x1e, 2, HDA_INPUT),
-
-	HDA_CODEC_VOLUME("CD Playback Volume", 0x20, 0x6, HDA_INPUT),
-	HDA_CODEC_MUTE("CD Playback Switch", 0x20, 0x6, HDA_INPUT),
-	HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x0, HDA_INPUT),
-	HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME("Line Playback Volume", 0x20, 0x1, HDA_INPUT),
-	HDA_CODEC_MUTE("Line Playback Switch", 0x20, 0x1, HDA_INPUT),
-
-	HDA_CODEC_VOLUME("Analog Mix Playback Volume", 0x21, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Analog Mix Playback Switch", 0x21, 0x0, HDA_OUTPUT),
-
-	HDA_CODEC_VOLUME("Mic Boost Volume", 0x39, 0x0, HDA_OUTPUT),
-
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		.name = "External Amplifier",
-		.subdevice = HDA_SUBDEV_NID_FLAG | 0x12,
-		.info = ad198x_eapd_info,
-		.get = ad198x_eapd_get,
-		.put = ad198x_eapd_put,
-		.private_value = 0x12, /* port-D */
-	},
-
-	{ } /* end */
-};
-
-/* capture */
-static const struct snd_kcontrol_new ad1988_capture_mixers[] = {
-	HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x0d, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x0d, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME_IDX("Capture Volume", 2, 0x0e, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE_IDX("Capture Switch", 2, 0x0e, 0x0, HDA_OUTPUT),
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		/* The multiple "Capture Source" controls confuse alsamixer
-		 * So call somewhat different..
-		 */
-		/* .name = "Capture Source", */
-		.name = "Input Source",
-		.count = 3,
-		.info = ad198x_mux_enum_info,
-		.get = ad198x_mux_enum_get,
-		.put = ad198x_mux_enum_put,
-	},
-	{ } /* end */
-};
-
-static int ad1988_spdif_playback_source_info(struct snd_kcontrol *kcontrol,
-					     struct snd_ctl_elem_info *uinfo)
+static int ad1988_auto_smux_enum_info(struct snd_kcontrol *kcontrol,
+				      struct snd_ctl_elem_info *uinfo)
 {
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
 	static const char * const texts[] = {
-		"PCM", "ADC1", "ADC2", "ADC3"
+		"PCM", "ADC1", "ADC2", "ADC3",
 	};
-	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
-	uinfo->count = 1;
-	uinfo->value.enumerated.items = 4;
-	if (uinfo->value.enumerated.item >= 4)
-		uinfo->value.enumerated.item = 3;
-	strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
-	return 0;
+	int num_conns = snd_hda_get_num_conns(codec, 0x0b) + 1;
+	if (num_conns > 4)
+		num_conns = 4;
+	return snd_hda_enum_helper_info(kcontrol, uinfo, num_conns, texts);
 }
 
-static int ad1988_spdif_playback_source_get(struct snd_kcontrol *kcontrol,
-					    struct snd_ctl_elem_value *ucontrol)
+static int ad1988_auto_smux_enum_get(struct snd_kcontrol *kcontrol,
+				     struct snd_ctl_elem_value *ucontrol)
 {
 	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
-	unsigned int sel;
-
-	sel = snd_hda_codec_read(codec, 0x1d, 0, AC_VERB_GET_AMP_GAIN_MUTE,
-				 AC_AMP_GET_INPUT);
-	if (!(sel & 0x80))
-		ucontrol->value.enumerated.item[0] = 0;
-	else {
-		sel = snd_hda_codec_read(codec, 0x0b, 0,
-					 AC_VERB_GET_CONNECT_SEL, 0);
-		if (sel < 3)
-			sel++;
-		else
-			sel = 0;
-		ucontrol->value.enumerated.item[0] = sel;
-	}
+	struct ad198x_spec *spec = codec->spec;
+
+	ucontrol->value.enumerated.item[0] = spec->cur_smux;
 	return 0;
 }
 
-static int ad1988_spdif_playback_source_put(struct snd_kcontrol *kcontrol,
-					    struct snd_ctl_elem_value *ucontrol)
+static int ad1988_auto_smux_enum_put(struct snd_kcontrol *kcontrol,
+				     struct snd_ctl_elem_value *ucontrol)
 {
 	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
-	unsigned int val, sel;
-	int change;
+	struct ad198x_spec *spec = codec->spec;
+	unsigned int val = ucontrol->value.enumerated.item[0];
+	struct nid_path *path;
+	int num_conns = snd_hda_get_num_conns(codec, 0x0b) + 1;
 
-	val = ucontrol->value.enumerated.item[0];
-	if (val > 3)
+	if (val >= num_conns)
 		return -EINVAL;
-	if (!val) {
-		sel = snd_hda_codec_read(codec, 0x1d, 0,
-					 AC_VERB_GET_AMP_GAIN_MUTE,
-					 AC_AMP_GET_INPUT);
-		change = sel & 0x80;
-		if (change) {
-			snd_hda_codec_write_cache(codec, 0x1d, 0,
-						  AC_VERB_SET_AMP_GAIN_MUTE,
-						  AMP_IN_UNMUTE(0));
-			snd_hda_codec_write_cache(codec, 0x1d, 0,
-						  AC_VERB_SET_AMP_GAIN_MUTE,
-						  AMP_IN_MUTE(1));
-		}
-	} else {
-		sel = snd_hda_codec_read(codec, 0x1d, 0,
-					 AC_VERB_GET_AMP_GAIN_MUTE,
-					 AC_AMP_GET_INPUT | 0x01);
-		change = sel & 0x80;
-		if (change) {
-			snd_hda_codec_write_cache(codec, 0x1d, 0,
-						  AC_VERB_SET_AMP_GAIN_MUTE,
-						  AMP_IN_MUTE(0));
-			snd_hda_codec_write_cache(codec, 0x1d, 0,
-						  AC_VERB_SET_AMP_GAIN_MUTE,
-						  AMP_IN_UNMUTE(1));
-		}
-		sel = snd_hda_codec_read(codec, 0x0b, 0,
-					 AC_VERB_GET_CONNECT_SEL, 0) + 1;
-		change |= sel != val;
-		if (change)
-			snd_hda_codec_write_cache(codec, 0x0b, 0,
-						  AC_VERB_SET_CONNECT_SEL,
-						  val - 1);
-	}
-	return change;
-}
+	if (spec->cur_smux == val)
+		return 0;
 
-static const struct snd_kcontrol_new ad1988_spdif_out_mixers[] = {
-	HDA_CODEC_VOLUME("IEC958 Playback Volume", 0x1b, 0x0, HDA_OUTPUT),
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		.name = "IEC958 Playback Source",
-		.subdevice = HDA_SUBDEV_NID_FLAG | 0x1b,
-		.info = ad1988_spdif_playback_source_info,
-		.get = ad1988_spdif_playback_source_get,
-		.put = ad1988_spdif_playback_source_put,
-	},
-	{ } /* end */
-};
+	mutex_lock(&codec->control_mutex);
+	codec->cached_write = 1;
+	path = snd_hda_get_path_from_idx(codec,
+					 spec->smux_paths[spec->cur_smux]);
+	if (path)
+		snd_hda_activate_path(codec, path, false, true);
+	path = snd_hda_get_path_from_idx(codec, spec->smux_paths[val]);
+	if (path)
+		snd_hda_activate_path(codec, path, true, true);
+	spec->cur_smux = val;
+	codec->cached_write = 0;
+	mutex_unlock(&codec->control_mutex);
+	snd_hda_codec_flush_cache(codec); /* flush the updates */
+	return 1;
+}
 
-static const struct snd_kcontrol_new ad1988_spdif_in_mixers[] = {
-	HDA_CODEC_VOLUME("IEC958 Capture Volume", 0x1c, 0x0, HDA_INPUT),
-	{ } /* end */
+static struct snd_kcontrol_new ad1988_auto_smux_mixer = {
+	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+	.name = "IEC958 Playback Source",
+	.info = ad1988_auto_smux_enum_info,
+	.get = ad1988_auto_smux_enum_get,
+	.put = ad1988_auto_smux_enum_put,
 };
 
-static const struct snd_kcontrol_new ad1989_spdif_out_mixers[] = {
-	HDA_CODEC_VOLUME("IEC958 Playback Volume", 0x1b, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("HDMI Playback Volume", 0x1d, 0x0, HDA_OUTPUT),
-	{ } /* end */
-};
-
-/*
- * initialization verbs
- */
-
-/*
- * for 6-stack (+dig)
- */
-static const struct hda_verb ad1988_6stack_init_verbs[] = {
-	/* Front, Surround, CLFE, side DAC; unmute as default */
-	{0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	{0x06, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	{0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	/* Port-A front headphon path */
-	{0x37, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC0:03h */
-	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
-	{0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	{0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
-	/* Port-D line-out path */
-	{0x29, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x29, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
-	{0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
-	/* Port-F surround path */
-	{0x2a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x2a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
-	{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
-	/* Port-G CLFE path */
-	{0x27, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x27, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
-	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	{0x24, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
-	/* Port-H side path */
-	{0x28, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x28, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
-	{0x25, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	{0x25, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
-	/* Mono out path */
-	{0x36, AC_VERB_SET_CONNECT_SEL, 0x1}, /* DAC1:04h */
-	{0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
-	{0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
-	{0x13, AC_VERB_SET_AMP_GAIN_MUTE, 0xb01f}, /* unmute, 0dB */
-	/* Port-B front mic-in path */
-	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
-	{0x39, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
-	/* Port-C line-in path */
-	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
-	{0x3a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
-	{0x33, AC_VERB_SET_CONNECT_SEL, 0x0},
-	/* Port-E mic-in path */
-	{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	{0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
-	{0x3c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
-	{0x34, AC_VERB_SET_CONNECT_SEL, 0x0},
-	/* Analog CD Input */
-	{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
-	/* Analog Mix output amp */
-	{0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x1f}, /* 0dB */
-
-	{ }
-};
-
-static const struct hda_verb ad1988_6stack_fp_init_verbs[] = {
-	/* Headphone; unmute as default */
-	{0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	/* Port-A front headphon path */
-	{0x37, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC0:03h */
-	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
-	{0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	{0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
-
-	{ }
-};
-
-static const struct hda_verb ad1988_capture_init_verbs[] = {
-	/* mute analog mix */
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)},
-	/* select ADCs - front-mic */
-	{0x0c, AC_VERB_SET_CONNECT_SEL, 0x1},
-	{0x0d, AC_VERB_SET_CONNECT_SEL, 0x1},
-	{0x0e, AC_VERB_SET_CONNECT_SEL, 0x1},
-
-	{ }
-};
-
-static const struct hda_verb ad1988_spdif_init_verbs[] = {
-	/* SPDIF out sel */
-	{0x02, AC_VERB_SET_CONNECT_SEL, 0x0}, /* PCM */
-	{0x0b, AC_VERB_SET_CONNECT_SEL, 0x0}, /* ADC1 */
-	{0x1d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	{0x1d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
-	/* SPDIF out pin */
-	{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x27}, /* 0dB */
-
-	{ }
-};
-
-static const struct hda_verb ad1988_spdif_in_init_verbs[] = {
-	/* unmute SPDIF input pin */
-	{0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	{ }
-};
-
-/* AD1989 has no ADC -> SPDIF route */
-static const struct hda_verb ad1989_spdif_init_verbs[] = {
-	/* SPDIF-1 out pin */
-	{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
-	{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x27}, /* 0dB */
-	/* SPDIF-2/HDMI out pin */
-	{0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
-	{0x1d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x27}, /* 0dB */
-	{ }
-};
-
-/*
- * verbs for 3stack (+dig)
- */
-static const struct hda_verb ad1988_3stack_ch2_init[] = {
-	/* set port-C to line-in */
-	{ 0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
-	{ 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },
-	/* set port-E to mic-in */
-	{ 0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
-	{ 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
-	{ } /* end */
-};
-
-static const struct hda_verb ad1988_3stack_ch6_init[] = {
-	/* set port-C to surround out */
-	{ 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
-	{ 0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
-	/* set port-E to CLFE out */
-	{ 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
-	{ 0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
-	{ } /* end */
-};
-
-static const struct hda_channel_mode ad1988_3stack_modes[2] = {
-	{ 2, ad1988_3stack_ch2_init },
-	{ 6, ad1988_3stack_ch6_init },
-};
-
-static const struct hda_verb ad1988_3stack_init_verbs[] = {
-	/* Front, Surround, CLFE, side DAC; unmute as default */
-	{0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	{0x06, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	{0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	/* Port-A front headphon path */
-	{0x37, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC0:03h */
-	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
-	{0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	{0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
-	/* Port-D line-out path */
-	{0x29, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x29, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
-	{0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
-	/* Mono out path */
-	{0x36, AC_VERB_SET_CONNECT_SEL, 0x1}, /* DAC1:04h */
-	{0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
-	{0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
-	{0x13, AC_VERB_SET_AMP_GAIN_MUTE, 0xb01f}, /* unmute, 0dB */
-	/* Port-B front mic-in path */
-	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
-	{0x39, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
-	/* Port-C line-in/surround path - 6ch mode as default */
-	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
-	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	{0x3a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
-	{0x31, AC_VERB_SET_CONNECT_SEL, 0x0}, /* output sel: DAC 0x05 */
-	{0x33, AC_VERB_SET_CONNECT_SEL, 0x0},
-	/* Port-E mic-in/CLFE path - 6ch mode as default */
-	{0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
-	{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	{0x3c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
-	{0x32, AC_VERB_SET_CONNECT_SEL, 0x1}, /* output sel: DAC 0x0a */
-	{0x34, AC_VERB_SET_CONNECT_SEL, 0x0},
-	/* mute analog mix */
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)},
-	/* select ADCs - front-mic */
-	{0x0c, AC_VERB_SET_CONNECT_SEL, 0x1},
-	{0x0d, AC_VERB_SET_CONNECT_SEL, 0x1},
-	{0x0e, AC_VERB_SET_CONNECT_SEL, 0x1},
-	/* Analog Mix output amp */
-	{0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x1f}, /* 0dB */
-	{ }
-};
-
-/*
- * verbs for laptop mode (+dig)
- */
-static const struct hda_verb ad1988_laptop_hp_on[] = {
-	/* unmute port-A and mute port-D */
-	{ 0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
-	{ 0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
-	{ } /* end */
-};
-static const struct hda_verb ad1988_laptop_hp_off[] = {
-	/* mute port-A and unmute port-D */
-	{ 0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
-	{ 0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
-	{ } /* end */
-};
-
-#define AD1988_HP_EVENT	0x01
-
-static const struct hda_verb ad1988_laptop_init_verbs[] = {
-	/* Front, Surround, CLFE, side DAC; unmute as default */
-	{0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	{0x06, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	{0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	/* Port-A front headphon path */
-	{0x37, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC0:03h */
-	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
-	{0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	{0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
-	/* unsolicited event for pin-sense */
-	{0x11, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1988_HP_EVENT },
-	/* Port-D line-out path + EAPD */
-	{0x29, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x29, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
-	{0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
-	{0x12, AC_VERB_SET_EAPD_BTLENABLE, 0x00}, /* EAPD-off */
-	/* Mono out path */
-	{0x36, AC_VERB_SET_CONNECT_SEL, 0x1}, /* DAC1:04h */
-	{0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
-	{0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
-	{0x13, AC_VERB_SET_AMP_GAIN_MUTE, 0xb01f}, /* unmute, 0dB */
-	/* Port-B mic-in path */
-	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
-	{0x39, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
-	/* Port-C docking station - try to output */
-	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
-	{0x3a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
-	{0x33, AC_VERB_SET_CONNECT_SEL, 0x0},
-	/* mute analog mix */
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)},
-	/* select ADCs - mic */
-	{0x0c, AC_VERB_SET_CONNECT_SEL, 0x1},
-	{0x0d, AC_VERB_SET_CONNECT_SEL, 0x1},
-	{0x0e, AC_VERB_SET_CONNECT_SEL, 0x1},
-	/* Analog Mix output amp */
-	{0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x1f}, /* 0dB */
-	{ }
-};
-
-static void ad1988_laptop_unsol_event(struct hda_codec *codec, unsigned int res)
-{
-	if ((res >> 26) != AD1988_HP_EVENT)
-		return;
-	if (snd_hda_jack_detect(codec, 0x11))
-		snd_hda_sequence_write(codec, ad1988_laptop_hp_on);
-	else
-		snd_hda_sequence_write(codec, ad1988_laptop_hp_off);
-} 
-
-#ifdef CONFIG_PM
-static const struct hda_amp_list ad1988_loopbacks[] = {
-	{ 0x20, HDA_INPUT, 0 }, /* Front Mic */
-	{ 0x20, HDA_INPUT, 1 }, /* Line */
-	{ 0x20, HDA_INPUT, 4 }, /* Mic */
-	{ 0x20, HDA_INPUT, 6 }, /* CD */
-	{ } /* end */
-};
-#endif
-#endif /* ENABLE_AD_STATIC_QUIRKS */
-
-static int ad1988_auto_smux_enum_info(struct snd_kcontrol *kcontrol,
-				      struct snd_ctl_elem_info *uinfo)
-{
-	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
-	static const char * const texts[] = {
-		"PCM", "ADC1", "ADC2", "ADC3",
-	};
-	int num_conns = snd_hda_get_num_conns(codec, 0x0b) + 1;
-	if (num_conns > 4)
-		num_conns = 4;
-	return snd_hda_enum_helper_info(kcontrol, uinfo, num_conns, texts);
-}
-
-static int ad1988_auto_smux_enum_get(struct snd_kcontrol *kcontrol,
-				     struct snd_ctl_elem_value *ucontrol)
-{
-	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
-	struct ad198x_spec *spec = codec->spec;
-
-	ucontrol->value.enumerated.item[0] = spec->cur_smux;
-	return 0;
-}
-
-static int ad1988_auto_smux_enum_put(struct snd_kcontrol *kcontrol,
-				     struct snd_ctl_elem_value *ucontrol)
-{
-	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
-	struct ad198x_spec *spec = codec->spec;
-	unsigned int val = ucontrol->value.enumerated.item[0];
-	struct nid_path *path;
-	int num_conns = snd_hda_get_num_conns(codec, 0x0b) + 1;
-
-	if (val >= num_conns)
-		return -EINVAL;
-	if (spec->cur_smux == val)
-		return 0;
-
-	mutex_lock(&codec->control_mutex);
-	codec->cached_write = 1;
-	path = snd_hda_get_path_from_idx(codec,
-					 spec->smux_paths[spec->cur_smux]);
-	if (path)
-		snd_hda_activate_path(codec, path, false, true);
-	path = snd_hda_get_path_from_idx(codec, spec->smux_paths[val]);
-	if (path)
-		snd_hda_activate_path(codec, path, true, true);
-	spec->cur_smux = val;
-	codec->cached_write = 0;
-	mutex_unlock(&codec->control_mutex);
-	snd_hda_codec_flush_cache(codec); /* flush the updates */
-	return 1;
-}
-
-static struct snd_kcontrol_new ad1988_auto_smux_mixer = {
-	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-	.name = "IEC958 Playback Source",
-	.info = ad1988_auto_smux_enum_info,
-	.get = ad1988_auto_smux_enum_get,
-	.put = ad1988_auto_smux_enum_put,
-};
-
-static int ad1988_auto_init(struct hda_codec *codec)
-{
-	struct ad198x_spec *spec = codec->spec;
-	int i, err;
+static int ad1988_auto_init(struct hda_codec *codec)
+{
+	struct ad198x_spec *spec = codec->spec;
+	int i, err;
 
 	err = snd_hda_gen_init(codec);
 	if (err < 0)
@@ -3220,7 +828,34 @@ static int ad1988_add_spdif_mux_ctl(struct hda_codec *codec)
 /*
  */
 
-static int ad1988_parse_auto_config(struct hda_codec *codec)
+enum {
+	AD1988_FIXUP_6STACK_DIG,
+};
+
+static const struct hda_fixup ad1988_fixups[] = {
+	[AD1988_FIXUP_6STACK_DIG] = {
+		.type = HDA_FIXUP_PINS,
+		.v.pins = (const struct hda_pintbl[]) {
+			{ 0x11, 0x02214130 }, /* front-hp */
+			{ 0x12, 0x01014010 }, /* line-out */
+			{ 0x14, 0x02a19122 }, /* front-mic */
+			{ 0x15, 0x01813021 }, /* line-in */
+			{ 0x16, 0x01011012 }, /* line-out */
+			{ 0x17, 0x01a19020 }, /* mic */
+			{ 0x1b, 0x0145f1f0 }, /* SPDIF */
+			{ 0x24, 0x01016011 }, /* line-out */
+			{ 0x25, 0x01012013 }, /* line-out */
+			{ }
+		}
+	},
+};
+
+static const struct hda_model_fixup ad1988_fixup_models[] = {
+	{ .id = AD1988_FIXUP_6STACK_DIG, .name = "6stack-dig" },
+	{}
+};
+
+static int patch_ad1988(struct hda_codec *codec)
 {
 	struct ad198x_spec *spec;
 	int err;
@@ -3234,12 +869,19 @@ static int ad1988_parse_auto_config(struct hda_codec *codec)
 	spec->gen.mixer_merge_nid = 0x21;
 	spec->gen.beep_nid = 0x10;
 	set_beep_amp(spec, 0x10, 0, HDA_OUTPUT);
+
+	snd_hda_pick_fixup(codec, ad1988_fixup_models, NULL, ad1988_fixups);
+	snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE);
+
 	err = ad198x_parse_auto_config(codec);
 	if (err < 0)
 		goto error;
 	err = ad1988_add_spdif_mux_ctl(codec);
 	if (err < 0)
 		goto error;
+
+	snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PROBE);
+
 	return 0;
 
  error:
@@ -3247,169 +889,6 @@ static int ad1988_parse_auto_config(struct hda_codec *codec)
 	return err;
 }
 
-/*
- */
-
-#ifdef ENABLE_AD_STATIC_QUIRKS
-static const char * const ad1988_models[AD1988_MODEL_LAST] = {
-	[AD1988_6STACK]		= "6stack",
-	[AD1988_6STACK_DIG]	= "6stack-dig",
-	[AD1988_3STACK]		= "3stack",
-	[AD1988_3STACK_DIG]	= "3stack-dig",
-	[AD1988_LAPTOP]		= "laptop",
-	[AD1988_LAPTOP_DIG]	= "laptop-dig",
-	[AD1988_AUTO]		= "auto",
-};
-
-static const struct snd_pci_quirk ad1988_cfg_tbl[] = {
-	SND_PCI_QUIRK(0x1043, 0x81ec, "Asus P5B-DLX", AD1988_6STACK_DIG),
-	SND_PCI_QUIRK(0x1043, 0x81f6, "Asus M2N-SLI", AD1988_6STACK_DIG),
-	SND_PCI_QUIRK(0x1043, 0x8277, "Asus P5K-E/WIFI-AP", AD1988_6STACK_DIG),
-	SND_PCI_QUIRK(0x1043, 0x82c0, "Asus M3N-HT Deluxe", AD1988_6STACK_DIG),
-	SND_PCI_QUIRK(0x1043, 0x8311, "Asus P5Q-Premium/Pro", AD1988_6STACK_DIG),
-	{}
-};
-
-static int patch_ad1988(struct hda_codec *codec)
-{
-	struct ad198x_spec *spec;
-	int err, board_config;
-
-	board_config = snd_hda_check_board_config(codec, AD1988_MODEL_LAST,
-						  ad1988_models, ad1988_cfg_tbl);
-	if (board_config < 0) {
-		printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n",
-		       codec->chip_name);
-		board_config = AD1988_AUTO;
-	}
-
-	if (board_config == AD1988_AUTO)
-		return ad1988_parse_auto_config(codec);
-
-	err = alloc_ad_spec(codec);
-	if (err < 0)
-		return err;
-	spec = codec->spec;
-
-	if (is_rev2(codec))
-		snd_printk(KERN_INFO "patch_analog: AD1988A rev.2 is detected, enable workarounds\n");
-
-	err = snd_hda_attach_beep_device(codec, 0x10);
-	if (err < 0) {
-		ad198x_free(codec);
-		return err;
-	}
-	set_beep_amp(spec, 0x10, 0, HDA_OUTPUT);
-
-	if (!spec->multiout.hp_nid)
-		spec->multiout.hp_nid = ad1988_alt_dac_nid[0];
-	switch (board_config) {
-	case AD1988_6STACK:
-	case AD1988_6STACK_DIG:
-		spec->multiout.max_channels = 8;
-		spec->multiout.num_dacs = 4;
-		if (is_rev2(codec))
-			spec->multiout.dac_nids = ad1988_6stack_dac_nids_rev2;
-		else
-			spec->multiout.dac_nids = ad1988_6stack_dac_nids;
-		spec->input_mux = &ad1988_6stack_capture_source;
-		spec->num_mixers = 2;
-		if (is_rev2(codec))
-			spec->mixers[0] = ad1988_6stack_mixers1_rev2;
-		else
-			spec->mixers[0] = ad1988_6stack_mixers1;
-		spec->mixers[1] = ad1988_6stack_mixers2;
-		spec->num_init_verbs = 1;
-		spec->init_verbs[0] = ad1988_6stack_init_verbs;
-		if (board_config == AD1988_6STACK_DIG) {
-			spec->multiout.dig_out_nid = AD1988_SPDIF_OUT;
-			spec->dig_in_nid = AD1988_SPDIF_IN;
-		}
-		break;
-	case AD1988_3STACK:
-	case AD1988_3STACK_DIG:
-		spec->multiout.max_channels = 6;
-		spec->multiout.num_dacs = 3;
-		if (is_rev2(codec))
-			spec->multiout.dac_nids = ad1988_3stack_dac_nids_rev2;
-		else
-			spec->multiout.dac_nids = ad1988_3stack_dac_nids;
-		spec->input_mux = &ad1988_6stack_capture_source;
-		spec->channel_mode = ad1988_3stack_modes;
-		spec->num_channel_mode = ARRAY_SIZE(ad1988_3stack_modes);
-		spec->num_mixers = 2;
-		if (is_rev2(codec))
-			spec->mixers[0] = ad1988_3stack_mixers1_rev2;
-		else
-			spec->mixers[0] = ad1988_3stack_mixers1;
-		spec->mixers[1] = ad1988_3stack_mixers2;
-		spec->num_init_verbs = 1;
-		spec->init_verbs[0] = ad1988_3stack_init_verbs;
-		if (board_config == AD1988_3STACK_DIG)
-			spec->multiout.dig_out_nid = AD1988_SPDIF_OUT;
-		break;
-	case AD1988_LAPTOP:
-	case AD1988_LAPTOP_DIG:
-		spec->multiout.max_channels = 2;
-		spec->multiout.num_dacs = 1;
-		spec->multiout.dac_nids = ad1988_3stack_dac_nids;
-		spec->input_mux = &ad1988_laptop_capture_source;
-		spec->num_mixers = 1;
-		spec->mixers[0] = ad1988_laptop_mixers;
-		codec->inv_eapd = 1; /* inverted EAPD */
-		spec->num_init_verbs = 1;
-		spec->init_verbs[0] = ad1988_laptop_init_verbs;
-		if (board_config == AD1988_LAPTOP_DIG)
-			spec->multiout.dig_out_nid = AD1988_SPDIF_OUT;
-		break;
-	}
-
-	spec->num_adc_nids = ARRAY_SIZE(ad1988_adc_nids);
-	spec->adc_nids = ad1988_adc_nids;
-	spec->capsrc_nids = ad1988_capsrc_nids;
-	spec->mixers[spec->num_mixers++] = ad1988_capture_mixers;
-	spec->init_verbs[spec->num_init_verbs++] = ad1988_capture_init_verbs;
-	if (spec->multiout.dig_out_nid) {
-		if (codec->vendor_id >= 0x11d4989a) {
-			spec->mixers[spec->num_mixers++] =
-				ad1989_spdif_out_mixers;
-			spec->init_verbs[spec->num_init_verbs++] =
-				ad1989_spdif_init_verbs;
-			codec->slave_dig_outs = ad1989b_slave_dig_outs;
-		} else {
-			spec->mixers[spec->num_mixers++] =
-				ad1988_spdif_out_mixers;
-			spec->init_verbs[spec->num_init_verbs++] =
-				ad1988_spdif_init_verbs;
-		}
-	}
-	if (spec->dig_in_nid && codec->vendor_id < 0x11d4989a) {
-		spec->mixers[spec->num_mixers++] = ad1988_spdif_in_mixers;
-		spec->init_verbs[spec->num_init_verbs++] =
-			ad1988_spdif_in_init_verbs;
-	}
-
-	codec->patch_ops = ad198x_patch_ops;
-	switch (board_config) {
-	case AD1988_LAPTOP:
-	case AD1988_LAPTOP_DIG:
-		codec->patch_ops.unsol_event = ad1988_laptop_unsol_event;
-		break;
-	}
-#ifdef CONFIG_PM
-	spec->loopback.amplist = ad1988_loopbacks;
-#endif
-	spec->vmaster_nid = 0x04;
-
-	codec->no_trigger_sense = 1;
-	codec->no_sticky_stream = 1;
-
-	return 0;
-}
-#else /* ENABLE_AD_STATIC_QUIRKS */
-#define patch_ad1988	ad1988_parse_auto_config
-#endif /* ENABLE_AD_STATIC_QUIRKS */
-
 
 /*
  * AD1884 / AD1984
@@ -3423,167 +902,19 @@ static int patch_ad1988(struct hda_codec *codec)
  *
  * AD1984 = AD1884 + two digital mic-ins
  *
- * FIXME:
- * For simplicity, we share the single DAC for both HP and line-outs
- * right now.  The inidividual playbacks could be easily implemented,
- * but no build-up framework is given, so far.
- */
-
-#ifdef ENABLE_AD_STATIC_QUIRKS
-static const hda_nid_t ad1884_dac_nids[1] = {
-	0x04,
-};
-
-static const hda_nid_t ad1884_adc_nids[2] = {
-	0x08, 0x09,
-};
-
-static const hda_nid_t ad1884_capsrc_nids[2] = {
-	0x0c, 0x0d,
-};
-
-#define AD1884_SPDIF_OUT	0x02
-
-static const struct hda_input_mux ad1884_capture_source = {
-	.num_items = 4,
-	.items = {
-		{ "Front Mic", 0x0 },
-		{ "Mic", 0x1 },
-		{ "CD", 0x2 },
-		{ "Mix", 0x3 },
-	},
-};
-
-static const struct snd_kcontrol_new ad1884_base_mixers[] = {
-	HDA_CODEC_VOLUME("PCM Playback Volume", 0x04, 0x0, HDA_OUTPUT),
-	/* HDA_CODEC_VOLUME_IDX("PCM Playback Volume", 1, 0x03, 0x0, HDA_OUTPUT), */
-	HDA_CODEC_MUTE("Headphone Playback Switch", 0x11, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Front Playback Switch", 0x12, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x13, 1, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE_MONO("Mono Playback Switch", 0x13, 1, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x20, 0x00, HDA_INPUT),
-	HDA_CODEC_MUTE("Front Mic Playback Switch", 0x20, 0x00, HDA_INPUT),
-	HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x01, HDA_INPUT),
-	HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x01, HDA_INPUT),
-	HDA_CODEC_VOLUME("CD Playback Volume", 0x20, 0x02, HDA_INPUT),
-	HDA_CODEC_MUTE("CD Playback Switch", 0x20, 0x02, HDA_INPUT),
-	HDA_CODEC_VOLUME("Mic Boost Volume", 0x15, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x14, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x0d, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x0d, 0x0, HDA_OUTPUT),
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		/* The multiple "Capture Source" controls confuse alsamixer
-		 * So call somewhat different..
-		 */
-		/* .name = "Capture Source", */
-		.name = "Input Source",
-		.count = 2,
-		.info = ad198x_mux_enum_info,
-		.get = ad198x_mux_enum_get,
-		.put = ad198x_mux_enum_put,
-	},
-	/* SPDIF controls */
-	HDA_CODEC_VOLUME("IEC958 Playback Volume", 0x1b, 0x0, HDA_OUTPUT),
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		.name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,NONE) "Source",
-		/* identical with ad1983 */
-		.info = ad1983_spdif_route_info,
-		.get = ad1983_spdif_route_get,
-		.put = ad1983_spdif_route_put,
-	},
-	{ } /* end */
-};
-
-static const struct snd_kcontrol_new ad1984_dmic_mixers[] = {
-	HDA_CODEC_VOLUME("Digital Mic Capture Volume", 0x05, 0x0, HDA_INPUT),
-	HDA_CODEC_MUTE("Digital Mic Capture Switch", 0x05, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME_IDX("Digital Mic Capture Volume", 1, 0x06, 0x0,
-			     HDA_INPUT),
-	HDA_CODEC_MUTE_IDX("Digital Mic Capture Switch", 1, 0x06, 0x0,
-			   HDA_INPUT),
-	{ } /* end */
-};
-
-/*
- * initialization verbs
+ * AD1883 / AD1884A / AD1984A / AD1984B
+ *
+ * port-B (0x14) - front mic-in
+ * port-E (0x1c) - rear mic-in
+ * port-F (0x16) - CD / ext out
+ * port-C (0x15) - rear line-in
+ * port-D (0x12) - rear line-out
+ * port-A (0x11) - front hp-out
+ *
+ * AD1984A = AD1884A + digital-mic
+ * AD1883 = equivalent with AD1984A
+ * AD1984B = AD1984A + extra SPDIF-out
  */
-static const struct hda_verb ad1884_init_verbs[] = {
-	/* DACs; mute as default */
-	{0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
-	{0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
-	/* Port-A (HP) mixer */
-	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-	/* Port-A pin */
-	{0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
-	{0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	/* HP selector - select DAC2 */
-	{0x22, AC_VERB_SET_CONNECT_SEL, 0x1},
-	/* Port-D (Line-out) mixer */
-	{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-	/* Port-D pin */
-	{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
-	{0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	/* Mono-out mixer */
-	{0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	{0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-	/* Mono-out pin */
-	{0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
-	{0x13, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	/* Mono selector */
-	{0x0e, AC_VERB_SET_CONNECT_SEL, 0x1},
-	/* Port-B (front mic) pin */
-	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
-	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	/* Port-C (rear mic) pin */
-	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
-	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	/* Analog mixer; mute as default */
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
-	/* Analog Mix output amp */
-	{0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x1f}, /* 0dB */
-	/* SPDIF output selector */
-	{0x02, AC_VERB_SET_CONNECT_SEL, 0x0}, /* PCM */
-	{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x27}, /* 0dB */
-	{ } /* end */
-};
-
-#ifdef CONFIG_PM
-static const struct hda_amp_list ad1884_loopbacks[] = {
-	{ 0x20, HDA_INPUT, 0 }, /* Front Mic */
-	{ 0x20, HDA_INPUT, 1 }, /* Mic */
-	{ 0x20, HDA_INPUT, 2 }, /* CD */
-	{ 0x20, HDA_INPUT, 4 }, /* Docking */
-	{ } /* end */
-};
-#endif
-
-static const char * const ad1884_slave_vols[] = {
-	"PCM", "Mic", "Mono", "Front Mic", "Mic", "CD",
-	"Internal Mic", "Dock Mic", /* "Beep", */ "IEC958",
-	NULL
-};
-
-enum {
-	AD1884_AUTO,
-	AD1884_BASIC,
-	AD1884_MODELS
-};
-
-static const char * const ad1884_models[AD1884_MODELS] = {
-	[AD1884_AUTO]		= "auto",
-	[AD1884_BASIC]		= "basic",
-};
-#endif /* ENABLE_AD_STATIC_QUIRKS */
-
 
 /* set the upper-limit for mixer amp to 0dB for avoiding the possible
  * damage by overloading
@@ -3599,14 +930,34 @@ static void ad1884_fixup_amp_override(struct hda_codec *codec,
 					  (1 << AC_AMPCAP_MUTE_SHIFT));
 }
 
+/* toggle GPIO1 according to the mute state */
+static void ad1884_vmaster_hp_gpio_hook(void *private_data, int enabled)
+{
+	struct hda_codec *codec = private_data;
+	struct ad198x_spec *spec = codec->spec;
+
+	if (spec->eapd_nid)
+		ad_vmaster_eapd_hook(private_data, enabled);
+	snd_hda_codec_update_cache(codec, 0x01, 0,
+				   AC_VERB_SET_GPIO_DATA,
+				   enabled ? 0x00 : 0x02);
+}
+
 static void ad1884_fixup_hp_eapd(struct hda_codec *codec,
 				 const struct hda_fixup *fix, int action)
 {
 	struct ad198x_spec *spec = codec->spec;
+	static const struct hda_verb gpio_init_verbs[] = {
+		{0x01, AC_VERB_SET_GPIO_MASK, 0x02},
+		{0x01, AC_VERB_SET_GPIO_DIRECTION, 0x02},
+		{0x01, AC_VERB_SET_GPIO_DATA, 0x02},
+		{},
+	};
 
 	switch (action) {
 	case HDA_FIXUP_ACT_PRE_PROBE:
-		spec->gen.vmaster_mute.hook = ad_vmaster_eapd_hook;
+		spec->gen.vmaster_mute.hook = ad1884_vmaster_hp_gpio_hook;
+		snd_hda_sequence_write_cache(codec, gpio_init_verbs);
 		break;
 	case HDA_FIXUP_ACT_PROBE:
 		if (spec->gen.autocfg.line_out_type == AUTO_PIN_SPEAKER_OUT)
@@ -3617,9 +968,18 @@ static void ad1884_fixup_hp_eapd(struct hda_codec *codec,
 	}
 }
 
+/* set magic COEFs for dmic */
+static const struct hda_verb ad1884_dmic_init_verbs[] = {
+	{0x01, AC_VERB_SET_COEF_INDEX, 0x13f7},
+	{0x01, AC_VERB_SET_PROC_COEF, 0x08},
+	{}
+};
+
 enum {
 	AD1884_FIXUP_AMP_OVERRIDE,
 	AD1884_FIXUP_HP_EAPD,
+	AD1884_FIXUP_DMIC_COEF,
+	AD1884_FIXUP_HP_TOUCHSMART,
 };
 
 static const struct hda_fixup ad1884_fixups[] = {
@@ -3633,15 +993,27 @@ static const struct hda_fixup ad1884_fixups[] = {
 		.chained = true,
 		.chain_id = AD1884_FIXUP_AMP_OVERRIDE,
 	},
+	[AD1884_FIXUP_DMIC_COEF] = {
+		.type = HDA_FIXUP_VERBS,
+		.v.verbs = ad1884_dmic_init_verbs,
+	},
+	[AD1884_FIXUP_HP_TOUCHSMART] = {
+		.type = HDA_FIXUP_VERBS,
+		.v.verbs = ad1884_dmic_init_verbs,
+		.chained = true,
+		.chain_id = AD1884_FIXUP_HP_EAPD,
+	},
 };
 
 static const struct snd_pci_quirk ad1884_fixup_tbl[] = {
+	SND_PCI_QUIRK(0x103c, 0x2a82, "HP Touchsmart", AD1884_FIXUP_HP_TOUCHSMART),
 	SND_PCI_QUIRK_VENDOR(0x103c, "HP", AD1884_FIXUP_HP_EAPD),
+	SND_PCI_QUIRK_VENDOR(0x17aa, "Lenovo Thinkpad", AD1884_FIXUP_DMIC_COEF),
 	{}
 };
 
 
-static int ad1884_parse_auto_config(struct hda_codec *codec)
+static int patch_ad1884(struct hda_codec *codec)
 {
 	struct ad198x_spec *spec;
 	int err;
@@ -3674,1170 +1046,6 @@ static int ad1884_parse_auto_config(struct hda_codec *codec)
 	return err;
 }
 
-#ifdef ENABLE_AD_STATIC_QUIRKS
-static int patch_ad1884_basic(struct hda_codec *codec)
-{
-	struct ad198x_spec *spec;
-	int err;
-
-	err = alloc_ad_spec(codec);
-	if (err < 0)
-		return err;
-	spec = codec->spec;
-
-	err = snd_hda_attach_beep_device(codec, 0x10);
-	if (err < 0) {
-		ad198x_free(codec);
-		return err;
-	}
-	set_beep_amp(spec, 0x10, 0, HDA_OUTPUT);
-
-	spec->multiout.max_channels = 2;
-	spec->multiout.num_dacs = ARRAY_SIZE(ad1884_dac_nids);
-	spec->multiout.dac_nids = ad1884_dac_nids;
-	spec->multiout.dig_out_nid = AD1884_SPDIF_OUT;
-	spec->num_adc_nids = ARRAY_SIZE(ad1884_adc_nids);
-	spec->adc_nids = ad1884_adc_nids;
-	spec->capsrc_nids = ad1884_capsrc_nids;
-	spec->input_mux = &ad1884_capture_source;
-	spec->num_mixers = 1;
-	spec->mixers[0] = ad1884_base_mixers;
-	spec->num_init_verbs = 1;
-	spec->init_verbs[0] = ad1884_init_verbs;
-	spec->spdif_route = 0;
-#ifdef CONFIG_PM
-	spec->loopback.amplist = ad1884_loopbacks;
-#endif
-	spec->vmaster_nid = 0x04;
-	/* we need to cover all playback volumes */
-	spec->slave_vols = ad1884_slave_vols;
-	/* slaves may contain input volumes, so we can't raise to 0dB blindly */
-	spec->avoid_init_slave_vol = 1;
-
-	codec->patch_ops = ad198x_patch_ops;
-
-	codec->no_trigger_sense = 1;
-	codec->no_sticky_stream = 1;
-
-	return 0;
-}
-
-static int patch_ad1884(struct hda_codec *codec)
-{
-	int board_config;
-
-	board_config = snd_hda_check_board_config(codec, AD1884_MODELS,
-						  ad1884_models, NULL);
-	if (board_config < 0) {
-		printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n",
-		       codec->chip_name);
-		board_config = AD1884_AUTO;
-	}
-
-	if (board_config == AD1884_AUTO)
-		return ad1884_parse_auto_config(codec);
-	else
-		return patch_ad1884_basic(codec);
-}
-#else /* ENABLE_AD_STATIC_QUIRKS */
-#define patch_ad1884	ad1884_parse_auto_config
-#endif /* ENABLE_AD_STATIC_QUIRKS */
-
-
-#ifdef ENABLE_AD_STATIC_QUIRKS
-/*
- * Lenovo Thinkpad T61/X61
- */
-static const struct hda_input_mux ad1984_thinkpad_capture_source = {
-	.num_items = 4,
-	.items = {
-		{ "Mic", 0x0 },
-		{ "Internal Mic", 0x1 },
-		{ "Mix", 0x3 },
-		{ "Dock Mic", 0x4 },
-	},
-};
-
-
-/*
- * Dell Precision T3400
- */
-static const struct hda_input_mux ad1984_dell_desktop_capture_source = {
-	.num_items = 3,
-	.items = {
-		{ "Front Mic", 0x0 },
-		{ "Line-In", 0x1 },
-		{ "Mix", 0x3 },
-	},
-};
-
-
-static const struct snd_kcontrol_new ad1984_thinkpad_mixers[] = {
-	HDA_CODEC_VOLUME("PCM Playback Volume", 0x04, 0x0, HDA_OUTPUT),
-	/* HDA_CODEC_VOLUME_IDX("PCM Playback Volume", 1, 0x03, 0x0, HDA_OUTPUT), */
-	HDA_CODEC_MUTE("Headphone Playback Switch", 0x11, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Speaker Playback Switch", 0x12, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x00, HDA_INPUT),
-	HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x00, HDA_INPUT),
-	HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x20, 0x01, HDA_INPUT),
-	HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x20, 0x01, HDA_INPUT),
-	HDA_CODEC_VOLUME("Beep Playback Volume", 0x20, 0x03, HDA_INPUT),
-	HDA_CODEC_MUTE("Beep Playback Switch", 0x20, 0x03, HDA_INPUT),
-	HDA_CODEC_VOLUME("Dock Mic Playback Volume", 0x20, 0x04, HDA_INPUT),
-	HDA_CODEC_MUTE("Dock Mic Playback Switch", 0x20, 0x04, HDA_INPUT),
-	HDA_CODEC_VOLUME("Mic Boost Volume", 0x14, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x15, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME("Dock Mic Boost Volume", 0x25, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x0d, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x0d, 0x0, HDA_OUTPUT),
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		/* The multiple "Capture Source" controls confuse alsamixer
-		 * So call somewhat different..
-		 */
-		/* .name = "Capture Source", */
-		.name = "Input Source",
-		.count = 2,
-		.info = ad198x_mux_enum_info,
-		.get = ad198x_mux_enum_get,
-		.put = ad198x_mux_enum_put,
-	},
-	/* SPDIF controls */
-	HDA_CODEC_VOLUME("IEC958 Playback Volume", 0x1b, 0x0, HDA_OUTPUT),
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		.name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,NONE) "Source",
-		/* identical with ad1983 */
-		.info = ad1983_spdif_route_info,
-		.get = ad1983_spdif_route_get,
-		.put = ad1983_spdif_route_put,
-	},
-	{ } /* end */
-};
-
-/* additional verbs */
-static const struct hda_verb ad1984_thinkpad_init_verbs[] = {
-	/* Port-E (docking station mic) pin */
-	{0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
-	{0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	/* docking mic boost */
-	{0x25, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
-	/* Analog PC Beeper - allow firmware/ACPI beeps */
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3) | 0x1a},
-	/* Analog mixer - docking mic; mute as default */
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
-	/* enable EAPD bit */
-	{0x12, AC_VERB_SET_EAPD_BTLENABLE, 0x02},
-	{ } /* end */
-};
-
-/*
- * Dell Precision T3400
- */
-static const struct snd_kcontrol_new ad1984_dell_desktop_mixers[] = {
-	HDA_CODEC_VOLUME("PCM Playback Volume", 0x04, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Headphone Playback Switch", 0x11, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Speaker Playback Switch", 0x12, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x13, 1, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE_MONO("Mono Playback Switch", 0x13, 1, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x20, 0x00, HDA_INPUT),
-	HDA_CODEC_MUTE("Front Mic Playback Switch", 0x20, 0x00, HDA_INPUT),
-	HDA_CODEC_VOLUME("Line-In Playback Volume", 0x20, 0x01, HDA_INPUT),
-	HDA_CODEC_MUTE("Line-In Playback Switch", 0x20, 0x01, HDA_INPUT),
-	HDA_CODEC_VOLUME("Line-In Boost Volume", 0x15, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x14, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x0d, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x0d, 0x0, HDA_OUTPUT),
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		/* The multiple "Capture Source" controls confuse alsamixer
-		 * So call somewhat different..
-		 */
-		/* .name = "Capture Source", */
-		.name = "Input Source",
-		.count = 2,
-		.info = ad198x_mux_enum_info,
-		.get = ad198x_mux_enum_get,
-		.put = ad198x_mux_enum_put,
-	},
-	{ } /* end */
-};
-
-/* Digial MIC ADC NID 0x05 + 0x06 */
-static int ad1984_pcm_dmic_prepare(struct hda_pcm_stream *hinfo,
-				   struct hda_codec *codec,
-				   unsigned int stream_tag,
-				   unsigned int format,
-				   struct snd_pcm_substream *substream)
-{
-	snd_hda_codec_setup_stream(codec, 0x05 + substream->number,
-				   stream_tag, 0, format);
-	return 0;
-}
-
-static int ad1984_pcm_dmic_cleanup(struct hda_pcm_stream *hinfo,
-				   struct hda_codec *codec,
-				   struct snd_pcm_substream *substream)
-{
-	snd_hda_codec_cleanup_stream(codec, 0x05 + substream->number);
-	return 0;
-}
-
-static const struct hda_pcm_stream ad1984_pcm_dmic_capture = {
-	.substreams = 2,
-	.channels_min = 2,
-	.channels_max = 2,
-	.nid = 0x05,
-	.ops = {
-		.prepare = ad1984_pcm_dmic_prepare,
-		.cleanup = ad1984_pcm_dmic_cleanup
-	},
-};
-
-static int ad1984_build_pcms(struct hda_codec *codec)
-{
-	struct ad198x_spec *spec = codec->spec;
-	struct hda_pcm *info;
-	int err;
-
-	err = ad198x_build_pcms(codec);
-	if (err < 0)
-		return err;
-
-	info = spec->pcm_rec + codec->num_pcms;
-	codec->num_pcms++;
-	info->name = "AD1984 Digital Mic";
-	info->stream[SNDRV_PCM_STREAM_CAPTURE] = ad1984_pcm_dmic_capture;
-	return 0;
-}
-
-/* models */
-enum {
-	AD1984_AUTO,
-	AD1984_BASIC,
-	AD1984_THINKPAD,
-	AD1984_DELL_DESKTOP,
-	AD1984_MODELS
-};
-
-static const char * const ad1984_models[AD1984_MODELS] = {
-	[AD1984_AUTO]		= "auto",
-	[AD1984_BASIC]		= "basic",
-	[AD1984_THINKPAD]	= "thinkpad",
-	[AD1984_DELL_DESKTOP]	= "dell_desktop",
-};
-
-static const struct snd_pci_quirk ad1984_cfg_tbl[] = {
-	/* Lenovo Thinkpad T61/X61 */
-	SND_PCI_QUIRK_VENDOR(0x17aa, "Lenovo Thinkpad", AD1984_THINKPAD),
-	SND_PCI_QUIRK(0x1028, 0x0214, "Dell T3400", AD1984_DELL_DESKTOP),
-	SND_PCI_QUIRK(0x1028, 0x0233, "Dell Latitude E6400", AD1984_DELL_DESKTOP),
-	{}
-};
-
-static int patch_ad1984(struct hda_codec *codec)
-{
-	struct ad198x_spec *spec;
-	int board_config, err;
-
-	board_config = snd_hda_check_board_config(codec, AD1984_MODELS,
-						  ad1984_models, ad1984_cfg_tbl);
-	if (board_config < 0) {
-		printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n",
-		       codec->chip_name);
-		board_config = AD1984_AUTO;
-	}
-
-	if (board_config == AD1984_AUTO)
-		return ad1884_parse_auto_config(codec);
-
-	err = patch_ad1884_basic(codec);
-	if (err < 0)
-		return err;
-	spec = codec->spec;
-
-	switch (board_config) {
-	case AD1984_BASIC:
-		/* additional digital mics */
-		spec->mixers[spec->num_mixers++] = ad1984_dmic_mixers;
-		codec->patch_ops.build_pcms = ad1984_build_pcms;
-		break;
-	case AD1984_THINKPAD:
-		if (codec->subsystem_id == 0x17aa20fb) {
-			/* Thinpad X300 does not have the ability to do SPDIF,
-			   or attach to docking station to use SPDIF */
-			spec->multiout.dig_out_nid = 0;
-		} else
-			spec->multiout.dig_out_nid = AD1884_SPDIF_OUT;
-		spec->input_mux = &ad1984_thinkpad_capture_source;
-		spec->mixers[0] = ad1984_thinkpad_mixers;
-		spec->init_verbs[spec->num_init_verbs++] = ad1984_thinkpad_init_verbs;
-		spec->analog_beep = 1;
-		break;
-	case AD1984_DELL_DESKTOP:
-		spec->multiout.dig_out_nid = 0;
-		spec->input_mux = &ad1984_dell_desktop_capture_source;
-		spec->mixers[0] = ad1984_dell_desktop_mixers;
-		break;
-	}
-	return 0;
-}
-#else /* ENABLE_AD_STATIC_QUIRKS */
-#define patch_ad1984	ad1884_parse_auto_config
-#endif /* ENABLE_AD_STATIC_QUIRKS */
-
-
-/*
- * AD1883 / AD1884A / AD1984A / AD1984B
- *
- * port-B (0x14) - front mic-in
- * port-E (0x1c) - rear mic-in
- * port-F (0x16) - CD / ext out
- * port-C (0x15) - rear line-in
- * port-D (0x12) - rear line-out
- * port-A (0x11) - front hp-out
- *
- * AD1984A = AD1884A + digital-mic
- * AD1883 = equivalent with AD1984A
- * AD1984B = AD1984A + extra SPDIF-out
- *
- * FIXME:
- * We share the single DAC for both HP and line-outs (see AD1884/1984).
- */
-
-#ifdef ENABLE_AD_STATIC_QUIRKS
-static const hda_nid_t ad1884a_dac_nids[1] = {
-	0x03,
-};
-
-#define ad1884a_adc_nids	ad1884_adc_nids
-#define ad1884a_capsrc_nids	ad1884_capsrc_nids
-
-#define AD1884A_SPDIF_OUT	0x02
-
-static const struct hda_input_mux ad1884a_capture_source = {
-	.num_items = 5,
-	.items = {
-		{ "Front Mic", 0x0 },
-		{ "Mic", 0x4 },
-		{ "Line", 0x1 },
-		{ "CD", 0x2 },
-		{ "Mix", 0x3 },
-	},
-};
-
-static const struct snd_kcontrol_new ad1884a_base_mixers[] = {
-	HDA_CODEC_VOLUME("Master Playback Volume", 0x21, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Master Playback Switch", 0x21, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Headphone Playback Switch", 0x11, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Front Playback Switch", 0x12, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x13, 1, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE_MONO("Mono Playback Switch", 0x13, 1, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("PCM Playback Volume", 0x20, 0x5, HDA_INPUT),
-	HDA_CODEC_MUTE("PCM Playback Switch", 0x20, 0x5, HDA_INPUT),
-	HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x20, 0x00, HDA_INPUT),
-	HDA_CODEC_MUTE("Front Mic Playback Switch", 0x20, 0x00, HDA_INPUT),
-	HDA_CODEC_VOLUME("Line Playback Volume", 0x20, 0x01, HDA_INPUT),
-	HDA_CODEC_MUTE("Line Playback Switch", 0x20, 0x01, HDA_INPUT),
-	HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x04, HDA_INPUT),
-	HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x04, HDA_INPUT),
-	HDA_CODEC_VOLUME("CD Playback Volume", 0x20, 0x02, HDA_INPUT),
-	HDA_CODEC_MUTE("CD Playback Switch", 0x20, 0x02, HDA_INPUT),
-	HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x14, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME("Line Boost Volume", 0x15, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME("Mic Boost Volume", 0x25, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x0d, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x0d, 0x0, HDA_OUTPUT),
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		/* The multiple "Capture Source" controls confuse alsamixer
-		 * So call somewhat different..
-		 */
-		/* .name = "Capture Source", */
-		.name = "Input Source",
-		.count = 2,
-		.info = ad198x_mux_enum_info,
-		.get = ad198x_mux_enum_get,
-		.put = ad198x_mux_enum_put,
-	},
-	/* SPDIF controls */
-	HDA_CODEC_VOLUME("IEC958 Playback Volume", 0x1b, 0x0, HDA_OUTPUT),
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		.name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,NONE) "Source",
-		/* identical with ad1983 */
-		.info = ad1983_spdif_route_info,
-		.get = ad1983_spdif_route_get,
-		.put = ad1983_spdif_route_put,
-	},
-	{ } /* end */
-};
-
-/*
- * initialization verbs
- */
-static const struct hda_verb ad1884a_init_verbs[] = {
-	/* DACs; unmute as default */
-	{0x03, AC_VERB_SET_AMP_GAIN_MUTE, 0x27}, /* 0dB */
-	{0x04, AC_VERB_SET_AMP_GAIN_MUTE, 0x27}, /* 0dB */
-	/* Port-A (HP) mixer - route only from analog mixer */
-	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-	/* Port-A pin */
-	{0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
-	{0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	/* Port-D (Line-out) mixer - route only from analog mixer */
-	{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-	/* Port-D pin */
-	{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
-	{0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	/* Mono-out mixer - route only from analog mixer */
-	{0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-	/* Mono-out pin */
-	{0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
-	{0x13, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	/* Port-B (front mic) pin */
-	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
-	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	/* Port-C (rear line-in) pin */
-	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
-	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	/* Port-E (rear mic) pin */
-	{0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
-	{0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	{0x25, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, /* no boost */
-	/* Port-F (CD) pin */
-	{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
-	{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	/* Analog mixer; mute as default */
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)}, /* aux */
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)},
-	/* Analog Mix output amp */
-	{0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	/* capture sources */
-	{0x0c, AC_VERB_SET_CONNECT_SEL, 0x0},
-	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	{0x0d, AC_VERB_SET_CONNECT_SEL, 0x0},
-	{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	/* SPDIF output amp */
-	{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x27}, /* 0dB */
-	{ } /* end */
-};
-
-#ifdef CONFIG_PM
-static const struct hda_amp_list ad1884a_loopbacks[] = {
-	{ 0x20, HDA_INPUT, 0 }, /* Front Mic */
-	{ 0x20, HDA_INPUT, 1 }, /* Mic */
-	{ 0x20, HDA_INPUT, 2 }, /* CD */
-	{ 0x20, HDA_INPUT, 4 }, /* Docking */
-	{ } /* end */
-};
-#endif
-
-/*
- * Laptop model
- *
- * Port A: Headphone jack
- * Port B: MIC jack
- * Port C: Internal MIC
- * Port D: Dock Line Out (if enabled)
- * Port E: Dock Line In (if enabled)
- * Port F: Internal speakers
- */
-
-static int ad1884a_mobile_master_sw_put(struct snd_kcontrol *kcontrol,
-					struct snd_ctl_elem_value *ucontrol)
-{
-	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
-	int ret = snd_hda_mixer_amp_switch_put(kcontrol, ucontrol);
-	int mute = (!ucontrol->value.integer.value[0] &&
-		    !ucontrol->value.integer.value[1]);
-	/* toggle GPIO1 according to the mute state */
-	snd_hda_codec_write_cache(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA,
-			    mute ? 0x02 : 0x0);
-	return ret;
-}
-
-static const struct snd_kcontrol_new ad1884a_laptop_mixers[] = {
-	HDA_CODEC_VOLUME("Master Playback Volume", 0x21, 0x0, HDA_OUTPUT),
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		.name = "Master Playback Switch",
-		.subdevice = HDA_SUBDEV_AMP_FLAG,
-		.info = snd_hda_mixer_amp_switch_info,
-		.get = snd_hda_mixer_amp_switch_get,
-		.put = ad1884a_mobile_master_sw_put,
-		.private_value = HDA_COMPOSE_AMP_VAL(0x21, 3, 0, HDA_OUTPUT),
-	},
-	HDA_CODEC_MUTE("Dock Playback Switch", 0x12, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("PCM Playback Volume", 0x20, 0x5, HDA_INPUT),
-	HDA_CODEC_MUTE("PCM Playback Switch", 0x20, 0x5, HDA_INPUT),
-	HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x00, HDA_INPUT),
-	HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x00, HDA_INPUT),
-	HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x20, 0x01, HDA_INPUT),
-	HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x20, 0x01, HDA_INPUT),
-	HDA_CODEC_VOLUME("Dock Mic Playback Volume", 0x20, 0x04, HDA_INPUT),
-	HDA_CODEC_MUTE("Dock Mic Playback Switch", 0x20, 0x04, HDA_INPUT),
-	HDA_CODEC_VOLUME("Mic Boost Volume", 0x14, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x15, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME("Dock Mic Boost Volume", 0x25, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT),
-	{ } /* end */
-};
-
-static const struct snd_kcontrol_new ad1884a_mobile_mixers[] = {
-	HDA_CODEC_VOLUME("Master Playback Volume", 0x21, 0x0, HDA_OUTPUT),
-	/*HDA_CODEC_MUTE("Master Playback Switch", 0x21, 0x0, HDA_OUTPUT),*/
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		.name = "Master Playback Switch",
-		.subdevice = HDA_SUBDEV_AMP_FLAG,
-		.info = snd_hda_mixer_amp_switch_info,
-		.get = snd_hda_mixer_amp_switch_get,
-		.put = ad1884a_mobile_master_sw_put,
-		.private_value = HDA_COMPOSE_AMP_VAL(0x21, 3, 0, HDA_OUTPUT),
-	},
-	HDA_CODEC_VOLUME("PCM Playback Volume", 0x20, 0x5, HDA_INPUT),
-	HDA_CODEC_MUTE("PCM Playback Switch", 0x20, 0x5, HDA_INPUT),
-	HDA_CODEC_VOLUME("Mic Capture Volume", 0x14, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME("Internal Mic Capture Volume", 0x15, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT),
-	{ } /* end */
-};
-
-/* mute internal speaker if HP is plugged */
-static void ad1884a_hp_automute(struct hda_codec *codec)
-{
-	unsigned int present;
-
-	present = snd_hda_jack_detect(codec, 0x11);
-	snd_hda_codec_amp_stereo(codec, 0x16, HDA_OUTPUT, 0,
-				 HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
-	snd_hda_codec_write(codec, 0x16, 0, AC_VERB_SET_EAPD_BTLENABLE,
-			    present ? 0x00 : 0x02);
-}
-
-/* switch to external mic if plugged */
-static void ad1884a_hp_automic(struct hda_codec *codec)
-{
-	unsigned int present;
-
-	present = snd_hda_jack_detect(codec, 0x14);
-	snd_hda_codec_write(codec, 0x0c, 0, AC_VERB_SET_CONNECT_SEL,
-			    present ? 0 : 1);
-}
-
-#define AD1884A_HP_EVENT		0x37
-#define AD1884A_MIC_EVENT		0x36
-
-/* unsolicited event for HP jack sensing */
-static void ad1884a_hp_unsol_event(struct hda_codec *codec, unsigned int res)
-{
-	switch (res >> 26) {
-	case AD1884A_HP_EVENT:
-		ad1884a_hp_automute(codec);
-		break;
-	case AD1884A_MIC_EVENT:
-		ad1884a_hp_automic(codec);
-		break;
-	}
-}
-
-/* initialize jack-sensing, too */
-static int ad1884a_hp_init(struct hda_codec *codec)
-{
-	ad198x_init(codec);
-	ad1884a_hp_automute(codec);
-	ad1884a_hp_automic(codec);
-	return 0;
-}
-
-/* mute internal speaker if HP or docking HP is plugged */
-static void ad1884a_laptop_automute(struct hda_codec *codec)
-{
-	unsigned int present;
-
-	present = snd_hda_jack_detect(codec, 0x11);
-	if (!present)
-		present = snd_hda_jack_detect(codec, 0x12);
-	snd_hda_codec_amp_stereo(codec, 0x16, HDA_OUTPUT, 0,
-				 HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
-	snd_hda_codec_write(codec, 0x16, 0, AC_VERB_SET_EAPD_BTLENABLE,
-			    present ? 0x00 : 0x02);
-}
-
-/* switch to external mic if plugged */
-static void ad1884a_laptop_automic(struct hda_codec *codec)
-{
-	unsigned int idx;
-
-	if (snd_hda_jack_detect(codec, 0x14))
-		idx = 0;
-	else if (snd_hda_jack_detect(codec, 0x1c))
-		idx = 4;
-	else
-		idx = 1;
-	snd_hda_codec_write(codec, 0x0c, 0, AC_VERB_SET_CONNECT_SEL, idx);
-}
-
-/* unsolicited event for HP jack sensing */
-static void ad1884a_laptop_unsol_event(struct hda_codec *codec,
-				       unsigned int res)
-{
-	switch (res >> 26) {
-	case AD1884A_HP_EVENT:
-		ad1884a_laptop_automute(codec);
-		break;
-	case AD1884A_MIC_EVENT:
-		ad1884a_laptop_automic(codec);
-		break;
-	}
-}
-
-/* initialize jack-sensing, too */
-static int ad1884a_laptop_init(struct hda_codec *codec)
-{
-	ad198x_init(codec);
-	ad1884a_laptop_automute(codec);
-	ad1884a_laptop_automic(codec);
-	return 0;
-}
-
-/* additional verbs for laptop model */
-static const struct hda_verb ad1884a_laptop_verbs[] = {
-	/* Port-A (HP) pin - always unmuted */
-	{0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	/* Port-F (int speaker) mixer - route only from analog mixer */
-	{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-	/* Port-F (int speaker) pin */
-	{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
-	{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	/* required for compaq 6530s/6531s speaker output */
-	{0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
-	/* Port-C pin - internal mic-in */
-	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
-	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, 0x7002}, /* raise mic as default */
-	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0x7002}, /* raise mic as default */
-	/* Port-D (docking line-out) pin - default unmuted */
-	{0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	/* analog mix */
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
-	/* unsolicited event for pin-sense */
-	{0x11, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_HP_EVENT},
-	{0x12, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_HP_EVENT},
-	{0x14, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_MIC_EVENT},
-	{0x1c, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_MIC_EVENT},
-	/* allow to touch GPIO1 (for mute control) */
-	{0x01, AC_VERB_SET_GPIO_MASK, 0x02},
-	{0x01, AC_VERB_SET_GPIO_DIRECTION, 0x02},
-	{0x01, AC_VERB_SET_GPIO_DATA, 0x02}, /* first muted */
-	{ } /* end */
-};
-
-static const struct hda_verb ad1884a_mobile_verbs[] = {
-	/* DACs; unmute as default */
-	{0x03, AC_VERB_SET_AMP_GAIN_MUTE, 0x27}, /* 0dB */
-	{0x04, AC_VERB_SET_AMP_GAIN_MUTE, 0x27}, /* 0dB */
-	/* Port-A (HP) mixer - route only from analog mixer */
-	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-	/* Port-A pin */
-	{0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
-	/* Port-A (HP) pin - always unmuted */
-	{0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	/* Port-B (mic jack) pin */
-	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
-	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, 0x7002}, /* raise mic as default */
-	/* Port-C (int mic) pin */
-	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
-	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0x7002}, /* raise mic as default */
-	/* Port-F (int speaker) mixer - route only from analog mixer */
-	{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-	/* Port-F pin */
-	{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
-	{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	/* Analog mixer; mute as default */
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)},
-	/* Analog Mix output amp */
-	{0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	/* capture sources */
-	/* {0x0c, AC_VERB_SET_CONNECT_SEL, 0x0}, */ /* set via unsol */
-	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	{0x0d, AC_VERB_SET_CONNECT_SEL, 0x0},
-	{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	/* unsolicited event for pin-sense */
-	{0x11, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_HP_EVENT},
-	{0x14, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_MIC_EVENT},
-	/* allow to touch GPIO1 (for mute control) */
-	{0x01, AC_VERB_SET_GPIO_MASK, 0x02},
-	{0x01, AC_VERB_SET_GPIO_DIRECTION, 0x02},
-	{0x01, AC_VERB_SET_GPIO_DATA, 0x02}, /* first muted */
-	{ } /* end */
-};
-
-/*
- * Thinkpad X300
- * 0x11 - HP
- * 0x12 - speaker
- * 0x14 - mic-in
- * 0x17 - built-in mic
- */
-
-static const struct hda_verb ad1984a_thinkpad_verbs[] = {
-	/* HP unmute */
-	{0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	/* analog mix */
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
-	/* turn on EAPD */
-	{0x12, AC_VERB_SET_EAPD_BTLENABLE, 0x02},
-	/* unsolicited event for pin-sense */
-	{0x11, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_HP_EVENT},
-	/* internal mic - dmic */
-	{0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
-	/* set magic COEFs for dmic */
-	{0x01, AC_VERB_SET_COEF_INDEX, 0x13f7},
-	{0x01, AC_VERB_SET_PROC_COEF, 0x08},
-	{ } /* end */
-};
-
-static const struct snd_kcontrol_new ad1984a_thinkpad_mixers[] = {
-	HDA_CODEC_VOLUME("Master Playback Volume", 0x21, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Master Playback Switch", 0x21, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("PCM Playback Volume", 0x20, 0x5, HDA_INPUT),
-	HDA_CODEC_MUTE("PCM Playback Switch", 0x20, 0x5, HDA_INPUT),
-	HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x00, HDA_INPUT),
-	HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x00, HDA_INPUT),
-	HDA_CODEC_VOLUME("Mic Boost Volume", 0x14, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x17, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT),
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		.name = "Capture Source",
-		.info = ad198x_mux_enum_info,
-		.get = ad198x_mux_enum_get,
-		.put = ad198x_mux_enum_put,
-	},
-	{ } /* end */
-};
-
-static const struct hda_input_mux ad1984a_thinkpad_capture_source = {
-	.num_items = 3,
-	.items = {
-		{ "Mic", 0x0 },
-		{ "Internal Mic", 0x5 },
-		{ "Mix", 0x3 },
-	},
-};
-
-/* mute internal speaker if HP is plugged */
-static void ad1984a_thinkpad_automute(struct hda_codec *codec)
-{
-	unsigned int present;
-
-	present = snd_hda_jack_detect(codec, 0x11);
-	snd_hda_codec_amp_stereo(codec, 0x12, HDA_OUTPUT, 0,
-				 HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
-}
-
-/* unsolicited event for HP jack sensing */
-static void ad1984a_thinkpad_unsol_event(struct hda_codec *codec,
-					 unsigned int res)
-{
-	if ((res >> 26) != AD1884A_HP_EVENT)
-		return;
-	ad1984a_thinkpad_automute(codec);
-}
-
-/* initialize jack-sensing, too */
-static int ad1984a_thinkpad_init(struct hda_codec *codec)
-{
-	ad198x_init(codec);
-	ad1984a_thinkpad_automute(codec);
-	return 0;
-}
-
-/*
- * Precision R5500
- * 0x12 - HP/line-out
- * 0x13 - speaker (mono)
- * 0x15 - mic-in
- */
-
-static const struct hda_verb ad1984a_precision_verbs[] = {
-	/* Unmute main output path */
-	{0x03, AC_VERB_SET_AMP_GAIN_MUTE, 0x27}, /* 0dB */
-	{0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE + 0x1f}, /* 0dB */
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(5) + 0x17}, /* 0dB */
-	/* Analog mixer; mute as default */
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
-	/* Select mic as input */
-	{0x0c, AC_VERB_SET_CONNECT_SEL, 0x1},
-	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE + 0x27}, /* 0dB */
-	/* Configure as mic */
-	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
-	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0x7002}, /* raise mic as default */
-	/* HP unmute */
-	{0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	/* turn on EAPD */
-	{0x13, AC_VERB_SET_EAPD_BTLENABLE, 0x02},
-	/* unsolicited event for pin-sense */
-	{0x12, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_HP_EVENT},
-	{ } /* end */
-};
-
-static const struct snd_kcontrol_new ad1984a_precision_mixers[] = {
-	HDA_CODEC_VOLUME("Master Playback Volume", 0x21, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Master Playback Switch", 0x21, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("PCM Playback Volume", 0x20, 0x5, HDA_INPUT),
-	HDA_CODEC_MUTE("PCM Playback Switch", 0x20, 0x5, HDA_INPUT),
-	HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x01, HDA_INPUT),
-	HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x01, HDA_INPUT),
-	HDA_CODEC_VOLUME("Mic Boost Volume", 0x15, 0x0, HDA_INPUT),
-	HDA_CODEC_MUTE("Front Playback Switch", 0x12, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Speaker Playback Volume", 0x13, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT),
-	{ } /* end */
-};
-
-
-/* mute internal speaker if HP is plugged */
-static void ad1984a_precision_automute(struct hda_codec *codec)
-{
-	unsigned int present;
-
-	present = snd_hda_jack_detect(codec, 0x12);
-	snd_hda_codec_amp_stereo(codec, 0x13, HDA_OUTPUT, 0,
-				 HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
-}
-
-
-/* unsolicited event for HP jack sensing */
-static void ad1984a_precision_unsol_event(struct hda_codec *codec,
-					 unsigned int res)
-{
-	if ((res >> 26) != AD1884A_HP_EVENT)
-		return;
-	ad1984a_precision_automute(codec);
-}
-
-/* initialize jack-sensing, too */
-static int ad1984a_precision_init(struct hda_codec *codec)
-{
-	ad198x_init(codec);
-	ad1984a_precision_automute(codec);
-	return 0;
-}
-
-
-/*
- * HP Touchsmart
- * port-A (0x11)      - front hp-out
- * port-B (0x14)      - unused
- * port-C (0x15)      - unused
- * port-D (0x12)      - rear line out
- * port-E (0x1c)      - front mic-in
- * port-F (0x16)      - Internal speakers
- * digital-mic (0x17) - Internal mic
- */
-
-static const struct hda_verb ad1984a_touchsmart_verbs[] = {
-	/* DACs; unmute as default */
-	{0x03, AC_VERB_SET_AMP_GAIN_MUTE, 0x27}, /* 0dB */
-	{0x04, AC_VERB_SET_AMP_GAIN_MUTE, 0x27}, /* 0dB */
-	/* Port-A (HP) mixer - route only from analog mixer */
-	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-	/* Port-A pin */
-	{0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
-	/* Port-A (HP) pin - always unmuted */
-	{0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	/* Port-E (int speaker) mixer - route only from analog mixer */
-	{0x25, AC_VERB_SET_AMP_GAIN_MUTE, 0x03},
-	/* Port-E pin */
-	{0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
-	{0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	{0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
-	/* Port-F (int speaker) mixer - route only from analog mixer */
-	{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-	/* Port-F pin */
-	{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
-	{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	/* Analog mixer; mute as default */
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)},
-	/* Analog Mix output amp */
-	{0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	/* capture sources */
-	/* {0x0c, AC_VERB_SET_CONNECT_SEL, 0x0}, */ /* set via unsol */
-	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	{0x0d, AC_VERB_SET_CONNECT_SEL, 0x0},
-	{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	/* unsolicited event for pin-sense */
-	{0x11, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_HP_EVENT},
-	{0x1c, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_MIC_EVENT},
-	/* allow to touch GPIO1 (for mute control) */
-	{0x01, AC_VERB_SET_GPIO_MASK, 0x02},
-	{0x01, AC_VERB_SET_GPIO_DIRECTION, 0x02},
-	{0x01, AC_VERB_SET_GPIO_DATA, 0x02}, /* first muted */
-	/* internal mic - dmic */
-	{0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
-	/* set magic COEFs for dmic */
-	{0x01, AC_VERB_SET_COEF_INDEX, 0x13f7},
-	{0x01, AC_VERB_SET_PROC_COEF, 0x08},
-	{ } /* end */
-};
-
-static const struct snd_kcontrol_new ad1984a_touchsmart_mixers[] = {
-	HDA_CODEC_VOLUME("Master Playback Volume", 0x21, 0x0, HDA_OUTPUT),
-/*	HDA_CODEC_MUTE("Master Playback Switch", 0x21, 0x0, HDA_OUTPUT),*/
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		.subdevice = HDA_SUBDEV_AMP_FLAG,
-		.name = "Master Playback Switch",
-		.info = snd_hda_mixer_amp_switch_info,
-		.get = snd_hda_mixer_amp_switch_get,
-		.put = ad1884a_mobile_master_sw_put,
-		.private_value = HDA_COMPOSE_AMP_VAL(0x21, 3, 0, HDA_OUTPUT),
-	},
-	HDA_CODEC_VOLUME("PCM Playback Volume", 0x20, 0x5, HDA_INPUT),
-	HDA_CODEC_MUTE("PCM Playback Switch", 0x20, 0x5, HDA_INPUT),
-	HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Mic Boost Volume", 0x25, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x17, 0x0, HDA_INPUT),
-	{ } /* end */
-};
-
-/* switch to external mic if plugged */
-static void ad1984a_touchsmart_automic(struct hda_codec *codec)
-{
-	if (snd_hda_jack_detect(codec, 0x1c))
-		snd_hda_codec_write(codec, 0x0c, 0,
-				     AC_VERB_SET_CONNECT_SEL, 0x4);
-	else
-		snd_hda_codec_write(codec, 0x0c, 0,
-				     AC_VERB_SET_CONNECT_SEL, 0x5);
-}
-
-
-/* unsolicited event for HP jack sensing */
-static void ad1984a_touchsmart_unsol_event(struct hda_codec *codec,
-	unsigned int res)
-{
-	switch (res >> 26) {
-	case AD1884A_HP_EVENT:
-		ad1884a_hp_automute(codec);
-		break;
-	case AD1884A_MIC_EVENT:
-		ad1984a_touchsmart_automic(codec);
-		break;
-	}
-}
-
-/* initialize jack-sensing, too */
-static int ad1984a_touchsmart_init(struct hda_codec *codec)
-{
-	ad198x_init(codec);
-	ad1884a_hp_automute(codec);
-	ad1984a_touchsmart_automic(codec);
-	return 0;
-}
-
-
-/*
- */
-
-enum {
-	AD1884A_AUTO,
-	AD1884A_DESKTOP,
-	AD1884A_LAPTOP,
-	AD1884A_MOBILE,
-	AD1884A_THINKPAD,
-	AD1984A_TOUCHSMART,
-	AD1984A_PRECISION,
-	AD1884A_MODELS
-};
-
-static const char * const ad1884a_models[AD1884A_MODELS] = {
-	[AD1884A_AUTO]		= "auto",
-	[AD1884A_DESKTOP]	= "desktop",
-	[AD1884A_LAPTOP]	= "laptop",
-	[AD1884A_MOBILE]	= "mobile",
-	[AD1884A_THINKPAD]	= "thinkpad",
-	[AD1984A_TOUCHSMART]	= "touchsmart",
-	[AD1984A_PRECISION]	= "precision",
-};
-
-static const struct snd_pci_quirk ad1884a_cfg_tbl[] = {
-	SND_PCI_QUIRK(0x1028, 0x04ac, "Precision R5500", AD1984A_PRECISION),
-	SND_PCI_QUIRK(0x103c, 0x3030, "HP", AD1884A_MOBILE),
-	SND_PCI_QUIRK(0x103c, 0x3037, "HP 2230s", AD1884A_LAPTOP),
-	SND_PCI_QUIRK(0x103c, 0x3056, "HP", AD1884A_MOBILE),
-	SND_PCI_QUIRK_MASK(0x103c, 0xfff0, 0x3070, "HP", AD1884A_MOBILE),
-	SND_PCI_QUIRK_MASK(0x103c, 0xfff0, 0x30d0, "HP laptop", AD1884A_LAPTOP),
-	SND_PCI_QUIRK_MASK(0x103c, 0xfff0, 0x30e0, "HP laptop", AD1884A_LAPTOP),
-	SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x3600, "HP laptop", AD1884A_LAPTOP),
-	SND_PCI_QUIRK_MASK(0x103c, 0xfff0, 0x7010, "HP laptop", AD1884A_MOBILE),
-	SND_PCI_QUIRK(0x17aa, 0x20ac, "Thinkpad X300", AD1884A_THINKPAD),
-	SND_PCI_QUIRK(0x103c, 0x2a82, "Touchsmart", AD1984A_TOUCHSMART),
-	{}
-};
-
-static int patch_ad1884a(struct hda_codec *codec)
-{
-	struct ad198x_spec *spec;
-	int err, board_config;
-
-	board_config = snd_hda_check_board_config(codec, AD1884A_MODELS,
-						  ad1884a_models,
-						  ad1884a_cfg_tbl);
-	if (board_config < 0) {
-		printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n",
-		       codec->chip_name);
-		board_config = AD1884A_AUTO;
-	}
-
-	if (board_config == AD1884A_AUTO)
-		return ad1884_parse_auto_config(codec);
-
-	err = alloc_ad_spec(codec);
-	if (err < 0)
-		return err;
-	spec = codec->spec;
-
-	err = snd_hda_attach_beep_device(codec, 0x10);
-	if (err < 0) {
-		ad198x_free(codec);
-		return err;
-	}
-	set_beep_amp(spec, 0x10, 0, HDA_OUTPUT);
-
-	spec->multiout.max_channels = 2;
-	spec->multiout.num_dacs = ARRAY_SIZE(ad1884a_dac_nids);
-	spec->multiout.dac_nids = ad1884a_dac_nids;
-	spec->multiout.dig_out_nid = AD1884A_SPDIF_OUT;
-	spec->num_adc_nids = ARRAY_SIZE(ad1884a_adc_nids);
-	spec->adc_nids = ad1884a_adc_nids;
-	spec->capsrc_nids = ad1884a_capsrc_nids;
-	spec->input_mux = &ad1884a_capture_source;
-	spec->num_mixers = 1;
-	spec->mixers[0] = ad1884a_base_mixers;
-	spec->num_init_verbs = 1;
-	spec->init_verbs[0] = ad1884a_init_verbs;
-	spec->spdif_route = 0;
-#ifdef CONFIG_PM
-	spec->loopback.amplist = ad1884a_loopbacks;
-#endif
-	codec->patch_ops = ad198x_patch_ops;
-
-	/* override some parameters */
-	switch (board_config) {
-	case AD1884A_LAPTOP:
-		spec->mixers[0] = ad1884a_laptop_mixers;
-		spec->init_verbs[spec->num_init_verbs++] = ad1884a_laptop_verbs;
-		spec->multiout.dig_out_nid = 0;
-		codec->patch_ops.unsol_event = ad1884a_laptop_unsol_event;
-		codec->patch_ops.init = ad1884a_laptop_init;
-		/* set the upper-limit for mixer amp to 0dB for avoiding the
-		 * possible damage by overloading
-		 */
-		snd_hda_override_amp_caps(codec, 0x20, HDA_INPUT,
-					  (0x17 << AC_AMPCAP_OFFSET_SHIFT) |
-					  (0x17 << AC_AMPCAP_NUM_STEPS_SHIFT) |
-					  (0x05 << AC_AMPCAP_STEP_SIZE_SHIFT) |
-					  (1 << AC_AMPCAP_MUTE_SHIFT));
-		break;
-	case AD1884A_MOBILE:
-		spec->mixers[0] = ad1884a_mobile_mixers;
-		spec->init_verbs[0] = ad1884a_mobile_verbs;
-		spec->multiout.dig_out_nid = 0;
-		codec->patch_ops.unsol_event = ad1884a_hp_unsol_event;
-		codec->patch_ops.init = ad1884a_hp_init;
-		/* set the upper-limit for mixer amp to 0dB for avoiding the
-		 * possible damage by overloading
-		 */
-		snd_hda_override_amp_caps(codec, 0x20, HDA_INPUT,
-					  (0x17 << AC_AMPCAP_OFFSET_SHIFT) |
-					  (0x17 << AC_AMPCAP_NUM_STEPS_SHIFT) |
-					  (0x05 << AC_AMPCAP_STEP_SIZE_SHIFT) |
-					  (1 << AC_AMPCAP_MUTE_SHIFT));
-		break;
-	case AD1884A_THINKPAD:
-		spec->mixers[0] = ad1984a_thinkpad_mixers;
-		spec->init_verbs[spec->num_init_verbs++] =
-			ad1984a_thinkpad_verbs;
-		spec->multiout.dig_out_nid = 0;
-		spec->input_mux = &ad1984a_thinkpad_capture_source;
-		codec->patch_ops.unsol_event = ad1984a_thinkpad_unsol_event;
-		codec->patch_ops.init = ad1984a_thinkpad_init;
-		break;
-	case AD1984A_PRECISION:
-		spec->mixers[0] = ad1984a_precision_mixers;
-		spec->init_verbs[spec->num_init_verbs++] =
-			ad1984a_precision_verbs;
-		spec->multiout.dig_out_nid = 0;
-		codec->patch_ops.unsol_event = ad1984a_precision_unsol_event;
-		codec->patch_ops.init = ad1984a_precision_init;
-		break;
-	case AD1984A_TOUCHSMART:
-		spec->mixers[0] = ad1984a_touchsmart_mixers;
-		spec->init_verbs[0] = ad1984a_touchsmart_verbs;
-		spec->multiout.dig_out_nid = 0;
-		codec->patch_ops.unsol_event = ad1984a_touchsmart_unsol_event;
-		codec->patch_ops.init = ad1984a_touchsmart_init;
-		/* set the upper-limit for mixer amp to 0dB for avoiding the
-		 * possible damage by overloading
-		 */
-		snd_hda_override_amp_caps(codec, 0x20, HDA_INPUT,
-					  (0x17 << AC_AMPCAP_OFFSET_SHIFT) |
-					  (0x17 << AC_AMPCAP_NUM_STEPS_SHIFT) |
-					  (0x05 << AC_AMPCAP_STEP_SIZE_SHIFT) |
-					  (1 << AC_AMPCAP_MUTE_SHIFT));
-		break;
-	}
-
-	codec->no_trigger_sense = 1;
-	codec->no_sticky_stream = 1;
-
-	return 0;
-}
-#else /* ENABLE_AD_STATIC_QUIRKS */
-#define patch_ad1884a	ad1884_parse_auto_config
-#endif /* ENABLE_AD_STATIC_QUIRKS */
-
-
 /*
  * AD1882 / AD1882A
  *
@@ -4850,299 +1058,7 @@ static int patch_ad1884a(struct hda_codec *codec)
  * port-G - rear clfe-out (6stack)
  */
 
-#ifdef ENABLE_AD_STATIC_QUIRKS
-static const hda_nid_t ad1882_dac_nids[3] = {
-	0x04, 0x03, 0x05
-};
-
-static const hda_nid_t ad1882_adc_nids[2] = {
-	0x08, 0x09,
-};
-
-static const hda_nid_t ad1882_capsrc_nids[2] = {
-	0x0c, 0x0d,
-};
-
-#define AD1882_SPDIF_OUT	0x02
-
-/* list: 0x11, 0x39, 0x3a, 0x18, 0x3c, 0x3b, 0x12, 0x20 */
-static const struct hda_input_mux ad1882_capture_source = {
-	.num_items = 5,
-	.items = {
-		{ "Front Mic", 0x1 },
-		{ "Mic", 0x4 },
-		{ "Line", 0x2 },
-		{ "CD", 0x3 },
-		{ "Mix", 0x7 },
-	},
-};
-
-/* list: 0x11, 0x39, 0x3a, 0x3c, 0x18, 0x1f, 0x12, 0x20 */
-static const struct hda_input_mux ad1882a_capture_source = {
-	.num_items = 5,
-	.items = {
-		{ "Front Mic", 0x1 },
-		{ "Mic", 0x4},
-		{ "Line", 0x2 },
-		{ "Digital Mic", 0x06 },
-		{ "Mix", 0x7 },
-	},
-};
-
-static const struct snd_kcontrol_new ad1882_base_mixers[] = {
-	HDA_CODEC_VOLUME("Front Playback Volume", 0x04, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Surround Playback Volume", 0x03, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x05, 1, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x05, 2, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Headphone Playback Switch", 0x11, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Front Playback Switch", 0x12, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x13, 1, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE_MONO("Mono Playback Switch", 0x13, 1, 0x0, HDA_OUTPUT),
-
-	HDA_CODEC_VOLUME("Mic Boost Volume", 0x3c, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x39, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Line-In Boost Volume", 0x3a, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x0d, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x0d, 0x0, HDA_OUTPUT),
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		/* The multiple "Capture Source" controls confuse alsamixer
-		 * So call somewhat different..
-		 */
-		/* .name = "Capture Source", */
-		.name = "Input Source",
-		.count = 2,
-		.info = ad198x_mux_enum_info,
-		.get = ad198x_mux_enum_get,
-		.put = ad198x_mux_enum_put,
-	},
-	/* SPDIF controls */
-	HDA_CODEC_VOLUME("IEC958 Playback Volume", 0x1b, 0x0, HDA_OUTPUT),
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		.name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,NONE) "Source",
-		/* identical with ad1983 */
-		.info = ad1983_spdif_route_info,
-		.get = ad1983_spdif_route_get,
-		.put = ad1983_spdif_route_put,
-	},
-	{ } /* end */
-};
-
-static const struct snd_kcontrol_new ad1882_loopback_mixers[] = {
-	HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x20, 0x00, HDA_INPUT),
-	HDA_CODEC_MUTE("Front Mic Playback Switch", 0x20, 0x00, HDA_INPUT),
-	HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x01, HDA_INPUT),
-	HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x01, HDA_INPUT),
-	HDA_CODEC_VOLUME("Line Playback Volume", 0x20, 0x04, HDA_INPUT),
-	HDA_CODEC_MUTE("Line Playback Switch", 0x20, 0x04, HDA_INPUT),
-	HDA_CODEC_VOLUME("CD Playback Volume", 0x20, 0x06, HDA_INPUT),
-	HDA_CODEC_MUTE("CD Playback Switch", 0x20, 0x06, HDA_INPUT),
-	{ } /* end */
-};
-
-static const struct snd_kcontrol_new ad1882a_loopback_mixers[] = {
-	HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x20, 0x00, HDA_INPUT),
-	HDA_CODEC_MUTE("Front Mic Playback Switch", 0x20, 0x00, HDA_INPUT),
-	HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x04, HDA_INPUT),
-	HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x04, HDA_INPUT),
-	HDA_CODEC_VOLUME("Line Playback Volume", 0x20, 0x01, HDA_INPUT),
-	HDA_CODEC_MUTE("Line Playback Switch", 0x20, 0x01, HDA_INPUT),
-	HDA_CODEC_VOLUME("CD Playback Volume", 0x20, 0x06, HDA_INPUT),
-	HDA_CODEC_MUTE("CD Playback Switch", 0x20, 0x06, HDA_INPUT),
-	HDA_CODEC_VOLUME("Digital Mic Boost Volume", 0x1f, 0x0, HDA_INPUT),
-	{ } /* end */
-};
-
-static const struct snd_kcontrol_new ad1882_3stack_mixers[] = {
-	HDA_CODEC_MUTE("Surround Playback Switch", 0x15, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x17, 1, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x17, 2, 0x0, HDA_OUTPUT),
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		.name = "Channel Mode",
-		.info = ad198x_ch_mode_info,
-		.get = ad198x_ch_mode_get,
-		.put = ad198x_ch_mode_put,
-	},
-	{ } /* end */
-};
-
-/* simple auto-mute control for AD1882 3-stack board */
-#define AD1882_HP_EVENT	0x01
-
-static void ad1882_3stack_automute(struct hda_codec *codec)
-{
-	bool mute = snd_hda_jack_detect(codec, 0x11);
-	snd_hda_codec_write(codec, 0x12, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
-			    mute ? 0 : PIN_OUT);
-}
-
-static int ad1882_3stack_automute_init(struct hda_codec *codec)
-{
-	ad198x_init(codec);
-	ad1882_3stack_automute(codec);
-	return 0;
-}
-
-static void ad1882_3stack_unsol_event(struct hda_codec *codec, unsigned int res)
-{
-	switch (res >> 26) {
-	case AD1882_HP_EVENT:
-		ad1882_3stack_automute(codec);
-		break;
-	}
-}
-
-static const struct snd_kcontrol_new ad1882_6stack_mixers[] = {
-	HDA_CODEC_MUTE("Surround Playback Switch", 0x16, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x24, 1, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x24, 2, 0x0, HDA_OUTPUT),
-	{ } /* end */
-};
-
-static const struct hda_verb ad1882_ch2_init[] = {
-	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
-	{0x2c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x2c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
-	{0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
-	{0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
-	{ } /* end */
-};
-
-static const struct hda_verb ad1882_ch4_init[] = {
-	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
-	{0x2c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	{0x2c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-	{0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
-	{0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
-	{ } /* end */
-};
-
-static const struct hda_verb ad1882_ch6_init[] = {
-	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
-	{0x2c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	{0x2c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-	{0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
-	{0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	{0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-	{ } /* end */
-};
-
-static const struct hda_channel_mode ad1882_modes[3] = {
-	{ 2, ad1882_ch2_init },
-	{ 4, ad1882_ch4_init },
-	{ 6, ad1882_ch6_init },
-};
-
-/*
- * initialization verbs
- */
-static const struct hda_verb ad1882_init_verbs[] = {
-	/* DACs; mute as default */
-	{0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
-	{0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
-	{0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
-	/* Port-A (HP) mixer */
-	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-	/* Port-A pin */
-	{0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
-	{0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	/* HP selector - select DAC2 */
-	{0x37, AC_VERB_SET_CONNECT_SEL, 0x1},
-	/* Port-D (Line-out) mixer */
-	{0x29, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	{0x29, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-	/* Port-D pin */
-	{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
-	{0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	/* Mono-out mixer */
-	{0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	{0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-	/* Mono-out pin */
-	{0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
-	{0x13, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	/* Port-B (front mic) pin */
-	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
-	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	{0x39, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, /* boost */
-	/* Port-C (line-in) pin */
-	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
-	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	{0x3a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, /* boost */
-	/* Port-C mixer - mute as input */
-	{0x2c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x2c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
-	/* Port-E (mic-in) pin */
-	{0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
-	{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	{0x3c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, /* boost */
-	/* Port-E mixer - mute as input */
-	{0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
-	/* Port-F (surround) */
-	{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
-	{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	/* Port-G (CLFE) */
-	{0x24, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
-	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	/* Analog mixer; mute as default */
-	/* list: 0x39, 0x3a, 0x11, 0x12, 0x3c, 0x3b, 0x18, 0x1a */
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)},
-	/* Analog Mix output amp */
-	{0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x1f}, /* 0dB */
-	/* SPDIF output selector */
-	{0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x27}, /* 0dB */
-	{0x02, AC_VERB_SET_CONNECT_SEL, 0x0}, /* PCM */
-	{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x27}, /* 0dB */
-	{ } /* end */
-};
-
-static const struct hda_verb ad1882_3stack_automute_verbs[] = {
-	{0x11, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1882_HP_EVENT},
-	{ } /* end */
-};
-
-#ifdef CONFIG_PM
-static const struct hda_amp_list ad1882_loopbacks[] = {
-	{ 0x20, HDA_INPUT, 0 }, /* Front Mic */
-	{ 0x20, HDA_INPUT, 1 }, /* Mic */
-	{ 0x20, HDA_INPUT, 4 }, /* Line */
-	{ 0x20, HDA_INPUT, 6 }, /* CD */
-	{ } /* end */
-};
-#endif
-
-/* models */
-enum {
-	AD1882_AUTO,
-	AD1882_3STACK,
-	AD1882_6STACK,
-	AD1882_3STACK_AUTOMUTE,
-	AD1882_MODELS
-};
-
-static const char * const ad1882_models[AD1986A_MODELS] = {
-	[AD1882_AUTO]		= "auto",
-	[AD1882_3STACK]		= "3stack",
-	[AD1882_6STACK]		= "6stack",
-	[AD1882_3STACK_AUTOMUTE] = "3stack-automute",
-};
-#endif /* ENABLE_AD_STATIC_QUIRKS */
-
-static int ad1882_parse_auto_config(struct hda_codec *codec)
+static int patch_ad1882(struct hda_codec *codec)
 {
 	struct ad198x_spec *spec;
 	int err;
@@ -5169,110 +1085,20 @@ static int ad1882_parse_auto_config(struct hda_codec *codec)
 	return err;
 }
 
-#ifdef ENABLE_AD_STATIC_QUIRKS
-static int patch_ad1882(struct hda_codec *codec)
-{
-	struct ad198x_spec *spec;
-	int err, board_config;
-
-	board_config = snd_hda_check_board_config(codec, AD1882_MODELS,
-						  ad1882_models, NULL);
-	if (board_config < 0) {
-		printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n",
-		       codec->chip_name);
-		board_config = AD1882_AUTO;
-	}
-
-	if (board_config == AD1882_AUTO)
-		return ad1882_parse_auto_config(codec);
-
-	err = alloc_ad_spec(codec);
-	if (err < 0)
-		return err;
-	spec = codec->spec;
-
-	err = snd_hda_attach_beep_device(codec, 0x10);
-	if (err < 0) {
-		ad198x_free(codec);
-		return err;
-	}
-	set_beep_amp(spec, 0x10, 0, HDA_OUTPUT);
-
-	spec->multiout.max_channels = 6;
-	spec->multiout.num_dacs = 3;
-	spec->multiout.dac_nids = ad1882_dac_nids;
-	spec->multiout.dig_out_nid = AD1882_SPDIF_OUT;
-	spec->num_adc_nids = ARRAY_SIZE(ad1882_adc_nids);
-	spec->adc_nids = ad1882_adc_nids;
-	spec->capsrc_nids = ad1882_capsrc_nids;
-	if (codec->vendor_id == 0x11d41882)
-		spec->input_mux = &ad1882_capture_source;
-	else
-		spec->input_mux = &ad1882a_capture_source;
-	spec->num_mixers = 2;
-	spec->mixers[0] = ad1882_base_mixers;
-	if (codec->vendor_id == 0x11d41882)
-		spec->mixers[1] = ad1882_loopback_mixers;
-	else
-		spec->mixers[1] = ad1882a_loopback_mixers;
-	spec->num_init_verbs = 1;
-	spec->init_verbs[0] = ad1882_init_verbs;
-	spec->spdif_route = 0;
-#ifdef CONFIG_PM
-	spec->loopback.amplist = ad1882_loopbacks;
-#endif
-	spec->vmaster_nid = 0x04;
-
-	codec->patch_ops = ad198x_patch_ops;
-
-	/* override some parameters */
-	switch (board_config) {
-	default:
-	case AD1882_3STACK:
-	case AD1882_3STACK_AUTOMUTE:
-		spec->num_mixers = 3;
-		spec->mixers[2] = ad1882_3stack_mixers;
-		spec->channel_mode = ad1882_modes;
-		spec->num_channel_mode = ARRAY_SIZE(ad1882_modes);
-		spec->need_dac_fix = 1;
-		spec->multiout.max_channels = 2;
-		spec->multiout.num_dacs = 1;
-		if (board_config != AD1882_3STACK) {
-			spec->init_verbs[spec->num_init_verbs++] =
-				ad1882_3stack_automute_verbs;
-			codec->patch_ops.unsol_event = ad1882_3stack_unsol_event;
-			codec->patch_ops.init = ad1882_3stack_automute_init;
-		}
-		break;
-	case AD1882_6STACK:
-		spec->num_mixers = 3;
-		spec->mixers[2] = ad1882_6stack_mixers;
-		break;
-	}
-
-	codec->no_trigger_sense = 1;
-	codec->no_sticky_stream = 1;
-
-	return 0;
-}
-#else /* ENABLE_AD_STATIC_QUIRKS */
-#define patch_ad1882	ad1882_parse_auto_config
-#endif /* ENABLE_AD_STATIC_QUIRKS */
-
 
 /*
  * patch entries
  */
 static const struct hda_codec_preset snd_hda_preset_analog[] = {
-	{ .id = 0x11d4184a, .name = "AD1884A", .patch = patch_ad1884a },
+	{ .id = 0x11d4184a, .name = "AD1884A", .patch = patch_ad1884 },
 	{ .id = 0x11d41882, .name = "AD1882", .patch = patch_ad1882 },
-	{ .id = 0x11d41883, .name = "AD1883", .patch = patch_ad1884a },
+	{ .id = 0x11d41883, .name = "AD1883", .patch = patch_ad1884 },
 	{ .id = 0x11d41884, .name = "AD1884", .patch = patch_ad1884 },
-	{ .id = 0x11d4194a, .name = "AD1984A", .patch = patch_ad1884a },
-	{ .id = 0x11d4194b, .name = "AD1984B", .patch = patch_ad1884a },
+	{ .id = 0x11d4194a, .name = "AD1984A", .patch = patch_ad1884 },
+	{ .id = 0x11d4194b, .name = "AD1984B", .patch = patch_ad1884 },
 	{ .id = 0x11d41981, .name = "AD1981", .patch = patch_ad1981 },
 	{ .id = 0x11d41983, .name = "AD1983", .patch = patch_ad1983 },
-	{ .id = 0x11d41984, .name = "AD1984", .patch = patch_ad1984 },
+	{ .id = 0x11d41984, .name = "AD1984", .patch = patch_ad1884 },
 	{ .id = 0x11d41986, .name = "AD1986A", .patch = patch_ad1986a },
 	{ .id = 0x11d41988, .name = "AD1988", .patch = patch_ad1988 },
 	{ .id = 0x11d4198b, .name = "AD1988B", .patch = patch_ad1988 },

sound/pci/hda/patch_conexant.c

@@ -66,6 +66,8 @@ struct conexant_spec {
 	hda_nid_t eapds[4];
 	bool dynamic_eapd;
 
+	unsigned int parse_flags; /* flag for snd_hda_parse_pin_defcfg() */
+
 #ifdef ENABLE_CXT_STATIC_QUIRKS
 	const struct snd_kcontrol_new *mixers[5];
 	int num_mixers;
@@ -3200,6 +3202,9 @@ static int cx_auto_init(struct hda_codec *codec)
 	snd_hda_gen_init(codec);
 	if (!spec->dynamic_eapd)
 		cx_auto_turn_eapd(codec, spec->num_eapds, spec->eapds, true);
+
+	snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_INIT);
+
 	return 0;
 }
 
@@ -3224,6 +3229,8 @@ enum {
 	CXT_PINCFG_LEMOTE_A1205,
 	CXT_FIXUP_STEREO_DMIC,
 	CXT_FIXUP_INC_MIC_BOOST,
+	CXT_FIXUP_HEADPHONE_MIC_PIN,
+	CXT_FIXUP_HEADPHONE_MIC,
 };
 
 static void cxt_fixup_stereo_dmic(struct hda_codec *codec,
@@ -3246,6 +3253,59 @@ static void cxt5066_increase_mic_boost(struct hda_codec *codec,
 				  (0 << AC_AMPCAP_MUTE_SHIFT));
 }
 
+static void cxt_update_headset_mode(struct hda_codec *codec)
+{
+	/* The verbs used in this function were tested on a Conexant CX20751/2 codec. */
+	int i;
+	bool mic_mode = false;
+	struct conexant_spec *spec = codec->spec;
+	struct auto_pin_cfg *cfg = &spec->gen.autocfg;
+
+	hda_nid_t mux_pin = spec->gen.imux_pins[spec->gen.cur_mux[0]];
+
+	for (i = 0; i < cfg->num_inputs; i++)
+		if (cfg->inputs[i].pin == mux_pin) {
+			mic_mode = !!cfg->inputs[i].is_headphone_mic;
+			break;
+		}
+
+	if (mic_mode) {
+		snd_hda_codec_write_cache(codec, 0x1c, 0, 0x410, 0x7c); /* enable merged mode for analog int-mic */
+		spec->gen.hp_jack_present = false;
+	} else {
+		snd_hda_codec_write_cache(codec, 0x1c, 0, 0x410, 0x54); /* disable merged mode for analog int-mic */
+		spec->gen.hp_jack_present = snd_hda_jack_detect(codec, spec->gen.autocfg.hp_pins[0]);
+	}
+
+	snd_hda_gen_update_outputs(codec);
+}
+
+static void cxt_update_headset_mode_hook(struct hda_codec *codec,
+			     struct snd_ctl_elem_value *ucontrol)
+{
+	cxt_update_headset_mode(codec);
+}
+
+static void cxt_fixup_headphone_mic(struct hda_codec *codec,
+				    const struct hda_fixup *fix, int action)
+{
+	struct conexant_spec *spec = codec->spec;
+
+	switch (action) {
+	case HDA_FIXUP_ACT_PRE_PROBE:
+		spec->parse_flags |= HDA_PINCFG_HEADPHONE_MIC;
+		break;
+	case HDA_FIXUP_ACT_PROBE:
+		spec->gen.cap_sync_hook = cxt_update_headset_mode_hook;
+		spec->gen.automute_hook = cxt_update_headset_mode;
+		break;
+	case HDA_FIXUP_ACT_INIT:
+		cxt_update_headset_mode(codec);
+		break;
+	}
+}
+
+
 /* ThinkPad X200 & co with cxt5051 */
 static const struct hda_pintbl cxt_pincfg_lenovo_x200[] = {
 	{ 0x16, 0x042140ff }, /* HP (seq# overridden) */
@@ -3302,6 +3362,19 @@ static const struct hda_fixup cxt_fixups[] = {
 		.type = HDA_FIXUP_FUNC,
 		.v.func = cxt5066_increase_mic_boost,
 	},
+	[CXT_FIXUP_HEADPHONE_MIC_PIN] = {
+		.type = HDA_FIXUP_PINS,
+		.chained = true,
+		.chain_id = CXT_FIXUP_HEADPHONE_MIC,
+		.v.pins = (const struct hda_pintbl[]) {
+			{ 0x18, 0x03a1913d }, /* use as headphone mic, without its own jack detect */
+			{ }
+		}
+	},
+	[CXT_FIXUP_HEADPHONE_MIC] = {
+		.type = HDA_FIXUP_FUNC,
+		.v.func = cxt_fixup_headphone_mic,
+	},
 };
 
 static const struct snd_pci_quirk cxt5051_fixups[] = {
@@ -3311,6 +3384,7 @@ static const struct snd_pci_quirk cxt5051_fixups[] = {
 
 static const struct snd_pci_quirk cxt5066_fixups[] = {
 	SND_PCI_QUIRK(0x1025, 0x0543, "Acer Aspire One 522", CXT_FIXUP_STEREO_DMIC),
+	SND_PCI_QUIRK(0x1043, 0x138d, "Asus", CXT_FIXUP_HEADPHONE_MIC_PIN),
 	SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo T400", CXT_PINCFG_LENOVO_TP410),
 	SND_PCI_QUIRK(0x17aa, 0x215e, "Lenovo T410", CXT_PINCFG_LENOVO_TP410),
 	SND_PCI_QUIRK(0x17aa, 0x215f, "Lenovo T510", CXT_PINCFG_LENOVO_TP410),
@@ -3395,7 +3469,8 @@ static int patch_conexant_auto(struct hda_codec *codec)
 
 	snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE);
 
-	err = snd_hda_parse_pin_defcfg(codec, &spec->gen.autocfg, NULL, 0);
+	err = snd_hda_parse_pin_defcfg(codec, &spec->gen.autocfg, NULL,
+				       spec->parse_flags);
 	if (err < 0)
 		goto error;
 
@@ -3416,6 +3491,8 @@ static int patch_conexant_auto(struct hda_codec *codec)
 		codec->bus->allow_bus_reset = 1;
 	}
 
+	snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PROBE);
+
 	return 0;
 
  error:

sound/pci/hda/patch_hdmi.c

@@ -67,6 +67,8 @@ struct hdmi_spec_per_pin {
 	struct delayed_work work;
 	struct snd_kcontrol *eld_ctl;
 	int repoll_count;
+	bool setup; /* the stream has been set up by prepare callback */
+	int channels; /* current number of channels */
 	bool non_pcm;
 	bool chmap_set;		/* channel-map override by ALSA API? */
 	unsigned char chmap[8]; /* ALSA API channel-map */
@@ -551,6 +553,17 @@ static int hdmi_channel_allocation(struct hdmi_eld *eld, int channels)
 		}
 	}
 
+	if (!ca) {
+		/* if there was no match, select the regular ALSA channel
+		 * allocation with the matching number of channels */
+		for (i = 0; i < ARRAY_SIZE(channel_allocations); i++) {
+			if (channels == channel_allocations[i].channels) {
+				ca = channel_allocations[i].ca_index;
+				break;
+			}
+		}
+	}
+
 	snd_print_channel_allocation(eld->info.spk_alloc, buf, sizeof(buf));
 	snd_printdd("HDMI: select CA 0x%x for %d-channel allocation: %s\n",
 		    ca, channels, buf);
@@ -868,18 +881,19 @@ static bool hdmi_infoframe_uptodate(struct hda_codec *codec, hda_nid_t pin_nid,
 	return true;
 }
 
-static void hdmi_setup_audio_infoframe(struct hda_codec *codec, int pin_idx,
-				       bool non_pcm,
-				       struct snd_pcm_substream *substream)
+static void hdmi_setup_audio_infoframe(struct hda_codec *codec,
+				       struct hdmi_spec_per_pin *per_pin,
+				       bool non_pcm)
 {
-	struct hdmi_spec *spec = codec->spec;
-	struct hdmi_spec_per_pin *per_pin = get_pin(spec, pin_idx);
 	hda_nid_t pin_nid = per_pin->pin_nid;
-	int channels = substream->runtime->channels;
+	int channels = per_pin->channels;
 	struct hdmi_eld *eld;
 	int ca;
 	union audio_infoframe ai;
 
+	if (!channels)
+		return;
+
 	eld = &per_pin->sink_eld;
 	if (!eld->monitor_present)
 		return;
@@ -959,6 +973,7 @@ static void hdmi_intrinsic_event(struct hda_codec *codec, unsigned int res)
 	int pin_nid;
 	int pin_idx;
 	struct hda_jack_tbl *jack;
+	int dev_entry = (res & AC_UNSOL_RES_DE) >> AC_UNSOL_RES_DE_SHIFT;
 
 	jack = snd_hda_jack_tbl_get_from_tag(codec, tag);
 	if (!jack)
@@ -967,8 +982,8 @@ static void hdmi_intrinsic_event(struct hda_codec *codec, unsigned int res)
 	jack->jack_dirty = 1;
 
 	_snd_printd(SND_PR_VERBOSE,
-		"HDMI hot plug event: Codec=%d Pin=%d Presence_Detect=%d ELD_Valid=%d\n",
-		codec->addr, pin_nid,
+		"HDMI hot plug event: Codec=%d Pin=%d Device=%d Inactive=%d Presence_Detect=%d ELD_Valid=%d\n",
+		codec->addr, pin_nid, dev_entry, !!(res & AC_UNSOL_RES_IA),
 		!!(res & AC_UNSOL_RES_PD), !!(res & AC_UNSOL_RES_ELDV));
 
 	pin_idx = pin_nid_to_pin_index(spec, pin_nid);
@@ -1329,6 +1344,7 @@ static void hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll)
 		eld_changed = true;
 	}
 	if (update_eld) {
+		bool old_eld_valid = pin_eld->eld_valid;
 		pin_eld->eld_valid = eld->eld_valid;
 		eld_changed = pin_eld->eld_size != eld->eld_size ||
 			      memcmp(pin_eld->eld_buffer, eld->eld_buffer,
@@ -1338,6 +1354,18 @@ static void hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll)
 			       eld->eld_size);
 		pin_eld->eld_size = eld->eld_size;
 		pin_eld->info = eld->info;
+
+		/* Haswell-specific workaround: re-setup when the transcoder is
+		 * changed during the stream playback
+		 */
+		if (codec->vendor_id == 0x80862807 &&
+		    eld->eld_valid && !old_eld_valid && per_pin->setup) {
+			snd_hda_codec_write(codec, pin_nid, 0,
+					    AC_VERB_SET_AMP_GAIN_MUTE,
+					    AMP_OUT_UNMUTE);
+			hdmi_setup_audio_infoframe(codec, per_pin,
+						   per_pin->non_pcm);
+		}
 	}
 	mutex_unlock(&pin_eld->lock);
 
@@ -1510,14 +1538,17 @@ static int generic_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
 	hda_nid_t cvt_nid = hinfo->nid;
 	struct hdmi_spec *spec = codec->spec;
 	int pin_idx = hinfo_to_pin_index(spec, hinfo);
-	hda_nid_t pin_nid = get_pin(spec, pin_idx)->pin_nid;
+	struct hdmi_spec_per_pin *per_pin = get_pin(spec, pin_idx);
+	hda_nid_t pin_nid = per_pin->pin_nid;
 	bool non_pcm;
 
 	non_pcm = check_non_pcm_per_cvt(codec, cvt_nid);
+	per_pin->channels = substream->runtime->channels;
+	per_pin->setup = true;
 
 	hdmi_set_channel_count(codec, cvt_nid, substream->runtime->channels);
 
-	hdmi_setup_audio_infoframe(codec, pin_idx, non_pcm, substream);
+	hdmi_setup_audio_infoframe(codec, per_pin, non_pcm);
 
 	return hdmi_setup_stream(codec, cvt_nid, pin_nid, stream_tag, format);
 }
@@ -1557,6 +1588,9 @@ static int hdmi_pcm_close(struct hda_pcm_stream *hinfo,
 		snd_hda_spdif_ctls_unassign(codec, pin_idx);
 		per_pin->chmap_set = false;
 		memset(per_pin->chmap, 0, sizeof(per_pin->chmap));
+
+		per_pin->setup = false;
+		per_pin->channels = 0;
 	}
 
 	return 0;
@@ -1692,8 +1726,7 @@ static int hdmi_chmap_ctl_put(struct snd_kcontrol *kcontrol,
 	per_pin->chmap_set = true;
 	memcpy(per_pin->chmap, chmap, sizeof(chmap));
 	if (prepared)
-		hdmi_setup_audio_infoframe(codec, pin_idx, per_pin->non_pcm,
-					   substream);
+		hdmi_setup_audio_infoframe(codec, per_pin, per_pin->non_pcm);
 
 	return 0;
 }
@@ -1992,8 +2025,10 @@ static int patch_generic_hdmi(struct hda_codec *codec)
 		return -EINVAL;
 	}
 	codec->patch_ops = generic_hdmi_patch_ops;
-	if (codec->vendor_id == 0x80862807)
+	if (codec->vendor_id == 0x80862807) {
 		codec->patch_ops.set_power_state = haswell_set_power_state;
+		codec->dp_mst = true;
+	}
 
 	generic_hdmi_init_per_pins(codec);
 

sound/pci/hda/patch_realtek.c

@@ -282,6 +282,7 @@ static void alc_eapd_shutup(struct hda_codec *codec)
 {
 	alc_auto_setup_eapd(codec, false);
 	msleep(200);
+	snd_hda_shutup_pins(codec);
 }
 
 /* generic EAPD initialization */
@@ -826,7 +827,8 @@ static inline void alc_shutup(struct hda_codec *codec)
 
 	if (spec && spec->shutup)
 		spec->shutup(codec);
-	snd_hda_shutup_pins(codec);
+	else
+		snd_hda_shutup_pins(codec);
 }
 
 #define alc_free	snd_hda_gen_free
@@ -1853,8 +1855,10 @@ static void alc882_fixup_no_primary_hp(struct hda_codec *codec,
 				       const struct hda_fixup *fix, int action)
 {
 	struct alc_spec *spec = codec->spec;
-	if (action == HDA_FIXUP_ACT_PRE_PROBE)
+	if (action == HDA_FIXUP_ACT_PRE_PROBE) {
 		spec->gen.no_primary_hp = 1;
+		spec->gen.no_multi_io = 1;
+	}
 }
 
 static const struct hda_fixup alc882_fixups[] = {
@@ -2533,6 +2537,7 @@ enum {
 	ALC269_TYPE_ALC269VD,
 	ALC269_TYPE_ALC280,
 	ALC269_TYPE_ALC282,
+	ALC269_TYPE_ALC283,
 	ALC269_TYPE_ALC284,
 	ALC269_TYPE_ALC286,
 };
@@ -2558,6 +2563,7 @@ static int alc269_parse_auto_config(struct hda_codec *codec)
 	case ALC269_TYPE_ALC269VB:
 	case ALC269_TYPE_ALC269VD:
 	case ALC269_TYPE_ALC282:
+	case ALC269_TYPE_ALC283:
 	case ALC269_TYPE_ALC286:
 		ssids = alc269_ssids;
 		break;
@@ -2583,15 +2589,81 @@ static void alc269_shutup(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
 
-	if (spec->codec_variant != ALC269_TYPE_ALC269VB)
-		return;
-
 	if (spec->codec_variant == ALC269_TYPE_ALC269VB)
 		alc269vb_toggle_power_output(codec, 0);
 	if (spec->codec_variant == ALC269_TYPE_ALC269VB &&
 			(alc_get_coef0(codec) & 0x00ff) == 0x018) {
 		msleep(150);
 	}
+	snd_hda_shutup_pins(codec);
+}
+
+static void alc283_init(struct hda_codec *codec)
+{
+	struct alc_spec *spec = codec->spec;
+	hda_nid_t hp_pin = spec->gen.autocfg.hp_pins[0];
+	bool hp_pin_sense;
+	int val;
+
+	if (!hp_pin)
+		return;
+	hp_pin_sense = snd_hda_jack_detect(codec, hp_pin);
+
+	/* Index 0x43 Direct Drive HP AMP LPM Control 1 */
+	/* Headphone capless set to high power mode */
+	alc_write_coef_idx(codec, 0x43, 0x9004);
+
+	snd_hda_codec_write(codec, hp_pin, 0,
+			    AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE);
+
+	if (hp_pin_sense)
+		msleep(85);
+
+	snd_hda_codec_write(codec, hp_pin, 0,
+			    AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT);
+
+	if (hp_pin_sense)
+		msleep(85);
+	/* Index 0x46 Combo jack auto switch control 2 */
+	/* 3k pull low control for Headset jack. */
+	val = alc_read_coef_idx(codec, 0x46);
+	alc_write_coef_idx(codec, 0x46, val & ~(3 << 12));
+	/* Headphone capless set to normal mode */
+	alc_write_coef_idx(codec, 0x43, 0x9614);
+}
+
+static void alc283_shutup(struct hda_codec *codec)
+{
+	struct alc_spec *spec = codec->spec;
+	hda_nid_t hp_pin = spec->gen.autocfg.hp_pins[0];
+	bool hp_pin_sense;
+	int val;
+
+	if (!hp_pin) {
+		alc269_shutup(codec);
+		return;
+	}
+
+	hp_pin_sense = snd_hda_jack_detect(codec, hp_pin);
+
+	alc_write_coef_idx(codec, 0x43, 0x9004);
+
+	snd_hda_codec_write(codec, hp_pin, 0,
+			    AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE);
+
+	if (hp_pin_sense)
+		msleep(85);
+
+	snd_hda_codec_write(codec, hp_pin, 0,
+			    AC_VERB_SET_PIN_WIDGET_CONTROL, 0x0);
+
+	val = alc_read_coef_idx(codec, 0x46);
+	alc_write_coef_idx(codec, 0x46, val | (3 << 12));
+
+	if (hp_pin_sense)
+		msleep(85);
+	snd_hda_shutup_pins(codec);
+	alc_write_coef_idx(codec, 0x43, 0x9614);
 }
 
 static void alc5505_coef_set(struct hda_codec *codec, unsigned int index_reg,
@@ -2722,6 +2794,7 @@ static int alc269_resume(struct hda_codec *codec)
 	hda_call_check_power_status(codec, 0x01);
 	if (spec->has_alc5505_dsp)
 		alc5505_dsp_resume(codec);
+
 	return 0;
 }
 #endif /* CONFIG_PM */
@@ -3261,6 +3334,28 @@ static void alc_fixup_headset_mode_alc668(struct hda_codec *codec,
 	alc_fixup_headset_mode(codec, fix, action);
 }
 
+/* Returns the nid of the external mic input pin, or 0 if it cannot be found. */
+static int find_ext_mic_pin(struct hda_codec *codec)
+{
+	struct alc_spec *spec = codec->spec;
+	struct auto_pin_cfg *cfg = &spec->gen.autocfg;
+	hda_nid_t nid;
+	unsigned int defcfg;
+	int i;
+
+	for (i = 0; i < cfg->num_inputs; i++) {
+		if (cfg->inputs[i].type != AUTO_PIN_MIC)
+			continue;
+		nid = cfg->inputs[i].pin;
+		defcfg = snd_hda_codec_get_pincfg(codec, nid);
+		if (snd_hda_get_input_pin_attr(defcfg) == INPUT_PIN_ATTR_INT)
+			continue;
+		return nid;
+	}
+
+	return 0;
+}
+
 static void alc271_hp_gate_mic_jack(struct hda_codec *codec,
 				    const struct hda_fixup *fix,
 				    int action)
@@ -3268,11 +3363,12 @@ static void alc271_hp_gate_mic_jack(struct hda_codec *codec,
 	struct alc_spec *spec = codec->spec;
 
 	if (action == HDA_FIXUP_ACT_PROBE) {
-		if (snd_BUG_ON(!spec->gen.am_entry[1].pin ||
-			       !spec->gen.autocfg.hp_pins[0]))
+		int mic_pin = find_ext_mic_pin(codec);
+		int hp_pin = spec->gen.autocfg.hp_pins[0];
+
+		if (snd_BUG_ON(!mic_pin || !hp_pin))
 			return;
-		snd_hda_jack_set_gating_jack(codec, spec->gen.am_entry[1].pin,
-					     spec->gen.autocfg.hp_pins[0]);
+		snd_hda_jack_set_gating_jack(codec, mic_pin, hp_pin);
 	}
 }
 
@@ -3308,6 +3404,45 @@ static void alc269_fixup_limit_int_mic_boost(struct hda_codec *codec,
 	}
 }
 
+static void alc283_hp_automute_hook(struct hda_codec *codec,
+				    struct hda_jack_tbl *jack)
+{
+	struct alc_spec *spec = codec->spec;
+	int vref;
+
+	msleep(200);
+	snd_hda_gen_hp_automute(codec, jack);
+
+	vref = spec->gen.hp_jack_present ? PIN_VREF80 : 0;
+
+	msleep(600);
+	snd_hda_codec_write(codec, 0x19, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
+			    vref);
+}
+
+static void alc283_chromebook_caps(struct hda_codec *codec)
+{
+	snd_hda_override_wcaps(codec, 0x03, 0);
+}
+
+static void alc283_fixup_chromebook(struct hda_codec *codec,
+				    const struct hda_fixup *fix, int action)
+{
+	struct alc_spec *spec = codec->spec;
+	int val;
+
+	switch (action) {
+	case HDA_FIXUP_ACT_PRE_PROBE:
+		alc283_chromebook_caps(codec);
+		spec->gen.hp_automute_hook = alc283_hp_automute_hook;
+		/* MIC2-VREF control */
+		/* Set to manual mode */
+		val = alc_read_coef_idx(codec, 0x06);
+		alc_write_coef_idx(codec, 0x06, val & ~0x000c);
+		break;
+	}
+}
+
 enum {
 	ALC269_FIXUP_SONY_VAIO,
 	ALC275_FIXUP_SONY_VAIO_GPIO2,
@@ -3344,6 +3479,7 @@ enum {
 	ALC269_FIXUP_ACER_AC700,
 	ALC269_FIXUP_LIMIT_INT_MIC_BOOST,
 	ALC269VB_FIXUP_ORDISSIMO_EVE2,
+	ALC283_FIXUP_CHROME_BOOK,
 };
 
 static const struct hda_fixup alc269_fixups[] = {
@@ -3595,11 +3731,20 @@ static const struct hda_fixup alc269_fixups[] = {
 			{ }
 		},
 	},
+	[ALC283_FIXUP_CHROME_BOOK] = {
+		.type = HDA_FIXUP_FUNC,
+		.v.func = alc283_fixup_chromebook,
+	},
 };
 
 static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1025, 0x029b, "Acer 1810TZ", ALC269_FIXUP_INV_DMIC),
 	SND_PCI_QUIRK(0x1025, 0x0349, "Acer AOD260", ALC269_FIXUP_INV_DMIC),
+	SND_PCI_QUIRK(0x1025, 0x047c, "Acer AC700", ALC269_FIXUP_ACER_AC700),
+	SND_PCI_QUIRK(0x1025, 0x0740, "Acer AO725", ALC271_FIXUP_HP_GATE_MIC_JACK),
+	SND_PCI_QUIRK(0x1025, 0x0742, "Acer AO756", ALC271_FIXUP_HP_GATE_MIC_JACK),
+	SND_PCI_QUIRK_VENDOR(0x1025, "Acer Aspire", ALC271_FIXUP_DMIC),
+	SND_PCI_QUIRK(0x1028, 0x0470, "Dell M101z", ALC269_FIXUP_DELL_M101Z),
 	SND_PCI_QUIRK(0x1028, 0x05bd, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1028, 0x05be, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1028, 0x05c4, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE),
@@ -3637,6 +3782,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x103c, 0x18e6, "HP", ALC269_FIXUP_HP_GPIO_LED),
 	SND_PCI_QUIRK(0x103c, 0x1973, "HP Pavilion", ALC269_FIXUP_HP_MUTE_LED_MIC1),
 	SND_PCI_QUIRK(0x103c, 0x1983, "HP Pavilion", ALC269_FIXUP_HP_MUTE_LED_MIC1),
+	SND_PCI_QUIRK(0x103c, 0x21ed, "HP Falco Chromebook", ALC283_FIXUP_CHROME_BOOK),
 	SND_PCI_QUIRK_VENDOR(0x103c, "HP", ALC269_FIXUP_HP_MUTE_LED),
 	SND_PCI_QUIRK(0x1043, 0x106d, "Asus K53BE", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
 	SND_PCI_QUIRK(0x1043, 0x115d, "Asus 1015E", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
@@ -3655,11 +3801,6 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x104d, 0x907b, "Sony VAIO", ALC275_FIXUP_SONY_HWEQ),
 	SND_PCI_QUIRK(0x104d, 0x9084, "Sony VAIO", ALC275_FIXUP_SONY_HWEQ),
 	SND_PCI_QUIRK_VENDOR(0x104d, "Sony VAIO", ALC269_FIXUP_SONY_VAIO),
-	SND_PCI_QUIRK(0x1028, 0x0470, "Dell M101z", ALC269_FIXUP_DELL_M101Z),
-	SND_PCI_QUIRK(0x1025, 0x047c, "Acer AC700", ALC269_FIXUP_ACER_AC700),
-	SND_PCI_QUIRK(0x1025, 0x0740, "Acer AO725", ALC271_FIXUP_HP_GATE_MIC_JACK),
-	SND_PCI_QUIRK(0x1025, 0x0742, "Acer AO756", ALC271_FIXUP_HP_GATE_MIC_JACK),
-	SND_PCI_QUIRK_VENDOR(0x1025, "Acer Aspire", ALC271_FIXUP_DMIC),
 	SND_PCI_QUIRK(0x10cf, 0x1475, "Lifebook", ALC269_FIXUP_LIFEBOOK),
 	SND_PCI_QUIRK(0x17aa, 0x20f2, "Thinkpad SL410/510", ALC269_FIXUP_SKU_IGNORE),
 	SND_PCI_QUIRK(0x17aa, 0x215e, "Thinkpad L512", ALC269_FIXUP_SKU_IGNORE),
@@ -3670,8 +3811,16 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x17aa, 0x21fa, "Thinkpad X230", ALC269_FIXUP_LENOVO_DOCK),
 	SND_PCI_QUIRK(0x17aa, 0x21f3, "Thinkpad T430", ALC269_FIXUP_LENOVO_DOCK),
 	SND_PCI_QUIRK(0x17aa, 0x21fb, "Thinkpad T430s", ALC269_FIXUP_LENOVO_DOCK),
-	SND_PCI_QUIRK(0x17aa, 0x2208, "Thinkpad T431s", ALC269_FIXUP_LENOVO_DOCK),
 	SND_PCI_QUIRK(0x17aa, 0x2203, "Thinkpad X230 Tablet", ALC269_FIXUP_LENOVO_DOCK),
+	SND_PCI_QUIRK(0x17aa, 0x2208, "Thinkpad T431s", ALC269_FIXUP_LENOVO_DOCK),
+	SND_PCI_QUIRK(0x17aa, 0x220c, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
+	SND_PCI_QUIRK(0x17aa, 0x2212, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
+	SND_PCI_QUIRK(0x17aa, 0x2214, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
+	SND_PCI_QUIRK(0x17aa, 0x2215, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
+	SND_PCI_QUIRK(0x17aa, 0x5013, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
+	SND_PCI_QUIRK(0x17aa, 0x501a, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
+	SND_PCI_QUIRK(0x17aa, 0x5026, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
+	SND_PCI_QUIRK(0x17aa, 0x5109, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
 	SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_PCM_44K),
 	SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD),
 	SND_PCI_QUIRK(0x1b7d, 0xa831, "Ordissimo EVE2 ", ALC269VB_FIXUP_ORDISSIMO_EVE2), /* Also known as Malata PC-B1303 */
@@ -3840,11 +3989,15 @@ static int patch_alc269(struct hda_codec *codec)
 	case 0x10ec0290:
 		spec->codec_variant = ALC269_TYPE_ALC280;
 		break;
-	case 0x10ec0233:
 	case 0x10ec0282:
-	case 0x10ec0283:
 		spec->codec_variant = ALC269_TYPE_ALC282;
 		break;
+	case 0x10ec0233:
+	case 0x10ec0283:
+		spec->codec_variant = ALC269_TYPE_ALC283;
+		spec->shutup = alc283_shutup;
+		spec->init_hook = alc283_init;
+		break;
 	case 0x10ec0284:
 	case 0x10ec0292:
 		spec->codec_variant = ALC269_TYPE_ALC284;
@@ -3872,7 +4025,8 @@ static int patch_alc269(struct hda_codec *codec)
 	codec->patch_ops.suspend = alc269_suspend;
 	codec->patch_ops.resume = alc269_resume;
 #endif
-	spec->shutup = alc269_shutup;
+	if (!spec->shutup)
+		spec->shutup = alc269_shutup;
 
 	snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PROBE);
 

sound/pci/hda/patch_sigmatel.c

@@ -158,6 +158,7 @@ enum {
 	STAC_D965_VERBS,
 	STAC_DELL_3ST,
 	STAC_DELL_BIOS,
+	STAC_DELL_BIOS_AMIC,
 	STAC_DELL_BIOS_SPDIF,
 	STAC_927X_DELL_DMIC,
 	STAC_927X_VOLKNOB,
@@ -3231,8 +3232,6 @@ static const struct hda_fixup stac927x_fixups[] = {
 	[STAC_DELL_BIOS] = {
 		.type = HDA_FIXUP_PINS,
 		.v.pins = (const struct hda_pintbl[]) {
-			/* configure the analog microphone on some laptops */
-			{ 0x0c, 0x90a79130 },
 			/* correct the front output jack as a hp out */
 			{ 0x0f, 0x0221101f },
 			/* correct the front input jack as a mic */
@@ -3242,6 +3241,16 @@ static const struct hda_fixup stac927x_fixups[] = {
 		.chained = true,
 		.chain_id = STAC_927X_DELL_DMIC,
 	},
+	[STAC_DELL_BIOS_AMIC] = {
+		.type = HDA_FIXUP_PINS,
+		.v.pins = (const struct hda_pintbl[]) {
+			/* configure the analog microphone on some laptops */
+			{ 0x0c, 0x90a79130 },
+			{}
+		},
+		.chained = true,
+		.chain_id = STAC_DELL_BIOS,
+	},
 	[STAC_DELL_BIOS_SPDIF] = {
 		.type = HDA_FIXUP_PINS,
 		.v.pins = (const struct hda_pintbl[]) {
@@ -3270,6 +3279,7 @@ static const struct hda_model_fixup stac927x_models[] = {
 	{ .id = STAC_D965_5ST_NO_FP, .name = "5stack-no-fp" },
 	{ .id = STAC_DELL_3ST, .name = "dell-3stack" },
 	{ .id = STAC_DELL_BIOS, .name = "dell-bios" },
+	{ .id = STAC_DELL_BIOS_AMIC, .name = "dell-bios-amic" },
 	{ .id = STAC_927X_VOLKNOB, .name = "volknob" },
 	{}
 };

sound/pci/hda/patch_via.c

@@ -207,9 +207,9 @@ static void vt1708_stop_hp_work(struct hda_codec *codec)
 		return;
 	if (spec->hp_work_active) {
 		snd_hda_codec_write(codec, 0x1, 0, 0xf81, 1);
+		codec->jackpoll_interval = 0;
 		cancel_delayed_work_sync(&codec->jackpoll_work);
 		spec->hp_work_active = false;
-		codec->jackpoll_interval = 0;
 	}
 }
 

sound/pci/rme96.c

@@ -28,6 +28,7 @@
 #include <linux/interrupt.h>
 #include <linux/pci.h>
 #include <linux/module.h>
+#include <linux/vmalloc.h>
 
 #include <sound/core.h>
 #include <sound/info.h>
@@ -198,6 +199,31 @@ MODULE_PARM_DESC(enable, "Enable RME Digi96 soundcard.");
 #define RME96_AD1852_VOL_BITS 14
 #define RME96_AD1855_VOL_BITS 10
 
+/* Defines for snd_rme96_trigger */
+#define RME96_TB_START_PLAYBACK 1
+#define RME96_TB_START_CAPTURE 2
+#define RME96_TB_STOP_PLAYBACK 4
+#define RME96_TB_STOP_CAPTURE 8
+#define RME96_TB_RESET_PLAYPOS 16
+#define RME96_TB_RESET_CAPTUREPOS 32
+#define RME96_TB_CLEAR_PLAYBACK_IRQ 64
+#define RME96_TB_CLEAR_CAPTURE_IRQ 128
+#define RME96_RESUME_PLAYBACK	(RME96_TB_START_PLAYBACK)
+#define RME96_RESUME_CAPTURE	(RME96_TB_START_CAPTURE)
+#define RME96_RESUME_BOTH	(RME96_RESUME_PLAYBACK \
+				| RME96_RESUME_CAPTURE)
+#define RME96_START_PLAYBACK	(RME96_TB_START_PLAYBACK \
+				| RME96_TB_RESET_PLAYPOS)
+#define RME96_START_CAPTURE	(RME96_TB_START_CAPTURE \
+				| RME96_TB_RESET_CAPTUREPOS)
+#define RME96_START_BOTH	(RME96_START_PLAYBACK \
+				| RME96_START_CAPTURE)
+#define RME96_STOP_PLAYBACK	(RME96_TB_STOP_PLAYBACK \
+				| RME96_TB_CLEAR_PLAYBACK_IRQ)
+#define RME96_STOP_CAPTURE	(RME96_TB_STOP_CAPTURE \
+				| RME96_TB_CLEAR_CAPTURE_IRQ)
+#define RME96_STOP_BOTH		(RME96_STOP_PLAYBACK \
+				| RME96_STOP_CAPTURE)
 
 struct rme96 {
 	spinlock_t    lock;
@@ -214,6 +240,13 @@ struct rme96 {
 
 	u8 rev; /* card revision number */
 
+#ifdef CONFIG_PM
+	u32 playback_pointer;
+	u32 capture_pointer;
+	void *playback_suspend_buffer;
+	void *capture_suspend_buffer;
+#endif
+
 	struct snd_pcm_substream *playback_substream;
 	struct snd_pcm_substream *capture_substream;
 
@@ -344,6 +377,8 @@ static struct snd_pcm_hardware snd_rme96_playback_spdif_info =
 {
 	.info =		     (SNDRV_PCM_INFO_MMAP_IOMEM |
 			      SNDRV_PCM_INFO_MMAP_VALID |
+			      SNDRV_PCM_INFO_SYNC_START |
+			      SNDRV_PCM_INFO_RESUME |
 			      SNDRV_PCM_INFO_INTERLEAVED |
 			      SNDRV_PCM_INFO_PAUSE),
 	.formats =	     (SNDRV_PCM_FMTBIT_S16_LE |
@@ -373,6 +408,8 @@ static struct snd_pcm_hardware snd_rme96_capture_spdif_info =
 {
 	.info =		     (SNDRV_PCM_INFO_MMAP_IOMEM |
 			      SNDRV_PCM_INFO_MMAP_VALID |
+			      SNDRV_PCM_INFO_SYNC_START |
+			      SNDRV_PCM_INFO_RESUME |
 			      SNDRV_PCM_INFO_INTERLEAVED |
 			      SNDRV_PCM_INFO_PAUSE),
 	.formats =	     (SNDRV_PCM_FMTBIT_S16_LE |
@@ -402,6 +439,8 @@ static struct snd_pcm_hardware snd_rme96_playback_adat_info =
 {
 	.info =		     (SNDRV_PCM_INFO_MMAP_IOMEM |
 			      SNDRV_PCM_INFO_MMAP_VALID |
+			      SNDRV_PCM_INFO_SYNC_START |
+			      SNDRV_PCM_INFO_RESUME |
 			      SNDRV_PCM_INFO_INTERLEAVED |
 			      SNDRV_PCM_INFO_PAUSE),
 	.formats =	     (SNDRV_PCM_FMTBIT_S16_LE |
@@ -427,6 +466,8 @@ static struct snd_pcm_hardware snd_rme96_capture_adat_info =
 {
 	.info =		     (SNDRV_PCM_INFO_MMAP_IOMEM |
 			      SNDRV_PCM_INFO_MMAP_VALID |
+			      SNDRV_PCM_INFO_SYNC_START |
+			      SNDRV_PCM_INFO_RESUME |
 			      SNDRV_PCM_INFO_INTERLEAVED |
 			      SNDRV_PCM_INFO_PAUSE),
 	.formats =	     (SNDRV_PCM_FMTBIT_S16_LE |
@@ -1045,54 +1086,35 @@ snd_rme96_capture_hw_params(struct snd_pcm_substream *substream,
 }
 
 static void
-snd_rme96_playback_start(struct rme96 *rme96,
-			 int from_pause)
+snd_rme96_trigger(struct rme96 *rme96,
+		  int op)
 {
-	if (!from_pause) {
+	if (op & RME96_TB_RESET_PLAYPOS)
 		writel(0, rme96->iobase + RME96_IO_RESET_PLAY_POS);
-	}
-
-	rme96->wcreg |= RME96_WCR_START;
-	writel(rme96->wcreg, rme96->iobase + RME96_IO_CONTROL_REGISTER);
-}
-
-static void
-snd_rme96_capture_start(struct rme96 *rme96,
-			int from_pause)
-{
-	if (!from_pause) {
+	if (op & RME96_TB_RESET_CAPTUREPOS)
 		writel(0, rme96->iobase + RME96_IO_RESET_REC_POS);
-	}
-
-	rme96->wcreg |= RME96_WCR_START_2;
+	if (op & RME96_TB_CLEAR_PLAYBACK_IRQ) {
+		rme96->rcreg = readl(rme96->iobase + RME96_IO_CONTROL_REGISTER);
+		if (rme96->rcreg & RME96_RCR_IRQ)
+			writel(0, rme96->iobase + RME96_IO_CONFIRM_PLAY_IRQ);
+	}
+	if (op & RME96_TB_CLEAR_CAPTURE_IRQ) {
+		rme96->rcreg = readl(rme96->iobase + RME96_IO_CONTROL_REGISTER);
+		if (rme96->rcreg & RME96_RCR_IRQ_2)
+			writel(0, rme96->iobase + RME96_IO_CONFIRM_REC_IRQ);
+	}
+	if (op & RME96_TB_START_PLAYBACK)
+		rme96->wcreg |= RME96_WCR_START;
+	if (op & RME96_TB_STOP_PLAYBACK)
+		rme96->wcreg &= ~RME96_WCR_START;
+	if (op & RME96_TB_START_CAPTURE)
+		rme96->wcreg |= RME96_WCR_START_2;
+	if (op & RME96_TB_STOP_CAPTURE)
+		rme96->wcreg &= ~RME96_WCR_START_2;
 	writel(rme96->wcreg, rme96->iobase + RME96_IO_CONTROL_REGISTER);
 }
 
-static void
-snd_rme96_playback_stop(struct rme96 *rme96)
-{
-	/*
-	 * Check if there is an unconfirmed IRQ, if so confirm it, or else
-	 * the hardware will not stop generating interrupts
-	 */
-	rme96->rcreg = readl(rme96->iobase + RME96_IO_CONTROL_REGISTER);
-	if (rme96->rcreg & RME96_RCR_IRQ) {
-		writel(0, rme96->iobase + RME96_IO_CONFIRM_PLAY_IRQ);
-	}	
-	rme96->wcreg &= ~RME96_WCR_START;
-	writel(rme96->wcreg, rme96->iobase + RME96_IO_CONTROL_REGISTER);
-}
 
-static void
-snd_rme96_capture_stop(struct rme96 *rme96)
-{
-	rme96->rcreg = readl(rme96->iobase + RME96_IO_CONTROL_REGISTER);
-	if (rme96->rcreg & RME96_RCR_IRQ_2) {
-		writel(0, rme96->iobase + RME96_IO_CONFIRM_REC_IRQ);
-	}	
-	rme96->wcreg &= ~RME96_WCR_START_2;
-	writel(rme96->wcreg, rme96->iobase + RME96_IO_CONTROL_REGISTER);
-}
 
 static irqreturn_t
 snd_rme96_interrupt(int irq,
@@ -1155,6 +1177,7 @@ snd_rme96_playback_spdif_open(struct snd_pcm_substream *substream)
 	struct rme96 *rme96 = snd_pcm_substream_chip(substream);
 	struct snd_pcm_runtime *runtime = substream->runtime;
 
+	snd_pcm_set_sync(substream);
 	spin_lock_irq(&rme96->lock);	
         if (rme96->playback_substream != NULL) {
 		spin_unlock_irq(&rme96->lock);
@@ -1191,6 +1214,7 @@ snd_rme96_capture_spdif_open(struct snd_pcm_substream *substream)
 	struct rme96 *rme96 = snd_pcm_substream_chip(substream);
 	struct snd_pcm_runtime *runtime = substream->runtime;
 
+	snd_pcm_set_sync(substream);
 	runtime->hw = snd_rme96_capture_spdif_info;
         if (snd_rme96_getinputtype(rme96) != RME96_INPUT_ANALOG &&
             (rate = snd_rme96_capture_getrate(rme96, &isadat)) > 0)
@@ -1222,6 +1246,7 @@ snd_rme96_playback_adat_open(struct snd_pcm_substream *substream)
 	struct rme96 *rme96 = snd_pcm_substream_chip(substream);
 	struct snd_pcm_runtime *runtime = substream->runtime;        
 	
+	snd_pcm_set_sync(substream);
 	spin_lock_irq(&rme96->lock);	
         if (rme96->playback_substream != NULL) {
 		spin_unlock_irq(&rme96->lock);
@@ -1253,6 +1278,7 @@ snd_rme96_capture_adat_open(struct snd_pcm_substream *substream)
 	struct rme96 *rme96 = snd_pcm_substream_chip(substream);
 	struct snd_pcm_runtime *runtime = substream->runtime;
 
+	snd_pcm_set_sync(substream);
 	runtime->hw = snd_rme96_capture_adat_info;
         if (snd_rme96_getinputtype(rme96) == RME96_INPUT_ANALOG) {
                 /* makes no sense to use analog input. Note that analog
@@ -1288,7 +1314,7 @@ snd_rme96_playback_close(struct snd_pcm_substream *substream)
 
 	spin_lock_irq(&rme96->lock);	
 	if (RME96_ISPLAYING(rme96)) {
-		snd_rme96_playback_stop(rme96);
+		snd_rme96_trigger(rme96, RME96_STOP_PLAYBACK);
 	}
 	rme96->playback_substream = NULL;
 	rme96->playback_periodsize = 0;
@@ -1309,7 +1335,7 @@ snd_rme96_capture_close(struct snd_pcm_substream *substream)
 	
 	spin_lock_irq(&rme96->lock);	
 	if (RME96_ISRECORDING(rme96)) {
-		snd_rme96_capture_stop(rme96);
+		snd_rme96_trigger(rme96, RME96_STOP_CAPTURE);
 	}
 	rme96->capture_substream = NULL;
 	rme96->capture_periodsize = 0;
@@ -1324,7 +1350,7 @@ snd_rme96_playback_prepare(struct snd_pcm_substream *substream)
 	
 	spin_lock_irq(&rme96->lock);	
 	if (RME96_ISPLAYING(rme96)) {
-		snd_rme96_playback_stop(rme96);
+		snd_rme96_trigger(rme96, RME96_STOP_PLAYBACK);
 	}
 	writel(0, rme96->iobase + RME96_IO_RESET_PLAY_POS);
 	spin_unlock_irq(&rme96->lock);
@@ -1338,7 +1364,7 @@ snd_rme96_capture_prepare(struct snd_pcm_substream *substream)
 	
 	spin_lock_irq(&rme96->lock);	
 	if (RME96_ISRECORDING(rme96)) {
-		snd_rme96_capture_stop(rme96);
+		snd_rme96_trigger(rme96, RME96_STOP_CAPTURE);
 	}
 	writel(0, rme96->iobase + RME96_IO_RESET_REC_POS);
 	spin_unlock_irq(&rme96->lock);
@@ -1350,41 +1376,55 @@ snd_rme96_playback_trigger(struct snd_pcm_substream *substream,
 			   int cmd)
 {
 	struct rme96 *rme96 = snd_pcm_substream_chip(substream);
+	struct snd_pcm_substream *s;
+	bool sync;
+
+	snd_pcm_group_for_each_entry(s, substream) {
+		if (snd_pcm_substream_chip(s) == rme96)
+			snd_pcm_trigger_done(s, substream);
+	}
+
+	sync = (rme96->playback_substream && rme96->capture_substream) &&
+	       (rme96->playback_substream->group ==
+		rme96->capture_substream->group);
 
 	switch (cmd) {
 	case SNDRV_PCM_TRIGGER_START:
 		if (!RME96_ISPLAYING(rme96)) {
-			if (substream != rme96->playback_substream) {
+			if (substream != rme96->playback_substream)
 				return -EBUSY;
-			}
-			snd_rme96_playback_start(rme96, 0);
+			snd_rme96_trigger(rme96, sync ? RME96_START_BOTH
+						 : RME96_START_PLAYBACK);
 		}
 		break;
 
+	case SNDRV_PCM_TRIGGER_SUSPEND:
 	case SNDRV_PCM_TRIGGER_STOP:
 		if (RME96_ISPLAYING(rme96)) {
-			if (substream != rme96->playback_substream) {
+			if (substream != rme96->playback_substream)
 				return -EBUSY;
-			}
-			snd_rme96_playback_stop(rme96);
+			snd_rme96_trigger(rme96, sync ? RME96_STOP_BOTH
+						 :  RME96_STOP_PLAYBACK);
 		}
 		break;
 
 	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
-		if (RME96_ISPLAYING(rme96)) {
-			snd_rme96_playback_stop(rme96);
-		}
+		if (RME96_ISPLAYING(rme96))
+			snd_rme96_trigger(rme96, sync ? RME96_STOP_BOTH
+						 : RME96_STOP_PLAYBACK);
 		break;
 
+	case SNDRV_PCM_TRIGGER_RESUME:
 	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
-		if (!RME96_ISPLAYING(rme96)) {
-			snd_rme96_playback_start(rme96, 1);
-		}
+		if (!RME96_ISPLAYING(rme96))
+			snd_rme96_trigger(rme96, sync ? RME96_RESUME_BOTH
+						 : RME96_RESUME_PLAYBACK);
 		break;
-		
+
 	default:
 		return -EINVAL;
 	}
+
 	return 0;
 }
 
@@ -1393,38 +1433,51 @@ snd_rme96_capture_trigger(struct snd_pcm_substream *substream,
 			  int cmd)
 {
 	struct rme96 *rme96 = snd_pcm_substream_chip(substream);
+	struct snd_pcm_substream *s;
+	bool sync;
+
+	snd_pcm_group_for_each_entry(s, substream) {
+		if (snd_pcm_substream_chip(s) == rme96)
+			snd_pcm_trigger_done(s, substream);
+	}
+
+	sync = (rme96->playback_substream && rme96->capture_substream) &&
+	       (rme96->playback_substream->group ==
+		rme96->capture_substream->group);
 
 	switch (cmd) {
 	case SNDRV_PCM_TRIGGER_START:
 		if (!RME96_ISRECORDING(rme96)) {
-			if (substream != rme96->capture_substream) {
+			if (substream != rme96->capture_substream)
 				return -EBUSY;
-			}
-			snd_rme96_capture_start(rme96, 0);
+			snd_rme96_trigger(rme96, sync ? RME96_START_BOTH
+						 : RME96_START_CAPTURE);
 		}
 		break;
 
+	case SNDRV_PCM_TRIGGER_SUSPEND:
 	case SNDRV_PCM_TRIGGER_STOP:
 		if (RME96_ISRECORDING(rme96)) {
-			if (substream != rme96->capture_substream) {
+			if (substream != rme96->capture_substream)
 				return -EBUSY;
-			}
-			snd_rme96_capture_stop(rme96);
+			snd_rme96_trigger(rme96, sync ? RME96_STOP_BOTH
+						 : RME96_STOP_CAPTURE);
 		}
 		break;
 
 	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
-		if (RME96_ISRECORDING(rme96)) {
-			snd_rme96_capture_stop(rme96);
-		}
+		if (RME96_ISRECORDING(rme96))
+			snd_rme96_trigger(rme96, sync ? RME96_STOP_BOTH
+						 : RME96_STOP_CAPTURE);
 		break;
 
+	case SNDRV_PCM_TRIGGER_RESUME:
 	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
-		if (!RME96_ISRECORDING(rme96)) {
-			snd_rme96_capture_start(rme96, 1);
-		}
+		if (!RME96_ISRECORDING(rme96))
+			snd_rme96_trigger(rme96, sync ? RME96_RESUME_BOTH
+						 : RME96_RESUME_CAPTURE);
 		break;
-		
+
 	default:
 		return -EINVAL;
 	}
@@ -1505,8 +1558,7 @@ snd_rme96_free(void *private_data)
 	        return;
 	}
 	if (rme96->irq >= 0) {
-		snd_rme96_playback_stop(rme96);
-		snd_rme96_capture_stop(rme96);
+		snd_rme96_trigger(rme96, RME96_STOP_BOTH);
 		rme96->areg &= ~RME96_AR_DAC_EN;
 		writel(rme96->areg, rme96->iobase + RME96_IO_ADDITIONAL_REG);
 		free_irq(rme96->irq, (void *)rme96);
@@ -1520,6 +1572,10 @@ snd_rme96_free(void *private_data)
 		pci_release_regions(rme96->pci);
 		rme96->port = 0;
 	}
+#ifdef CONFIG_PM
+	vfree(rme96->playback_suspend_buffer);
+	vfree(rme96->capture_suspend_buffer);
+#endif
 	pci_disable_device(rme96->pci);
 }
 
@@ -1606,8 +1662,7 @@ snd_rme96_create(struct rme96 *rme96)
 	rme96->capture_periodsize = 0;
 	
 	/* make sure playback/capture is stopped, if by some reason active */
-	snd_rme96_playback_stop(rme96);
-	snd_rme96_capture_stop(rme96);
+	snd_rme96_trigger(rme96, RME96_STOP_BOTH);
 	
 	/* set default values in registers */
 	rme96->wcreg =
@@ -2319,6 +2374,87 @@ snd_rme96_create_switches(struct snd_card *card,
  * Card initialisation
  */
 
+#ifdef CONFIG_PM
+
+static int
+snd_rme96_suspend(struct pci_dev *pci,
+		  pm_message_t state)
+{
+	struct snd_card *card = pci_get_drvdata(pci);
+	struct rme96 *rme96 = card->private_data;
+
+	snd_power_change_state(card, SNDRV_CTL_POWER_D3hot);
+	snd_pcm_suspend(rme96->playback_substream);
+	snd_pcm_suspend(rme96->capture_substream);
+
+	/* save capture & playback pointers */
+	rme96->playback_pointer = readl(rme96->iobase + RME96_IO_GET_PLAY_POS)
+				  & RME96_RCR_AUDIO_ADDR_MASK;
+	rme96->capture_pointer = readl(rme96->iobase + RME96_IO_GET_REC_POS)
+				 & RME96_RCR_AUDIO_ADDR_MASK;
+
+	/* save playback and capture buffers */
+	memcpy_fromio(rme96->playback_suspend_buffer,
+		      rme96->iobase + RME96_IO_PLAY_BUFFER, RME96_BUFFER_SIZE);
+	memcpy_fromio(rme96->capture_suspend_buffer,
+		      rme96->iobase + RME96_IO_REC_BUFFER, RME96_BUFFER_SIZE);
+
+	/* disable the DAC  */
+	rme96->areg &= ~RME96_AR_DAC_EN;
+	writel(rme96->areg, rme96->iobase + RME96_IO_ADDITIONAL_REG);
+
+	pci_disable_device(pci);
+	pci_save_state(pci);
+
+	return 0;
+}
+
+static int
+snd_rme96_resume(struct pci_dev *pci)
+{
+	struct snd_card *card = pci_get_drvdata(pci);
+	struct rme96 *rme96 = card->private_data;
+
+	pci_restore_state(pci);
+	if (pci_enable_device(pci) < 0) {
+		printk(KERN_ERR "rme96: pci_enable_device failed, disabling device\n");
+		snd_card_disconnect(card);
+		return -EIO;
+	}
+
+	/* reset playback and record buffer pointers */
+	writel(0, rme96->iobase + RME96_IO_SET_PLAY_POS
+		  + rme96->playback_pointer);
+	writel(0, rme96->iobase + RME96_IO_SET_REC_POS
+		  + rme96->capture_pointer);
+
+	/* restore playback and capture buffers */
+	memcpy_toio(rme96->iobase + RME96_IO_PLAY_BUFFER,
+		    rme96->playback_suspend_buffer, RME96_BUFFER_SIZE);
+	memcpy_toio(rme96->iobase + RME96_IO_REC_BUFFER,
+		    rme96->capture_suspend_buffer, RME96_BUFFER_SIZE);
+
+	/* reset the ADC */
+	writel(rme96->areg | RME96_AR_PD2,
+	       rme96->iobase + RME96_IO_ADDITIONAL_REG);
+	writel(rme96->areg, rme96->iobase + RME96_IO_ADDITIONAL_REG);
+
+	/* reset and enable DAC, restore analog volume */
+	snd_rme96_reset_dac(rme96);
+	rme96->areg |= RME96_AR_DAC_EN;
+	writel(rme96->areg, rme96->iobase + RME96_IO_ADDITIONAL_REG);
+	if (RME96_HAS_ANALOG_OUT(rme96)) {
+		usleep_range(3000, 10000);
+		snd_rme96_apply_dac_volume(rme96);
+	}
+
+	snd_power_change_state(card, SNDRV_CTL_POWER_D0);
+
+	return 0;
+}
+
+#endif
+
 static void snd_rme96_card_free(struct snd_card *card)
 {
 	snd_rme96_free(card->private_data);
@@ -2355,6 +2491,23 @@ snd_rme96_probe(struct pci_dev *pci,
 		return err;
 	}
 	
+#ifdef CONFIG_PM
+	rme96->playback_suspend_buffer = vmalloc(RME96_BUFFER_SIZE);
+	if (!rme96->playback_suspend_buffer) {
+		snd_printk(KERN_ERR
+			   "Failed to allocate playback suspend buffer!\n");
+		snd_card_free(card);
+		return -ENOMEM;
+	}
+	rme96->capture_suspend_buffer = vmalloc(RME96_BUFFER_SIZE);
+	if (!rme96->capture_suspend_buffer) {
+		snd_printk(KERN_ERR
+			   "Failed to allocate capture suspend buffer!\n");
+		snd_card_free(card);
+		return -ENOMEM;
+	}
+#endif
+
 	strcpy(card->driver, "Digi96");
 	switch (rme96->pci->device) {
 	case PCI_DEVICE_ID_RME_DIGI96:
@@ -2397,6 +2550,10 @@ static struct pci_driver rme96_driver = {
 	.id_table = snd_rme96_ids,
 	.probe = snd_rme96_probe,
 	.remove = snd_rme96_remove,
+#ifdef CONFIG_PM
+	.suspend = snd_rme96_suspend,
+	.resume = snd_rme96_resume,
+#endif
 };
 
 module_pci_driver(rme96_driver);

sound/pci/rme9652/hdspm.c

@@ -38,6 +38,97 @@
  *   Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307 USA
  *
  */
+
+/* *************    Register Documentation   *******************************************************
+ *
+ * Work in progress! Documentation is based on the code in this file.
+ *
+ * --------- HDSPM_controlRegister ---------
+ * :7654.3210:7654.3210:7654.3210:7654.3210: bit number per byte
+ * :||||.||||:||||.||||:||||.||||:||||.||||:
+ * :3322.2222:2222.1111:1111.1100:0000.0000: bit number
+ * :1098.7654:3210.9876:5432.1098:7654.3210: 0..31
+ * :||||.||||:||||.||||:||||.||||:||||.||||:
+ * :8421.8421:8421.8421:8421.8421:8421.8421: hex digit
+ * :    .    :    .    :    .    :  x .    :  HDSPM_AudioInterruptEnable \_ setting both bits
+ * :    .    :    .    :    .    :    .   x:  HDSPM_Start                /  enables audio IO
+ * :    .    :    .    :    .    :   x.    :  HDSPM_ClockModeMaster - 1: Master, 0: Slave
+ * :    .    :    .    :    .    :    .210 :  HDSPM_LatencyMask - 3 Bit value for latency
+ * :    .    :    .    :    .    :    .    :      0:64, 1:128, 2:256, 3:512,
+ * :    .    :    .    :    .    :    .    :      4:1024, 5:2048, 6:4096, 7:8192
+ * :x   .    :    .    :    .   x:xx  .    :  HDSPM_FrequencyMask
+ * :    .    :    .    :    .    :10  .    :  HDSPM_Frequency1|HDSPM_Frequency0: 1=32K,2=44.1K,3=48K,0=??
+ * :    .    :    .    :    .   x:    .    :  <MADI> HDSPM_DoubleSpeed
+ * :x   .    :    .    :    .    :    .    :  <MADI> HDSPM_QuadSpeed
+ * :    .  3 :    .  10:  2 .    :    .    :  HDSPM_SyncRefMask :
+ * :    .    :    .   x:    .    :    .    :  HDSPM_SyncRef0
+ * :    .    :    .  x :    .    :    .    :  HDSPM_SyncRef1
+ * :    .    :    .    :  x .    :    .    :  <AES32> HDSPM_SyncRef2
+ * :    .  x :    .    :    .    :    .    :  <AES32> HDSPM_SyncRef3
+ * :    .    :    .  10:    .    :    .    :  <MADI> sync ref: 0:WC, 1:Madi, 2:TCO, 3:SyncIn
+ * :    .  3 :    .  10:  2 .    :    .    :  <AES32>  0:WC, 1:AES1 ... 8:AES8, 9: TCO, 10:SyncIn?
+ * :    .  x :    .    :    .    :    .    :  <MADIe> HDSPe_FLOAT_FORMAT
+ * :    .    :    .    : x  .    :    .    :  <MADI> HDSPM_InputSelect0 : 0=optical,1=coax
+ * :    .    :    .    :x   .    :    .    :  <MADI> HDSPM_InputSelect1
+ * :    .    :    .x   :    .    :    .    :  <MADI> HDSPM_clr_tms
+ * :    .    :    .    :    . x  :    .    :  <MADI> HDSPM_TX_64ch
+ * :    .    :    .    :    . x  :    .    :  <AES32> HDSPM_Emphasis
+ * :    .    :    .    :    .x   :    .    :  <MADI> HDSPM_AutoInp
+ * :    .    :    . x  :    .    :    .    :  <MADI> HDSPM_SMUX
+ * :    .    :    .x   :    .    :    .    :  <MADI> HDSPM_clr_tms
+ * :    .    :   x.    :    .    :    .    :  <MADI> HDSPM_taxi_reset
+ * :    .   x:    .    :    .    :    .    :  <MADI> HDSPM_LineOut
+ * :    .   x:    .    :    .    :    .    :  <AES32> ??????????????????
+ * :    .    :   x.    :    .    :    .    :  <AES32> HDSPM_WCK48
+ * :    .    :    .    :    .x   :    .    :  <AES32> HDSPM_Dolby
+ * :    .    : x  .    :    .    :    .    :  HDSPM_Midi0InterruptEnable
+ * :    .    :x   .    :    .    :    .    :  HDSPM_Midi1InterruptEnable
+ * :    .    :  x .    :    .    :    .    :  HDSPM_Midi2InterruptEnable
+ * :    . x  :    .    :    .    :    .    :  <MADI> HDSPM_Midi3InterruptEnable
+ * :    . x  :    .    :    .    :    .    :  <AES32> HDSPM_DS_DoubleWire
+ * :    .x   :    .    :    .    :    .    :  <AES32> HDSPM_QS_DoubleWire
+ * :   x.    :    .    :    .    :    .    :  <AES32> HDSPM_QS_QuadWire
+ * :    .    :    .    :    .  x :    .    :  <AES32> HDSPM_Professional
+ * : x  .    :    .    :    .    :    .    :  HDSPM_wclk_sel
+ * :    .    :    .    :    .    :    .    :
+ * :7654.3210:7654.3210:7654.3210:7654.3210: bit number per byte
+ * :||||.||||:||||.||||:||||.||||:||||.||||:
+ * :3322.2222:2222.1111:1111.1100:0000.0000: bit number
+ * :1098.7654:3210.9876:5432.1098:7654.3210: 0..31
+ * :||||.||||:||||.||||:||||.||||:||||.||||:
+ * :8421.8421:8421.8421:8421.8421:8421.8421:hex digit
+ *
+ *
+ *
+ * AIO / RayDAT only
+ *
+ * ------------ HDSPM_WR_SETTINGS ----------
+ * :3322.2222:2222.1111:1111.1100:0000.0000: bit number per byte
+ * :1098.7654:3210.9876:5432.1098:7654.3210:
+ * :||||.||||:||||.||||:||||.||||:||||.||||: bit number
+ * :7654.3210:7654.3210:7654.3210:7654.3210: 0..31
+ * :||||.||||:||||.||||:||||.||||:||||.||||:
+ * :8421.8421:8421.8421:8421.8421:8421.8421: hex digit
+ * :    .    :    .    :    .    :    .   x: HDSPM_c0Master 1: Master, 0: Slave
+ * :    .    :    .    :    .    :    .  x : HDSPM_c0_SyncRef0
+ * :    .    :    .    :    .    :    . x  : HDSPM_c0_SyncRef1
+ * :    .    :    .    :    .    :    .x   : HDSPM_c0_SyncRef2
+ * :    .    :    .    :    .    :   x.    : HDSPM_c0_SyncRef3
+ * :    .    :    .    :    .    :   3.210 : HDSPM_c0_SyncRefMask:
+ * :    .    :    .    :    .    :    .    :  RayDat: 0:WC, 1:AES, 2:SPDIF, 3..6: ADAT1..4,
+ * :    .    :    .    :    .    :    .    :          9:TCO, 10:SyncIn
+ * :    .    :    .    :    .    :    .    :  AIO: 0:WC, 1:AES, 2: SPDIF, 3: ATAT,
+ * :    .    :    .    :    .    :    .    :          9:TCO, 10:SyncIn
+ * :    .    :    .    :    .    :    .    :
+ * :    .    :    .    :    .    :    .    :
+ * :3322.2222:2222.1111:1111.1100:0000.0000: bit number per byte
+ * :1098.7654:3210.9876:5432.1098:7654.3210:
+ * :||||.||||:||||.||||:||||.||||:||||.||||: bit number
+ * :7654.3210:7654.3210:7654.3210:7654.3210: 0..31
+ * :||||.||||:||||.||||:||||.||||:||||.||||:
+ * :8421.8421:8421.8421:8421.8421:8421.8421: hex digit
+ *
+ */
 #include <linux/init.h>
 #include <linux/delay.h>
 #include <linux/interrupt.h>
@@ -95,7 +186,7 @@ MODULE_SUPPORTED_DEVICE("{{RME HDSPM-MADI}}");
 #define HDSPM_controlRegister	     64
 #define HDSPM_interruptConfirmation  96
 #define HDSPM_control2Reg	     256  /* not in specs ???????? */
-#define HDSPM_freqReg                256  /* for AES32 */
+#define HDSPM_freqReg                256  /* for setting arbitrary clock values (DDS feature) */
 #define HDSPM_midiDataOut0	     352  /* just believe in old code */
 #define HDSPM_midiDataOut1	     356
 #define HDSPM_eeprom_wr		     384  /* for AES32 */
@@ -258,6 +349,25 @@ MODULE_SUPPORTED_DEVICE("{{RME HDSPM-MADI}}");
 
 #define HDSPM_wclk_sel (1<<30)
 
+/* additional control register bits for AIO*/
+#define HDSPM_c0_Wck48				0x20 /* also RayDAT */
+#define HDSPM_c0_Input0				0x1000
+#define HDSPM_c0_Input1				0x2000
+#define HDSPM_c0_Spdif_Opt			0x4000
+#define HDSPM_c0_Pro				0x8000
+#define HDSPM_c0_clr_tms			0x10000
+#define HDSPM_c0_AEB1				0x20000
+#define HDSPM_c0_AEB2				0x40000
+#define HDSPM_c0_LineOut			0x80000
+#define HDSPM_c0_AD_GAIN0			0x100000
+#define HDSPM_c0_AD_GAIN1			0x200000
+#define HDSPM_c0_DA_GAIN0			0x400000
+#define HDSPM_c0_DA_GAIN1			0x800000
+#define HDSPM_c0_PH_GAIN0			0x1000000
+#define HDSPM_c0_PH_GAIN1			0x2000000
+#define HDSPM_c0_Sym6db				0x4000000
+
+
 /* --- bit helper defines */
 #define HDSPM_LatencyMask    (HDSPM_Latency0|HDSPM_Latency1|HDSPM_Latency2)
 #define HDSPM_FrequencyMask  (HDSPM_Frequency0|HDSPM_Frequency1|\
@@ -341,11 +451,11 @@ MODULE_SUPPORTED_DEVICE("{{RME HDSPM-MADI}}");
 #define HDSPM_madiLock           (1<<3)	/* MADI Locked =1, no=0 */
 #define HDSPM_madiSync          (1<<18) /* MADI is in sync */
 
-#define HDSPM_tcoLock    0x00000020 /* Optional TCO locked status FOR HDSPe MADI! */
-#define HDSPM_tcoSync    0x10000000 /* Optional TCO sync status */
+#define HDSPM_tcoLockMadi    0x00000020 /* Optional TCO locked status for HDSPe MADI*/
+#define HDSPM_tcoSync    0x10000000 /* Optional TCO sync status for HDSPe MADI and AES32!*/
 
-#define HDSPM_syncInLock 0x00010000 /* Sync In lock status FOR HDSPe MADI! */
-#define HDSPM_syncInSync 0x00020000 /* Sync In sync status FOR HDSPe MADI! */
+#define HDSPM_syncInLock 0x00010000 /* Sync In lock status for HDSPe MADI! */
+#define HDSPM_syncInSync 0x00020000 /* Sync In sync status for HDSPe MADI! */
 
 #define HDSPM_BufferPositionMask 0x000FFC0 /* Bit 6..15 : h/w buffer pointer */
 			/* since 64byte accurate, last 6 bits are not used */
@@ -363,7 +473,7 @@ MODULE_SUPPORTED_DEVICE("{{RME HDSPM-MADI}}");
 					 * Interrupt
 					 */
 #define HDSPM_tco_detect         0x08000000
-#define HDSPM_tco_lock	         0x20000000
+#define HDSPM_tcoLockAes         0x20000000 /* Optional TCO locked status for HDSPe AES */
 
 #define HDSPM_s2_tco_detect      0x00000040
 #define HDSPM_s2_AEBO_D          0x00000080
@@ -461,7 +571,9 @@ MODULE_SUPPORTED_DEVICE("{{RME HDSPM-MADI}}");
 #define HDSPM_AES32_AUTOSYNC_FROM_AES6 6
 #define HDSPM_AES32_AUTOSYNC_FROM_AES7 7
 #define HDSPM_AES32_AUTOSYNC_FROM_AES8 8
-#define HDSPM_AES32_AUTOSYNC_FROM_NONE 9
+#define HDSPM_AES32_AUTOSYNC_FROM_TCO 9
+#define HDSPM_AES32_AUTOSYNC_FROM_SYNC_IN 10
+#define HDSPM_AES32_AUTOSYNC_FROM_NONE 11
 
 /*  status2 */
 /* HDSPM_LockAES_bit is given by HDSPM_LockAES >> (AES# - 1) */
@@ -537,36 +649,39 @@ MODULE_SUPPORTED_DEVICE("{{RME HDSPM-MADI}}");
 /* names for speed modes */
 static char *hdspm_speed_names[] = { "single", "double", "quad" };
 
-static char *texts_autosync_aes_tco[] = { "Word Clock",
+static const char *const texts_autosync_aes_tco[] = { "Word Clock",
 					  "AES1", "AES2", "AES3", "AES4",
 					  "AES5", "AES6", "AES7", "AES8",
-					  "TCO" };
-static char *texts_autosync_aes[] = { "Word Clock",
+					  "TCO", "Sync In"
+};
+static const char *const texts_autosync_aes[] = { "Word Clock",
 				      "AES1", "AES2", "AES3", "AES4",
-				      "AES5", "AES6", "AES7", "AES8" };
-static char *texts_autosync_madi_tco[] = { "Word Clock",
+				      "AES5", "AES6", "AES7", "AES8",
+				      "Sync In"
+};
+static const char *const texts_autosync_madi_tco[] = { "Word Clock",
 					   "MADI", "TCO", "Sync In" };
-static char *texts_autosync_madi[] = { "Word Clock",
+static const char *const texts_autosync_madi[] = { "Word Clock",
 				       "MADI", "Sync In" };
 
-static char *texts_autosync_raydat_tco[] = {
+static const char *const texts_autosync_raydat_tco[] = {
 	"Word Clock",
 	"ADAT 1", "ADAT 2", "ADAT 3", "ADAT 4",
 	"AES", "SPDIF", "TCO", "Sync In"
 };
-static char *texts_autosync_raydat[] = {
+static const char *const texts_autosync_raydat[] = {
 	"Word Clock",
 	"ADAT 1", "ADAT 2", "ADAT 3", "ADAT 4",
 	"AES", "SPDIF", "Sync In"
 };
-static char *texts_autosync_aio_tco[] = {
+static const char *const texts_autosync_aio_tco[] = {
 	"Word Clock",
 	"ADAT", "AES", "SPDIF", "TCO", "Sync In"
 };
-static char *texts_autosync_aio[] = { "Word Clock",
+static const char *const texts_autosync_aio[] = { "Word Clock",
 				      "ADAT", "AES", "SPDIF", "Sync In" };
 
-static char *texts_freq[] = {
+static const char *const texts_freq[] = {
 	"No Lock",
 	"32 kHz",
 	"44.1 kHz",
@@ -629,7 +744,8 @@ static char *texts_ports_aio_in_ss[] = {
 	"AES.L", "AES.R",
 	"SPDIF.L", "SPDIF.R",
 	"ADAT.1", "ADAT.2", "ADAT.3", "ADAT.4", "ADAT.5", "ADAT.6",
-	"ADAT.7", "ADAT.8"
+	"ADAT.7", "ADAT.8",
+	"AEB.1", "AEB.2", "AEB.3", "AEB.4"
 };
 
 static char *texts_ports_aio_out_ss[] = {
@@ -638,14 +754,16 @@ static char *texts_ports_aio_out_ss[] = {
 	"SPDIF.L", "SPDIF.R",
 	"ADAT.1", "ADAT.2", "ADAT.3", "ADAT.4", "ADAT.5", "ADAT.6",
 	"ADAT.7", "ADAT.8",
-	"Phone.L", "Phone.R"
+	"Phone.L", "Phone.R",
+	"AEB.1", "AEB.2", "AEB.3", "AEB.4"
 };
 
 static char *texts_ports_aio_in_ds[] = {
 	"Analogue.L", "Analogue.R",
 	"AES.L", "AES.R",
 	"SPDIF.L", "SPDIF.R",
-	"ADAT.1", "ADAT.2", "ADAT.3", "ADAT.4"
+	"ADAT.1", "ADAT.2", "ADAT.3", "ADAT.4",
+	"AEB.1", "AEB.2", "AEB.3", "AEB.4"
 };
 
 static char *texts_ports_aio_out_ds[] = {
@@ -653,14 +771,16 @@ static char *texts_ports_aio_out_ds[] = {
 	"AES.L", "AES.R",
 	"SPDIF.L", "SPDIF.R",
 	"ADAT.1", "ADAT.2", "ADAT.3", "ADAT.4",
-	"Phone.L", "Phone.R"
+	"Phone.L", "Phone.R",
+	"AEB.1", "AEB.2", "AEB.3", "AEB.4"
 };
 
 static char *texts_ports_aio_in_qs[] = {
 	"Analogue.L", "Analogue.R",
 	"AES.L", "AES.R",
 	"SPDIF.L", "SPDIF.R",
-	"ADAT.1", "ADAT.2", "ADAT.3", "ADAT.4"
+	"ADAT.1", "ADAT.2", "ADAT.3", "ADAT.4",
+	"AEB.1", "AEB.2", "AEB.3", "AEB.4"
 };
 
 static char *texts_ports_aio_out_qs[] = {
@@ -668,7 +788,8 @@ static char *texts_ports_aio_out_qs[] = {
 	"AES.L", "AES.R",
 	"SPDIF.L", "SPDIF.R",
 	"ADAT.1", "ADAT.2", "ADAT.3", "ADAT.4",
-	"Phone.L", "Phone.R"
+	"Phone.L", "Phone.R",
+	"AEB.1", "AEB.2", "AEB.3", "AEB.4"
 };
 
 static char *texts_ports_aes32[] = {
@@ -745,8 +866,8 @@ static char channel_map_aio_in_ss[HDSPM_MAX_CHANNELS] = {
 	8, 9,			/* aes in, */
 	10, 11,			/* spdif in */
 	12, 13, 14, 15, 16, 17, 18, 19,	/* ADAT in */
-	-1, -1,
-	-1, -1, -1, -1, -1, -1, -1, -1,
+	2, 3, 4, 5,		/* AEB */
+	-1, -1, -1, -1, -1, -1,
 	-1, -1, -1, -1, -1, -1, -1, -1,
 	-1, -1, -1, -1, -1, -1, -1, -1,
 	-1, -1, -1, -1, -1, -1, -1, -1,
@@ -760,7 +881,8 @@ static char channel_map_aio_out_ss[HDSPM_MAX_CHANNELS] = {
 	10, 11,			/* spdif out */
 	12, 13, 14, 15, 16, 17, 18, 19,	/* ADAT out */
 	6, 7,			/* phone out */
-	-1, -1, -1, -1, -1, -1, -1, -1,
+	2, 3, 4, 5,		/* AEB */
+	-1, -1, -1, -1,
 	-1, -1, -1, -1, -1, -1, -1, -1,
 	-1, -1, -1, -1, -1, -1, -1, -1,
 	-1, -1, -1, -1, -1, -1, -1, -1,
@@ -773,7 +895,8 @@ static char channel_map_aio_in_ds[HDSPM_MAX_CHANNELS] = {
 	8, 9,			/* aes in */
 	10, 11,			/* spdif in */
 	12, 14, 16, 18,		/* adat in */
-	-1, -1, -1, -1, -1, -1,
+	2, 3, 4, 5,		/* AEB */
+	-1, -1,
 	-1, -1, -1, -1, -1, -1, -1, -1,
 	-1, -1, -1, -1, -1, -1, -1, -1,
 	-1, -1, -1, -1, -1, -1, -1, -1,
@@ -788,7 +911,7 @@ static char channel_map_aio_out_ds[HDSPM_MAX_CHANNELS] = {
 	10, 11,			/* spdif out */
 	12, 14, 16, 18,		/* adat out */
 	6, 7,			/* phone out */
-	-1, -1, -1, -1,
+	2, 3, 4, 5,		/* AEB */
 	-1, -1, -1, -1, -1, -1, -1, -1,
 	-1, -1, -1, -1, -1, -1, -1, -1,
 	-1, -1, -1, -1, -1, -1, -1, -1,
@@ -802,7 +925,8 @@ static char channel_map_aio_in_qs[HDSPM_MAX_CHANNELS] = {
 	8, 9,			/* aes in */
 	10, 11,			/* spdif in */
 	12, 16,			/* adat in */
-	-1, -1, -1, -1, -1, -1, -1, -1,
+	2, 3, 4, 5,		/* AEB */
+	-1, -1, -1, -1,
 	-1, -1, -1, -1, -1, -1, -1, -1,
 	-1, -1, -1, -1, -1, -1, -1, -1,
 	-1, -1, -1, -1, -1, -1, -1, -1,
@@ -817,7 +941,8 @@ static char channel_map_aio_out_qs[HDSPM_MAX_CHANNELS] = {
 	10, 11,			/* spdif out */
 	12, 16,			/* adat out */
 	6, 7,			/* phone out */
-	-1, -1, -1, -1, -1, -1,
+	2, 3, 4, 5,		/* AEB */
+	-1, -1,
 	-1, -1, -1, -1, -1, -1, -1, -1,
 	-1, -1, -1, -1, -1, -1, -1, -1,
 	-1, -1, -1, -1, -1, -1, -1, -1,
@@ -856,11 +981,11 @@ struct hdspm_midi {
 };
 
 struct hdspm_tco {
-	int input;
-	int framerate;
-	int wordclock;
-	int samplerate;
-	int pull;
+	int input; /* 0: LTC, 1:Video, 2: WC*/
+	int framerate; /* 0=24, 1=25, 2=29.97, 3=29.97d, 4=30, 5=30d */
+	int wordclock; /* 0=1:1, 1=44.1->48, 2=48->44.1 */
+	int samplerate; /* 0=44.1, 1=48, 2= freq from app */
+	int pull; /*   0=0, 1=+0.1%, 2=-0.1%, 3=+4%, 4=-4%*/
 	int term; /* 0 = off, 1 = on */
 };
 
@@ -879,7 +1004,7 @@ struct hdspm {
 
 	u32 control_register;	/* cached value */
 	u32 control2_register;	/* cached value */
-	u32 settings_register;
+	u32 settings_register;  /* cached value for AIO / RayDat (sync reference, master/slave) */
 
 	struct hdspm_midi midi[4];
 	struct tasklet_struct midi_tasklet;
@@ -941,7 +1066,7 @@ struct hdspm {
 
 	struct hdspm_tco *tco;  /* NULL if no TCO detected */
 
-	char **texts_autosync;
+	const char *const *texts_autosync;
 	int texts_autosync_items;
 
 	cycles_t last_interrupt;
@@ -976,12 +1101,24 @@ static inline void snd_hdspm_initialize_midi_flush(struct hdspm *hdspm);
 static inline int hdspm_get_pll_freq(struct hdspm *hdspm);
 static int hdspm_update_simple_mixer_controls(struct hdspm *hdspm);
 static int hdspm_autosync_ref(struct hdspm *hdspm);
+static int hdspm_set_toggle_setting(struct hdspm *hdspm, u32 regmask, int out);
 static int snd_hdspm_set_defaults(struct hdspm *hdspm);
 static int hdspm_system_clock_mode(struct hdspm *hdspm);
 static void hdspm_set_sgbuf(struct hdspm *hdspm,
 			    struct snd_pcm_substream *substream,
 			     unsigned int reg, int channels);
 
+static int hdspm_aes_sync_check(struct hdspm *hdspm, int idx);
+static int hdspm_wc_sync_check(struct hdspm *hdspm);
+static int hdspm_tco_sync_check(struct hdspm *hdspm);
+static int hdspm_sync_in_sync_check(struct hdspm *hdspm);
+
+static int hdspm_get_aes_sample_rate(struct hdspm *hdspm, int index);
+static int hdspm_get_tco_sample_rate(struct hdspm *hdspm);
+static int hdspm_get_wc_sample_rate(struct hdspm *hdspm);
+
+
+
 static inline int HDSPM_bit2freq(int n)
 {
 	static const int bit2freq_tab[] = {
@@ -992,6 +1129,12 @@ static inline int HDSPM_bit2freq(int n)
 	return bit2freq_tab[n];
 }
 
+static bool hdspm_is_raydat_or_aio(struct hdspm *hdspm)
+{
+	return ((AIO == hdspm->io_type) || (RayDAT == hdspm->io_type));
+}
+
+
 /* Write/read to/from HDSPM with Adresses in Bytes
    not words but only 32Bit writes are allowed */
 
@@ -1107,14 +1250,11 @@ static int hdspm_rate_multiplier(struct hdspm *hdspm, int rate)
 		else if (hdspm->control_register &
 				HDSPM_DoubleSpeed)
 			return rate * 2;
-	};
+	}
 	return rate;
 }
 
-static int hdspm_tco_sync_check(struct hdspm *hdspm);
-static int hdspm_sync_in_sync_check(struct hdspm *hdspm);
-
-/* check for external sample rate */
+/* check for external sample rate, returns the sample rate in Hz*/
 static int hdspm_external_sample_rate(struct hdspm *hdspm)
 {
 	unsigned int status, status2, timecode;
@@ -1127,17 +1267,36 @@ static int hdspm_external_sample_rate(struct hdspm *hdspm)
 		timecode = hdspm_read(hdspm, HDSPM_timecodeRegister);
 
 		syncref = hdspm_autosync_ref(hdspm);
+		switch (syncref) {
+		case HDSPM_AES32_AUTOSYNC_FROM_WORD:
+		/* Check WC sync and get sample rate */
+			if (hdspm_wc_sync_check(hdspm))
+				return HDSPM_bit2freq(hdspm_get_wc_sample_rate(hdspm));
+			break;
 
-		if (syncref == HDSPM_AES32_AUTOSYNC_FROM_WORD &&
-				status & HDSPM_AES32_wcLock)
-			return HDSPM_bit2freq((status >> HDSPM_AES32_wcFreq_bit) & 0xF);
+		case HDSPM_AES32_AUTOSYNC_FROM_AES1:
+		case HDSPM_AES32_AUTOSYNC_FROM_AES2:
+		case HDSPM_AES32_AUTOSYNC_FROM_AES3:
+		case HDSPM_AES32_AUTOSYNC_FROM_AES4:
+		case HDSPM_AES32_AUTOSYNC_FROM_AES5:
+		case HDSPM_AES32_AUTOSYNC_FROM_AES6:
+		case HDSPM_AES32_AUTOSYNC_FROM_AES7:
+		case HDSPM_AES32_AUTOSYNC_FROM_AES8:
+		/* Check AES sync and get sample rate */
+			if (hdspm_aes_sync_check(hdspm, syncref - HDSPM_AES32_AUTOSYNC_FROM_AES1))
+				return HDSPM_bit2freq(hdspm_get_aes_sample_rate(hdspm,
+							syncref - HDSPM_AES32_AUTOSYNC_FROM_AES1));
+			break;
 
-		if (syncref >= HDSPM_AES32_AUTOSYNC_FROM_AES1 &&
-				syncref <= HDSPM_AES32_AUTOSYNC_FROM_AES8 &&
-				status2 & (HDSPM_LockAES >>
-				(syncref - HDSPM_AES32_AUTOSYNC_FROM_AES1)))
-			return HDSPM_bit2freq((timecode >> (4*(syncref-HDSPM_AES32_AUTOSYNC_FROM_AES1))) & 0xF);
-		return 0;
+
+		case HDSPM_AES32_AUTOSYNC_FROM_TCO:
+		/* Check TCO sync and get sample rate */
+			if (hdspm_tco_sync_check(hdspm))
+				return HDSPM_bit2freq(hdspm_get_tco_sample_rate(hdspm));
+			break;
+		default:
+			return 0;
+		} /* end switch(syncref) */
 		break;
 
 	case MADIface:
@@ -2129,6 +2288,9 @@ static int hdspm_get_wc_sample_rate(struct hdspm *hdspm)
 		status = hdspm_read(hdspm, HDSPM_RD_STATUS_1);
 		return (status >> 16) & 0xF;
 		break;
+	case AES32:
+		status = hdspm_read(hdspm, HDSPM_statusRegister);
+		return (status >> HDSPM_AES32_wcFreq_bit) & 0xF;
 	default:
 		break;
 	}
@@ -2152,6 +2314,9 @@ static int hdspm_get_tco_sample_rate(struct hdspm *hdspm)
 			status = hdspm_read(hdspm, HDSPM_RD_STATUS_1);
 			return (status >> 20) & 0xF;
 			break;
+		case AES32:
+			status = hdspm_read(hdspm, HDSPM_statusRegister);
+			return (status >> 1) & 0xF;
 		default:
 			break;
 		}
@@ -2183,6 +2348,23 @@ static int hdspm_get_sync_in_sample_rate(struct hdspm *hdspm)
 	return 0;
 }
 
+/**
+ * Returns the AES sample rate class for the given card.
+ **/
+static int hdspm_get_aes_sample_rate(struct hdspm *hdspm, int index)
+{
+	int timecode;
+
+	switch (hdspm->io_type) {
+	case AES32:
+		timecode = hdspm_read(hdspm, HDSPM_timecodeRegister);
+		return (timecode >> (4*index)) & 0xF;
+		break;
+	default:
+		break;
+	}
+	return 0;
+}
 
 /**
  * Returns the sample rate class for input source <idx> for
@@ -2196,15 +2378,23 @@ static int hdspm_get_s1_sample_rate(struct hdspm *hdspm, unsigned int idx)
 }
 
 #define ENUMERATED_CTL_INFO(info, texts) \
-{ \
-	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; \
-	uinfo->count = 1; \
-	uinfo->value.enumerated.items = ARRAY_SIZE(texts); \
-	if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) \
-		uinfo->value.enumerated.item =	uinfo->value.enumerated.items - 1; \
-	strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); \
-}
+	snd_ctl_enum_info(info, 1, ARRAY_SIZE(texts), texts)
+
 
+/* Helper function to query the external sample rate and return the
+ * corresponding enum to be returned to userspace.
+ */
+static int hdspm_external_rate_to_enum(struct hdspm *hdspm)
+{
+	int rate = hdspm_external_sample_rate(hdspm);
+	int i, selected_rate = 0;
+	for (i = 1; i < 10; i++)
+		if (HDSPM_bit2freq(i) == rate) {
+			selected_rate = i;
+			break;
+		}
+	return selected_rate;
+}
 
 
 #define HDSPM_AUTOSYNC_SAMPLE_RATE(xname, xindex) \
@@ -2270,7 +2460,7 @@ static int snd_hdspm_get_autosync_sample_rate(struct snd_kcontrol *kcontrol,
 		default:
 			ucontrol->value.enumerated.item[0] =
 				hdspm_get_s1_sample_rate(hdspm,
-						ucontrol->id.index-1);
+						kcontrol->private_value-1);
 		}
 		break;
 
@@ -2289,28 +2479,24 @@ static int snd_hdspm_get_autosync_sample_rate(struct snd_kcontrol *kcontrol,
 			ucontrol->value.enumerated.item[0] =
 				hdspm_get_sync_in_sample_rate(hdspm);
 			break;
+		case 11: /* External Rate */
+			ucontrol->value.enumerated.item[0] =
+				hdspm_external_rate_to_enum(hdspm);
+			break;
 		default: /* AES1 to AES8 */
 			ucontrol->value.enumerated.item[0] =
-				hdspm_get_s1_sample_rate(hdspm,
-						kcontrol->private_value-1);
+				hdspm_get_aes_sample_rate(hdspm,
+						kcontrol->private_value -
+						HDSPM_AES32_AUTOSYNC_FROM_AES1);
 			break;
 		}
 		break;
 
 	case MADI:
 	case MADIface:
-		{
-			int rate = hdspm_external_sample_rate(hdspm);
-			int i, selected_rate = 0;
-			for (i = 1; i < 10; i++)
-				if (HDSPM_bit2freq(i) == rate) {
-					selected_rate = i;
-					break;
-				}
-			ucontrol->value.enumerated.item[0] = selected_rate;
-		}
+		ucontrol->value.enumerated.item[0] =
+			hdspm_external_rate_to_enum(hdspm);
 		break;
-
 	default:
 		break;
 	}
@@ -2359,33 +2545,17 @@ static int hdspm_system_clock_mode(struct hdspm *hdspm)
  **/
 static void hdspm_set_system_clock_mode(struct hdspm *hdspm, int mode)
 {
-	switch (hdspm->io_type) {
-	case AIO:
-	case RayDAT:
-		if (0 == mode)
-			hdspm->settings_register |= HDSPM_c0Master;
-		else
-			hdspm->settings_register &= ~HDSPM_c0Master;
-
-		hdspm_write(hdspm, HDSPM_WR_SETTINGS, hdspm->settings_register);
-		break;
-
-	default:
-		if (0 == mode)
-			hdspm->control_register |= HDSPM_ClockModeMaster;
-		else
-			hdspm->control_register &= ~HDSPM_ClockModeMaster;
-
-		hdspm_write(hdspm, HDSPM_controlRegister,
-				hdspm->control_register);
-	}
+	hdspm_set_toggle_setting(hdspm,
+			(hdspm_is_raydat_or_aio(hdspm)) ?
+			HDSPM_c0Master : HDSPM_ClockModeMaster,
+			(0 == mode));
 }
 
 
 static int snd_hdspm_info_system_clock_mode(struct snd_kcontrol *kcontrol,
 					    struct snd_ctl_elem_info *uinfo)
 {
-	static char *texts[] = { "Master", "AutoSync" };
+	static const char *const texts[] = { "Master", "AutoSync" };
 	ENUMERATED_CTL_INFO(uinfo, texts);
 	return 0;
 }
@@ -2809,16 +2979,7 @@ static int snd_hdspm_info_pref_sync_ref(struct snd_kcontrol *kcontrol,
 {
 	struct hdspm *hdspm = snd_kcontrol_chip(kcontrol);
 
-	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
-	uinfo->count = 1;
-	uinfo->value.enumerated.items = hdspm->texts_autosync_items;
-
-	if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items)
-		uinfo->value.enumerated.item =
-			uinfo->value.enumerated.items - 1;
-
-	strcpy(uinfo->value.enumerated.name,
-			hdspm->texts_autosync[uinfo->value.enumerated.item]);
+	snd_ctl_enum_info(uinfo, 1, hdspm->texts_autosync_items, hdspm->texts_autosync);
 
 	return 0;
 }
@@ -2873,19 +3034,20 @@ static int snd_hdspm_put_pref_sync_ref(struct snd_kcontrol *kcontrol,
 
 static int hdspm_autosync_ref(struct hdspm *hdspm)
 {
+	/* This looks at the autosync selected sync reference */
 	if (AES32 == hdspm->io_type) {
+
 		unsigned int status = hdspm_read(hdspm, HDSPM_statusRegister);
-		unsigned int syncref =
-			(status >> HDSPM_AES32_syncref_bit) & 0xF;
-		if (syncref == 0)
-			return HDSPM_AES32_AUTOSYNC_FROM_WORD;
-		if (syncref <= 8)
+		unsigned int syncref = (status >> HDSPM_AES32_syncref_bit) & 0xF;
+		if ((syncref >= HDSPM_AES32_AUTOSYNC_FROM_WORD) &&
+				(syncref <= HDSPM_AES32_AUTOSYNC_FROM_SYNC_IN)) {
 			return syncref;
+		}
 		return HDSPM_AES32_AUTOSYNC_FROM_NONE;
+
 	} else if (MADI == hdspm->io_type) {
-		/* This looks at the autosync selected sync reference */
-		unsigned int status2 = hdspm_read(hdspm, HDSPM_statusRegister2);
 
+		unsigned int status2 = hdspm_read(hdspm, HDSPM_statusRegister2);
 		switch (status2 & HDSPM_SelSyncRefMask) {
 		case HDSPM_SelSyncRef_WORD:
 			return HDSPM_AUTOSYNC_FROM_WORD;
@@ -2898,7 +3060,7 @@ static int hdspm_autosync_ref(struct hdspm *hdspm)
 		case HDSPM_SelSyncRef_NVALID:
 			return HDSPM_AUTOSYNC_FROM_NONE;
 		default:
-			return 0;
+			return HDSPM_AUTOSYNC_FROM_NONE;
 		}
 
 	}
@@ -2912,31 +3074,15 @@ static int snd_hdspm_info_autosync_ref(struct snd_kcontrol *kcontrol,
 	struct hdspm *hdspm = snd_kcontrol_chip(kcontrol);
 
 	if (AES32 == hdspm->io_type) {
-		static char *texts[] = { "WordClock", "AES1", "AES2", "AES3",
-			"AES4",	"AES5", "AES6", "AES7", "AES8", "None"};
-
-		uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
-		uinfo->count = 1;
-		uinfo->value.enumerated.items = 10;
-		if (uinfo->value.enumerated.item >=
-		    uinfo->value.enumerated.items)
-			uinfo->value.enumerated.item =
-				uinfo->value.enumerated.items - 1;
-		strcpy(uinfo->value.enumerated.name,
-				texts[uinfo->value.enumerated.item]);
+		static const char *const texts[] = { "WordClock", "AES1", "AES2", "AES3",
+			"AES4",	"AES5", "AES6", "AES7", "AES8", "TCO", "Sync In", "None"};
+
+		ENUMERATED_CTL_INFO(uinfo, texts);
 	} else if (MADI == hdspm->io_type) {
-		static char *texts[] = {"Word Clock", "MADI", "TCO",
+		static const char *const texts[] = {"Word Clock", "MADI", "TCO",
 			"Sync In", "None" };
 
-		uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
-		uinfo->count = 1;
-		uinfo->value.enumerated.items = 5;
-		if (uinfo->value.enumerated.item >=
-				uinfo->value.enumerated.items)
-			uinfo->value.enumerated.item =
-				uinfo->value.enumerated.items - 1;
-		strcpy(uinfo->value.enumerated.name,
-				texts[uinfo->value.enumerated.item]);
+		ENUMERATED_CTL_INFO(uinfo, texts);
 	}
 	return 0;
 }
@@ -2964,7 +3110,7 @@ static int snd_hdspm_get_autosync_ref(struct snd_kcontrol *kcontrol,
 static int snd_hdspm_info_tco_video_input_format(struct snd_kcontrol *kcontrol,
 				       struct snd_ctl_elem_info *uinfo)
 {
-	static char *texts[] = {"No video", "NTSC", "PAL"};
+	static const char *const texts[] = {"No video", "NTSC", "PAL"};
 	ENUMERATED_CTL_INFO(uinfo, texts);
 	return 0;
 }
@@ -3010,7 +3156,7 @@ static int snd_hdspm_get_tco_video_input_format(struct snd_kcontrol *kcontrol,
 static int snd_hdspm_info_tco_ltc_frames(struct snd_kcontrol *kcontrol,
 				       struct snd_ctl_elem_info *uinfo)
 {
-	static char *texts[] = {"No lock", "24 fps", "25 fps", "29.97 fps",
+	static const char *const texts[] = {"No lock", "24 fps", "25 fps", "29.97 fps",
 				"30 fps"};
 	ENUMERATED_CTL_INFO(uinfo, texts);
 	return 0;
@@ -3027,19 +3173,19 @@ static int hdspm_tco_ltc_frames(struct hdspm *hdspm)
 					HDSPM_TCO1_LTC_Format_MSB)) {
 		case 0:
 			/* 24 fps */
-			ret = 1;
+			ret = fps_24;
 			break;
 		case HDSPM_TCO1_LTC_Format_LSB:
 			/* 25 fps */
-			ret = 2;
+			ret = fps_25;
 			break;
 		case HDSPM_TCO1_LTC_Format_MSB:
-			/* 25 fps */
-			ret = 3;
+			/* 29.97 fps */
+			ret = fps_2997;
 			break;
 		default:
 			/* 30 fps */
-			ret = 4;
+			ret = fps_30;
 			break;
 		}
 	}
@@ -3067,16 +3213,35 @@ static int snd_hdspm_get_tco_ltc_frames(struct snd_kcontrol *kcontrol,
 
 static int hdspm_toggle_setting(struct hdspm *hdspm, u32 regmask)
 {
-	return (hdspm->control_register & regmask) ? 1 : 0;
+	u32 reg;
+
+	if (hdspm_is_raydat_or_aio(hdspm))
+		reg = hdspm->settings_register;
+	else
+		reg = hdspm->control_register;
+
+	return (reg & regmask) ? 1 : 0;
 }
 
 static int hdspm_set_toggle_setting(struct hdspm *hdspm, u32 regmask, int out)
 {
+	u32 *reg;
+	u32 target_reg;
+
+	if (hdspm_is_raydat_or_aio(hdspm)) {
+		reg = &(hdspm->settings_register);
+		target_reg = HDSPM_WR_SETTINGS;
+	} else {
+		reg = &(hdspm->control_register);
+		target_reg = HDSPM_controlRegister;
+	}
+
 	if (out)
-		hdspm->control_register |= regmask;
+		*reg |= regmask;
 	else
-		hdspm->control_register &= ~regmask;
-	hdspm_write(hdspm, HDSPM_controlRegister, hdspm->control_register);
+		*reg &= ~regmask;
+
+	hdspm_write(hdspm, target_reg, *reg);
 
 	return 0;
 }
@@ -3141,7 +3306,7 @@ static int hdspm_set_input_select(struct hdspm * hdspm, int out)
 static int snd_hdspm_info_input_select(struct snd_kcontrol *kcontrol,
 				       struct snd_ctl_elem_info *uinfo)
 {
-	static char *texts[] = { "optical", "coaxial" };
+	static const char *const texts[] = { "optical", "coaxial" };
 	ENUMERATED_CTL_INFO(uinfo, texts);
 	return 0;
 }
@@ -3203,7 +3368,7 @@ static int hdspm_set_ds_wire(struct hdspm * hdspm, int ds)
 static int snd_hdspm_info_ds_wire(struct snd_kcontrol *kcontrol,
 				  struct snd_ctl_elem_info *uinfo)
 {
-	static char *texts[] = { "Single", "Double" };
+	static const char *const texts[] = { "Single", "Double" };
 	ENUMERATED_CTL_INFO(uinfo, texts);
 	return 0;
 }
@@ -3276,7 +3441,7 @@ static int hdspm_set_qs_wire(struct hdspm * hdspm, int mode)
 static int snd_hdspm_info_qs_wire(struct snd_kcontrol *kcontrol,
 				       struct snd_ctl_elem_info *uinfo)
 {
-	static char *texts[] = { "Single", "Double", "Quad" };
+	static const char *const texts[] = { "Single", "Double", "Quad" };
 	ENUMERATED_CTL_INFO(uinfo, texts);
 	return 0;
 }
@@ -3313,6 +3478,84 @@ static int snd_hdspm_put_qs_wire(struct snd_kcontrol *kcontrol,
 	return change;
 }
 
+#define HDSPM_CONTROL_TRISTATE(xname, xindex) \
+{	.iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+	.name = xname, \
+	.private_value = xindex, \
+	.info = snd_hdspm_info_tristate, \
+	.get = snd_hdspm_get_tristate, \
+	.put = snd_hdspm_put_tristate \
+}
+
+static int hdspm_tristate(struct hdspm *hdspm, u32 regmask)
+{
+	u32 reg = hdspm->settings_register & (regmask * 3);
+	return reg / regmask;
+}
+
+static int hdspm_set_tristate(struct hdspm *hdspm, int mode, u32 regmask)
+{
+	hdspm->settings_register &= ~(regmask * 3);
+	hdspm->settings_register |= (regmask * mode);
+	hdspm_write(hdspm, HDSPM_WR_SETTINGS, hdspm->settings_register);
+
+	return 0;
+}
+
+static int snd_hdspm_info_tristate(struct snd_kcontrol *kcontrol,
+				       struct snd_ctl_elem_info *uinfo)
+{
+	u32 regmask = kcontrol->private_value;
+
+	static const char *const texts_spdif[] = { "Optical", "Coaxial", "Internal" };
+	static const char *const texts_levels[] = { "Hi Gain", "+4 dBu", "-10 dBV" };
+
+	switch (regmask) {
+	case HDSPM_c0_Input0:
+		ENUMERATED_CTL_INFO(uinfo, texts_spdif);
+		break;
+	default:
+		ENUMERATED_CTL_INFO(uinfo, texts_levels);
+		break;
+	}
+	return 0;
+}
+
+static int snd_hdspm_get_tristate(struct snd_kcontrol *kcontrol,
+				      struct snd_ctl_elem_value *ucontrol)
+{
+	struct hdspm *hdspm = snd_kcontrol_chip(kcontrol);
+	u32 regmask = kcontrol->private_value;
+
+	spin_lock_irq(&hdspm->lock);
+	ucontrol->value.enumerated.item[0] = hdspm_tristate(hdspm, regmask);
+	spin_unlock_irq(&hdspm->lock);
+	return 0;
+}
+
+static int snd_hdspm_put_tristate(struct snd_kcontrol *kcontrol,
+				      struct snd_ctl_elem_value *ucontrol)
+{
+	struct hdspm *hdspm = snd_kcontrol_chip(kcontrol);
+	u32 regmask = kcontrol->private_value;
+	int change;
+	int val;
+
+	if (!snd_hdspm_use_is_exclusive(hdspm))
+		return -EBUSY;
+	val = ucontrol->value.integer.value[0];
+	if (val < 0)
+		val = 0;
+	if (val > 2)
+		val = 2;
+
+	spin_lock_irq(&hdspm->lock);
+	change = val != hdspm_tristate(hdspm, regmask);
+	hdspm_set_tristate(hdspm, val, regmask);
+	spin_unlock_irq(&hdspm->lock);
+	return change;
+}
+
 #define HDSPM_MADI_SPEEDMODE(xname, xindex) \
 {	.iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
 	.name = xname, \
@@ -3352,7 +3595,7 @@ static int hdspm_set_madi_speedmode(struct hdspm *hdspm, int mode)
 static int snd_hdspm_info_madi_speedmode(struct snd_kcontrol *kcontrol,
 				       struct snd_ctl_elem_info *uinfo)
 {
-	static char *texts[] = { "Single", "Double", "Quad" };
+	static const char *const texts[] = { "Single", "Double", "Quad" };
 	ENUMERATED_CTL_INFO(uinfo, texts);
 	return 0;
 }
@@ -3587,7 +3830,7 @@ static int snd_hdspm_put_playback_mixer(struct snd_kcontrol *kcontrol,
 static int snd_hdspm_info_sync_check(struct snd_kcontrol *kcontrol,
 				     struct snd_ctl_elem_info *uinfo)
 {
-	static char *texts[] = { "No Lock", "Lock", "Sync", "N/A" };
+	static const char *const texts[] = { "No Lock", "Lock", "Sync", "N/A" };
 	ENUMERATED_CTL_INFO(uinfo, texts);
 	return 0;
 }
@@ -3595,7 +3838,7 @@ static int snd_hdspm_info_sync_check(struct snd_kcontrol *kcontrol,
 static int snd_hdspm_tco_info_lock_check(struct snd_kcontrol *kcontrol,
 				     struct snd_ctl_elem_info *uinfo)
 {
-	static char *texts[] = { "No Lock", "Lock" };
+	static const char *const texts[] = { "No Lock", "Lock" };
 	ENUMERATED_CTL_INFO(uinfo, texts);
 	return 0;
 }
@@ -3745,9 +3988,18 @@ static int hdspm_tco_sync_check(struct hdspm *hdspm)
 	if (hdspm->tco) {
 		switch (hdspm->io_type) {
 		case MADI:
+			status = hdspm_read(hdspm, HDSPM_statusRegister);
+			if (status & HDSPM_tcoLockMadi) {
+				if (status & HDSPM_tcoSync)
+					return 2;
+				else
+					return 1;
+			}
+			return 0;
+			break;
 		case AES32:
 			status = hdspm_read(hdspm, HDSPM_statusRegister);
-			if (status & HDSPM_tcoLock) {
+			if (status & HDSPM_tcoLockAes) {
 				if (status & HDSPM_tcoSync)
 					return 2;
 				else
@@ -3807,7 +4059,8 @@ static int snd_hdspm_get_sync_check(struct snd_kcontrol *kcontrol,
 		case 5: /* SYNC IN */
 			val = hdspm_sync_in_sync_check(hdspm); break;
 		default:
-			val = hdspm_s1_sync_check(hdspm, ucontrol->id.index-1);
+			val = hdspm_s1_sync_check(hdspm,
+					kcontrol->private_value-1);
 		}
 		break;
 
@@ -3975,7 +4228,8 @@ static void hdspm_tco_write(struct hdspm *hdspm)
 static int snd_hdspm_info_tco_sample_rate(struct snd_kcontrol *kcontrol,
 					  struct snd_ctl_elem_info *uinfo)
 {
-	static char *texts[] = { "44.1 kHz", "48 kHz" };
+	/* TODO freq from app could be supported here, see tco->samplerate */
+	static const char *const texts[] = { "44.1 kHz", "48 kHz" };
 	ENUMERATED_CTL_INFO(uinfo, texts);
 	return 0;
 }
@@ -4021,7 +4275,8 @@ static int snd_hdspm_put_tco_sample_rate(struct snd_kcontrol *kcontrol,
 static int snd_hdspm_info_tco_pull(struct snd_kcontrol *kcontrol,
 				   struct snd_ctl_elem_info *uinfo)
 {
-	static char *texts[] = { "0", "+ 0.1 %", "- 0.1 %", "+ 4 %", "- 4 %" };
+	static const char *const texts[] = { "0", "+ 0.1 %", "- 0.1 %",
+		"+ 4 %", "- 4 %" };
 	ENUMERATED_CTL_INFO(uinfo, texts);
 	return 0;
 }
@@ -4066,7 +4321,7 @@ static int snd_hdspm_put_tco_pull(struct snd_kcontrol *kcontrol,
 static int snd_hdspm_info_tco_wck_conversion(struct snd_kcontrol *kcontrol,
 					     struct snd_ctl_elem_info *uinfo)
 {
-	static char *texts[] = { "1:1", "44.1 -> 48", "48 -> 44.1" };
+	static const char *const texts[] = { "1:1", "44.1 -> 48", "48 -> 44.1" };
 	ENUMERATED_CTL_INFO(uinfo, texts);
 	return 0;
 }
@@ -4112,7 +4367,7 @@ static int snd_hdspm_put_tco_wck_conversion(struct snd_kcontrol *kcontrol,
 static int snd_hdspm_info_tco_frame_rate(struct snd_kcontrol *kcontrol,
 					  struct snd_ctl_elem_info *uinfo)
 {
-	static char *texts[] = { "24 fps", "25 fps", "29.97fps",
+	static const char *const texts[] = { "24 fps", "25 fps", "29.97fps",
 		"29.97 dfps", "30 fps", "30 dfps" };
 	ENUMERATED_CTL_INFO(uinfo, texts);
 	return 0;
@@ -4159,7 +4414,7 @@ static int snd_hdspm_put_tco_frame_rate(struct snd_kcontrol *kcontrol,
 static int snd_hdspm_info_tco_sync_source(struct snd_kcontrol *kcontrol,
 					  struct snd_ctl_elem_info *uinfo)
 {
-	static char *texts[] = { "LTC", "Video", "WCK" };
+	static const char *const texts[] = { "LTC", "Video", "WCK" };
 	ENUMERATED_CTL_INFO(uinfo, texts);
 	return 0;
 }
@@ -4284,7 +4539,6 @@ static struct snd_kcontrol_new snd_hdspm_controls_aio[] = {
 	HDSPM_INTERNAL_CLOCK("Internal Clock", 0),
 	HDSPM_SYSTEM_CLOCK_MODE("System Clock Mode", 0),
 	HDSPM_PREF_SYNC_REF("Preferred Sync Reference", 0),
-	HDSPM_AUTOSYNC_REF("AutoSync Reference", 0),
 	HDSPM_SYSTEM_SAMPLE_RATE("System Sample Rate", 0),
 	HDSPM_AUTOSYNC_SAMPLE_RATE("External Rate", 0),
 	HDSPM_SYNC_CHECK("WC SyncCheck", 0),
@@ -4298,7 +4552,16 @@ static struct snd_kcontrol_new snd_hdspm_controls_aio[] = {
 	HDSPM_AUTOSYNC_SAMPLE_RATE("SPDIF Frequency", 2),
 	HDSPM_AUTOSYNC_SAMPLE_RATE("ADAT Frequency", 3),
 	HDSPM_AUTOSYNC_SAMPLE_RATE("TCO Frequency", 4),
-	HDSPM_AUTOSYNC_SAMPLE_RATE("SYNC IN Frequency", 5)
+	HDSPM_AUTOSYNC_SAMPLE_RATE("SYNC IN Frequency", 5),
+	HDSPM_CONTROL_TRISTATE("S/PDIF Input", HDSPM_c0_Input0),
+	HDSPM_TOGGLE_SETTING("S/PDIF Out Optical", HDSPM_c0_Spdif_Opt),
+	HDSPM_TOGGLE_SETTING("S/PDIF Out Professional", HDSPM_c0_Pro),
+	HDSPM_TOGGLE_SETTING("ADAT internal (AEB/TEB)", HDSPM_c0_AEB1),
+	HDSPM_TOGGLE_SETTING("XLR Breakout Cable", HDSPM_c0_Sym6db),
+	HDSPM_TOGGLE_SETTING("Single Speed WordClock Out", HDSPM_c0_Wck48),
+	HDSPM_CONTROL_TRISTATE("Input Level", HDSPM_c0_AD_GAIN0),
+	HDSPM_CONTROL_TRISTATE("Output Level", HDSPM_c0_DA_GAIN0),
+	HDSPM_CONTROL_TRISTATE("Phones Level", HDSPM_c0_PH_GAIN0)
 
 		/*
 		   HDSPM_INPUT_SELECT("Input Select", 0),
@@ -4335,7 +4598,9 @@ static struct snd_kcontrol_new snd_hdspm_controls_raydat[] = {
 	HDSPM_AUTOSYNC_SAMPLE_RATE("ADAT3 Frequency", 5),
 	HDSPM_AUTOSYNC_SAMPLE_RATE("ADAT4 Frequency", 6),
 	HDSPM_AUTOSYNC_SAMPLE_RATE("TCO Frequency", 7),
-	HDSPM_AUTOSYNC_SAMPLE_RATE("SYNC IN Frequency", 8)
+	HDSPM_AUTOSYNC_SAMPLE_RATE("SYNC IN Frequency", 8),
+	HDSPM_TOGGLE_SETTING("S/PDIF Out Professional", HDSPM_c0_Pro),
+	HDSPM_TOGGLE_SETTING("Single Speed WordClock Out", HDSPM_c0_Wck48)
 };
 
 static struct snd_kcontrol_new snd_hdspm_controls_aes32[] = {
@@ -4345,7 +4610,7 @@ static struct snd_kcontrol_new snd_hdspm_controls_aes32[] = {
 	HDSPM_PREF_SYNC_REF("Preferred Sync Reference", 0),
 	HDSPM_AUTOSYNC_REF("AutoSync Reference", 0),
 	HDSPM_SYSTEM_SAMPLE_RATE("System Sample Rate", 0),
-	HDSPM_AUTOSYNC_SAMPLE_RATE("External Rate", 0),
+	HDSPM_AUTOSYNC_SAMPLE_RATE("External Rate", 11),
 	HDSPM_SYNC_CHECK("WC Sync Check", 0),
 	HDSPM_SYNC_CHECK("AES1 Sync Check", 1),
 	HDSPM_SYNC_CHECK("AES2 Sync Check", 2),
@@ -4501,77 +4766,22 @@ static int snd_hdspm_create_controls(struct snd_card *card,
  ------------------------------------------------------------*/
 
 static void
-snd_hdspm_proc_read_madi(struct snd_info_entry * entry,
-			 struct snd_info_buffer *buffer)
+snd_hdspm_proc_read_tco(struct snd_info_entry *entry,
+					struct snd_info_buffer *buffer)
 {
 	struct hdspm *hdspm = entry->private_data;
-	unsigned int status, status2, control, freq;
-
-	char *pref_sync_ref;
-	char *autosync_ref;
-	char *system_clock_mode;
-	char *insel;
-	int x, x2;
-
-	/* TCO stuff */
+	unsigned int status, control;
 	int a, ltc, frames, seconds, minutes, hours;
 	unsigned int period;
 	u64 freq_const = 0;
 	u32 rate;
 
+	snd_iprintf(buffer, "--- TCO ---\n");
+
 	status = hdspm_read(hdspm, HDSPM_statusRegister);
-	status2 = hdspm_read(hdspm, HDSPM_statusRegister2);
 	control = hdspm->control_register;
-	freq = hdspm_read(hdspm, HDSPM_timecodeRegister);
 
-	snd_iprintf(buffer, "%s (Card #%d) Rev.%x Status2first3bits: %x\n",
-			hdspm->card_name, hdspm->card->number + 1,
-			hdspm->firmware_rev,
-			(status2 & HDSPM_version0) |
-			(status2 & HDSPM_version1) | (status2 &
-				HDSPM_version2));
 
-	snd_iprintf(buffer, "HW Serial: 0x%06x%06x\n",
-			(hdspm_read(hdspm, HDSPM_midiStatusIn1)>>8) & 0xFFFFFF,
-			hdspm->serial);
-
-	snd_iprintf(buffer, "IRQ: %d Registers bus: 0x%lx VM: 0x%lx\n",
-			hdspm->irq, hdspm->port, (unsigned long)hdspm->iobase);
-
-	snd_iprintf(buffer, "--- System ---\n");
-
-	snd_iprintf(buffer,
-		"IRQ Pending: Audio=%d, MIDI0=%d, MIDI1=%d, IRQcount=%d\n",
-		status & HDSPM_audioIRQPending,
-		(status & HDSPM_midi0IRQPending) ? 1 : 0,
-		(status & HDSPM_midi1IRQPending) ? 1 : 0,
-		hdspm->irq_count);
-	snd_iprintf(buffer,
-		"HW pointer: id = %d, rawptr = %d (%d->%d) "
-		"estimated= %ld (bytes)\n",
-		((status & HDSPM_BufferID) ? 1 : 0),
-		(status & HDSPM_BufferPositionMask),
-		(status & HDSPM_BufferPositionMask) %
-		(2 * (int)hdspm->period_bytes),
-		((status & HDSPM_BufferPositionMask) - 64) %
-		(2 * (int)hdspm->period_bytes),
-		(long) hdspm_hw_pointer(hdspm) * 4);
-
-	snd_iprintf(buffer,
-		"MIDI FIFO: Out1=0x%x, Out2=0x%x, In1=0x%x, In2=0x%x \n",
-		hdspm_read(hdspm, HDSPM_midiStatusOut0) & 0xFF,
-		hdspm_read(hdspm, HDSPM_midiStatusOut1) & 0xFF,
-		hdspm_read(hdspm, HDSPM_midiStatusIn0) & 0xFF,
-		hdspm_read(hdspm, HDSPM_midiStatusIn1) & 0xFF);
-	snd_iprintf(buffer,
-		"MIDIoverMADI FIFO: In=0x%x, Out=0x%x \n",
-		hdspm_read(hdspm, HDSPM_midiStatusIn2) & 0xFF,
-		hdspm_read(hdspm, HDSPM_midiStatusOut2) & 0xFF);
-	snd_iprintf(buffer,
-		"Register: ctrl1=0x%x, ctrl2=0x%x, status1=0x%x, "
-		"status2=0x%x\n",
-		hdspm->control_register, hdspm->control2_register,
-		status, status2);
 	if (status & HDSPM_tco_detect) {
 		snd_iprintf(buffer, "TCO module detected.\n");
 		a = hdspm_read(hdspm, HDSPM_RD_TCO+4);
@@ -4665,6 +4875,75 @@ snd_hdspm_proc_read_madi(struct snd_info_entry * entry,
 	} else {
 		snd_iprintf(buffer, "No TCO module detected.\n");
 	}
+}
+
+static void
+snd_hdspm_proc_read_madi(struct snd_info_entry *entry,
+			 struct snd_info_buffer *buffer)
+{
+	struct hdspm *hdspm = entry->private_data;
+	unsigned int status, status2, control, freq;
+
+	char *pref_sync_ref;
+	char *autosync_ref;
+	char *system_clock_mode;
+	char *insel;
+	int x, x2;
+
+	status = hdspm_read(hdspm, HDSPM_statusRegister);
+	status2 = hdspm_read(hdspm, HDSPM_statusRegister2);
+	control = hdspm->control_register;
+	freq = hdspm_read(hdspm, HDSPM_timecodeRegister);
+
+	snd_iprintf(buffer, "%s (Card #%d) Rev.%x Status2first3bits: %x\n",
+			hdspm->card_name, hdspm->card->number + 1,
+			hdspm->firmware_rev,
+			(status2 & HDSPM_version0) |
+			(status2 & HDSPM_version1) | (status2 &
+				HDSPM_version2));
+
+	snd_iprintf(buffer, "HW Serial: 0x%06x%06x\n",
+			(hdspm_read(hdspm, HDSPM_midiStatusIn1)>>8) & 0xFFFFFF,
+			hdspm->serial);
+
+	snd_iprintf(buffer, "IRQ: %d Registers bus: 0x%lx VM: 0x%lx\n",
+			hdspm->irq, hdspm->port, (unsigned long)hdspm->iobase);
+
+	snd_iprintf(buffer, "--- System ---\n");
+
+	snd_iprintf(buffer,
+		"IRQ Pending: Audio=%d, MIDI0=%d, MIDI1=%d, IRQcount=%d\n",
+		status & HDSPM_audioIRQPending,
+		(status & HDSPM_midi0IRQPending) ? 1 : 0,
+		(status & HDSPM_midi1IRQPending) ? 1 : 0,
+		hdspm->irq_count);
+	snd_iprintf(buffer,
+		"HW pointer: id = %d, rawptr = %d (%d->%d) "
+		"estimated= %ld (bytes)\n",
+		((status & HDSPM_BufferID) ? 1 : 0),
+		(status & HDSPM_BufferPositionMask),
+		(status & HDSPM_BufferPositionMask) %
+		(2 * (int)hdspm->period_bytes),
+		((status & HDSPM_BufferPositionMask) - 64) %
+		(2 * (int)hdspm->period_bytes),
+		(long) hdspm_hw_pointer(hdspm) * 4);
+
+	snd_iprintf(buffer,
+		"MIDI FIFO: Out1=0x%x, Out2=0x%x, In1=0x%x, In2=0x%x \n",
+		hdspm_read(hdspm, HDSPM_midiStatusOut0) & 0xFF,
+		hdspm_read(hdspm, HDSPM_midiStatusOut1) & 0xFF,
+		hdspm_read(hdspm, HDSPM_midiStatusIn0) & 0xFF,
+		hdspm_read(hdspm, HDSPM_midiStatusIn1) & 0xFF);
+	snd_iprintf(buffer,
+		"MIDIoverMADI FIFO: In=0x%x, Out=0x%x \n",
+		hdspm_read(hdspm, HDSPM_midiStatusIn2) & 0xFF,
+		hdspm_read(hdspm, HDSPM_midiStatusOut2) & 0xFF);
+	snd_iprintf(buffer,
+		"Register: ctrl1=0x%x, ctrl2=0x%x, status1=0x%x, "
+		"status2=0x%x\n",
+		hdspm->control_register, hdspm->control2_register,
+		status, status2);
+
 
 	snd_iprintf(buffer, "--- Settings ---\n");
 
@@ -4768,6 +5047,9 @@ snd_hdspm_proc_read_madi(struct snd_info_entry * entry,
 		(status & HDSPM_RX_64ch) ? "64 channels" :
 		"56 channels");
 
+	/* call readout function for TCO specific status */
+	snd_hdspm_proc_read_tco(entry, buffer);
+
 	snd_iprintf(buffer, "\n");
 }
 
@@ -4909,11 +5191,18 @@ snd_hdspm_proc_read_aes32(struct snd_info_entry * entry,
 		autosync_ref = "AES7"; break;
 	case HDSPM_AES32_AUTOSYNC_FROM_AES8:
 		autosync_ref = "AES8"; break;
+	case HDSPM_AES32_AUTOSYNC_FROM_TCO:
+		autosync_ref = "TCO"; break;
+	case HDSPM_AES32_AUTOSYNC_FROM_SYNC_IN:
+		autosync_ref = "Sync In"; break;
 	default:
 		autosync_ref = "---"; break;
 	}
 	snd_iprintf(buffer, "AutoSync ref = %s\n", autosync_ref);
 
+	/* call readout function for TCO specific status */
+	snd_hdspm_proc_read_tco(entry, buffer);
+
 	snd_iprintf(buffer, "\n");
 }
 
@@ -5097,7 +5386,7 @@ static int snd_hdspm_set_defaults(struct hdspm * hdspm)
 
 	case AES32:
 		hdspm->control_register =
-			HDSPM_ClockModeMaster |	/* Master Cloack Mode on */
+			HDSPM_ClockModeMaster |	/* Master Clock Mode on */
 			hdspm_encode_latency(7) | /* latency max=8192samples */
 			HDSPM_SyncRef0 |	/* AES1 is syncclock */
 			HDSPM_LineOut |	/* Analog output in */
@@ -5123,9 +5412,8 @@ static int snd_hdspm_set_defaults(struct hdspm * hdspm)
 
 	all_in_all_mixer(hdspm, 0 * UNITY_GAIN);
 
-	if (hdspm->io_type == AIO || hdspm->io_type == RayDAT) {
+	if (hdspm_is_raydat_or_aio(hdspm))
 		hdspm_write(hdspm, HDSPM_WR_SETTINGS, hdspm->settings_register);
-	}
 
 	/* set a default rate so that the channel map is set up. */
 	hdspm_set_rate(hdspm, 48000, 1);
@@ -5371,6 +5659,16 @@ static int snd_hdspm_hw_params(struct snd_pcm_substream *substream,
 	   */
 
 
+	/*  For AES cards, the float format bit is the same as the
+	 *  preferred sync reference. Since we don't want to break
+	 *  sync settings, we have to skip the remaining part of this
+	 *  function.
+	 */
+	if (hdspm->io_type == AES32) {
+		return 0;
+	}
+
+
 	/* Switch to native float format if requested */
 	if (SNDRV_PCM_FORMAT_FLOAT_LE == params_format(params)) {
 		if (!(hdspm->control_register & HDSPe_FLOAT_FORMAT))
@@ -6013,7 +6311,7 @@ static int snd_hdspm_hwdep_ioctl(struct snd_hwdep *hw, struct file *file,
 				ltc.format = fps_2997;
 				break;
 			default:
-				ltc.format = 30;
+				ltc.format = fps_30;
 				break;
 			}
 			if (i & HDSPM_TCO1_set_drop_frame_flag) {
@@ -6479,10 +6777,6 @@ static int snd_hdspm_create(struct snd_card *card,
 		break;
 
 	case AIO:
-		if (0 == (hdspm_read(hdspm, HDSPM_statusRegister2) & HDSPM_s2_AEBI_D)) {
-			snd_printk(KERN_INFO "HDSPM: AEB input board found, but not supported\n");
-		}
-
 		hdspm->ss_in_channels = AIO_IN_SS_CHANNELS;
 		hdspm->ds_in_channels = AIO_IN_DS_CHANNELS;
 		hdspm->qs_in_channels = AIO_IN_QS_CHANNELS;
@@ -6490,6 +6784,20 @@ static int snd_hdspm_create(struct snd_card *card,
 		hdspm->ds_out_channels = AIO_OUT_DS_CHANNELS;
 		hdspm->qs_out_channels = AIO_OUT_QS_CHANNELS;
 
+		if (0 == (hdspm_read(hdspm, HDSPM_statusRegister2) & HDSPM_s2_AEBI_D)) {
+			snd_printk(KERN_INFO "HDSPM: AEB input board found\n");
+			hdspm->ss_in_channels += 4;
+			hdspm->ds_in_channels += 4;
+			hdspm->qs_in_channels += 4;
+		}
+
+		if (0 == (hdspm_read(hdspm, HDSPM_statusRegister2) & HDSPM_s2_AEBO_D)) {
+			snd_printk(KERN_INFO "HDSPM: AEB output board found\n");
+			hdspm->ss_out_channels += 4;
+			hdspm->ds_out_channels += 4;
+			hdspm->qs_out_channels += 4;
+		}
+
 		hdspm->channel_map_out_ss = channel_map_aio_out_ss;
 		hdspm->channel_map_out_ds = channel_map_aio_out_ds;
 		hdspm->channel_map_out_qs = channel_map_aio_out_qs;
@@ -6558,6 +6866,7 @@ static int snd_hdspm_create(struct snd_card *card,
 		break;
 
 	case MADI:
+	case AES32:
 		if (hdspm_read(hdspm, HDSPM_statusRegister) & HDSPM_tco_detect) {
 			hdspm->midiPorts++;
 			hdspm->tco = kzalloc(sizeof(struct hdspm_tco),
@@ -6565,7 +6874,7 @@ static int snd_hdspm_create(struct snd_card *card,
 			if (NULL != hdspm->tco) {
 				hdspm_tco_write(hdspm);
 			}
-			snd_printk(KERN_INFO "HDSPM: MADI TCO module found\n");
+			snd_printk(KERN_INFO "HDSPM: MADI/AES TCO module found\n");
 		} else {
 			hdspm->tco = NULL;
 		}
@@ -6580,10 +6889,12 @@ static int snd_hdspm_create(struct snd_card *card,
 	case AES32:
 		if (hdspm->tco) {
 			hdspm->texts_autosync = texts_autosync_aes_tco;
-			hdspm->texts_autosync_items = 10;
+			hdspm->texts_autosync_items =
+				ARRAY_SIZE(texts_autosync_aes_tco);
 		} else {
 			hdspm->texts_autosync = texts_autosync_aes;
-			hdspm->texts_autosync_items = 9;
+			hdspm->texts_autosync_items =
+				ARRAY_SIZE(texts_autosync_aes);
 		}
 		break;
 

sound/soc/Kconfig

@@ -26,12 +26,9 @@ if SND_SOC
 config SND_SOC_AC97_BUS
 	bool
 
-config SND_SOC_DMAENGINE_PCM
-	bool
-
 config SND_SOC_GENERIC_DMAENGINE_PCM
 	bool
-	select SND_SOC_DMAENGINE_PCM
+	select SND_DMAENGINE_PCM
 
 # All the supported SoCs
 source "sound/soc/atmel/Kconfig"

sound/soc/Makefile

@@ -1,10 +1,6 @@
 snd-soc-core-objs := soc-core.o soc-dapm.o soc-jack.o soc-cache.o soc-utils.o
 snd-soc-core-objs += soc-pcm.o soc-compress.o soc-io.o
 
-ifneq ($(CONFIG_SND_SOC_DMAENGINE_PCM),)
-snd-soc-core-objs += soc-dmaengine-pcm.o
-endif
-
 ifneq ($(CONFIG_SND_SOC_GENERIC_DMAENGINE_PCM),)
 snd-soc-core-objs += soc-generic-dmaengine-pcm.o
 endif

sound/soc/atmel/Kconfig

@@ -13,6 +13,7 @@ config SND_ATMEL_SOC_PDC
 config SND_ATMEL_SOC_DMA
 	tristate
 	depends on SND_ATMEL_SOC
+	select SND_SOC_GENERIC_DMAENGINE_PCM
 
 config SND_ATMEL_SOC_SSC
 	tristate
@@ -32,6 +33,26 @@ config SND_AT91_SOC_SAM9G20_WM8731
 	  Say Y if you want to add support for SoC audio on WM8731-based
 	  AT91sam9g20 evaluation board.
 
+config SND_ATMEL_SOC_WM8904
+	tristate "Atmel ASoC driver for boards using WM8904 codec"
+	depends on ARCH_AT91 && ATMEL_SSC && SND_ATMEL_SOC
+	select SND_ATMEL_SOC_SSC
+	select SND_ATMEL_SOC_DMA
+	select SND_SOC_WM8904
+	help
+	  Say Y if you want to add support for Atmel ASoC driver for boards using
+	  WM8904 codec.
+
+config SND_AT91_SOC_SAM9X5_WM8731
+	tristate "SoC Audio support for WM8731-based at91sam9x5 board"
+	depends on ATMEL_SSC && SND_ATMEL_SOC && SOC_AT91SAM9X5
+	select SND_ATMEL_SOC_SSC
+	select SND_ATMEL_SOC_DMA
+	select SND_SOC_WM8731
+	help
+	  Say Y if you want to add support for audio SoC on an
+	  at91sam9x5 based board that is using WM8731 codec.
+
 config SND_AT91_SOC_AFEB9260
 	tristate "SoC Audio support for AFEB9260 board"
 	depends on ARCH_AT91 && ATMEL_SSC && ARCH_AT91 && MACH_AFEB9260 && SND_ATMEL_SOC

sound/soc/atmel/Makefile

@@ -11,6 +11,10 @@ obj-$(CONFIG_SND_ATMEL_SOC_SSC) += snd-soc-atmel_ssc_dai.o
 
 # AT91 Machine Support
 snd-soc-sam9g20-wm8731-objs := sam9g20_wm8731.o
+snd-atmel-soc-wm8904-objs := atmel_wm8904.o
+snd-soc-sam9x5-wm8731-objs := sam9x5_wm8731.o
 
 obj-$(CONFIG_SND_AT91_SOC_SAM9G20_WM8731) += snd-soc-sam9g20-wm8731.o
+obj-$(CONFIG_SND_ATMEL_SOC_WM8904) += snd-atmel-soc-wm8904.o
+obj-$(CONFIG_SND_AT91_SOC_SAM9X5_WM8731) += snd-soc-sam9x5-wm8731.o
 obj-$(CONFIG_SND_AT91_SOC_AFEB9260) += snd-soc-afeb9260.o

sound/soc/atmel/atmel-pcm-dma.c

@@ -91,138 +91,52 @@ static void atmel_pcm_dma_irq(u32 ssc_sr,
 	}
 }
 
-/*--------------------------------------------------------------------------*\
- * DMAENGINE operations
-\*--------------------------------------------------------------------------*/
-static bool filter(struct dma_chan *chan, void *slave)
-{
-	struct at_dma_slave *sl = slave;
-
-	if (sl->dma_dev == chan->device->dev) {
-		chan->private = sl;
-		return true;
-	} else {
-		return false;
-	}
-}
-
 static int atmel_pcm_configure_dma(struct snd_pcm_substream *substream,
-	struct snd_pcm_hw_params *params, struct atmel_pcm_dma_params *prtd)
+	struct snd_pcm_hw_params *params, struct dma_slave_config *slave_config)
 {
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct atmel_pcm_dma_params *prtd;
 	struct ssc_device *ssc;
-	struct dma_chan *dma_chan;
-	struct dma_slave_config slave_config;
 	int ret;
 
+	prtd = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
 	ssc = prtd->ssc;
 
-	ret = snd_hwparams_to_dma_slave_config(substream, params,
-			&slave_config);
+	ret = snd_hwparams_to_dma_slave_config(substream, params, slave_config);
 	if (ret) {
 		pr_err("atmel-pcm: hwparams to dma slave configure failed\n");
 		return ret;
 	}
 
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
-		slave_config.dst_addr = (dma_addr_t)ssc->phybase + SSC_THR;
-		slave_config.dst_maxburst = 1;
+		slave_config->dst_addr = ssc->phybase + SSC_THR;
+		slave_config->dst_maxburst = 1;
 	} else {
-		slave_config.src_addr = (dma_addr_t)ssc->phybase + SSC_RHR;
-		slave_config.src_maxburst = 1;
-	}
-
-	dma_chan = snd_dmaengine_pcm_get_chan(substream);
-	if (dmaengine_slave_config(dma_chan, &slave_config)) {
-		pr_err("atmel-pcm: failed to configure dma channel\n");
-		ret = -EBUSY;
-		return ret;
-	}
-
-	return 0;
-}
-
-static int atmel_pcm_hw_params(struct snd_pcm_substream *substream,
-	struct snd_pcm_hw_params *params)
-{
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct atmel_pcm_dma_params *prtd;
-	struct ssc_device *ssc;
-	struct at_dma_slave *sdata = NULL;
-	int ret;
-
-	snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
-
-	prtd = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
-	ssc = prtd->ssc;
-	if (ssc->pdev)
-		sdata = ssc->pdev->dev.platform_data;
-
-	ret = snd_dmaengine_pcm_open_request_chan(substream, filter, sdata);
-	if (ret) {
-		pr_err("atmel-pcm: dmaengine pcm open failed\n");
-		return -EINVAL;
-	}
-
-	ret = atmel_pcm_configure_dma(substream, params, prtd);
-	if (ret) {
-		pr_err("atmel-pcm: failed to configure dmai\n");
-		goto err;
+		slave_config->src_addr = ssc->phybase + SSC_RHR;
+		slave_config->src_maxburst = 1;
 	}
 
 	prtd->dma_intr_handler = atmel_pcm_dma_irq;
 
 	return 0;
-err:
-	snd_dmaengine_pcm_close_release_chan(substream);
-	return ret;
 }
 
-static int atmel_pcm_dma_prepare(struct snd_pcm_substream *substream)
-{
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct atmel_pcm_dma_params *prtd;
-
-	prtd = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
-
-	ssc_writex(prtd->ssc->regs, SSC_IER, prtd->mask->ssc_error);
-	ssc_writex(prtd->ssc->regs, SSC_CR, prtd->mask->ssc_enable);
-
-	return 0;
-}
-
-static int atmel_pcm_open(struct snd_pcm_substream *substream)
-{
-	snd_soc_set_runtime_hwparams(substream, &atmel_pcm_dma_hardware);
-
-	return 0;
-}
-
-static struct snd_pcm_ops atmel_pcm_ops = {
-	.open		= atmel_pcm_open,
-	.close		= snd_dmaengine_pcm_close_release_chan,
-	.ioctl		= snd_pcm_lib_ioctl,
-	.hw_params	= atmel_pcm_hw_params,
-	.prepare	= atmel_pcm_dma_prepare,
-	.trigger	= snd_dmaengine_pcm_trigger,
-	.pointer	= snd_dmaengine_pcm_pointer_no_residue,
-	.mmap		= atmel_pcm_mmap,
-};
-
-static struct snd_soc_platform_driver atmel_soc_platform = {
-	.ops		= &atmel_pcm_ops,
-	.pcm_new	= atmel_pcm_new,
-	.pcm_free	= atmel_pcm_free,
+static const struct snd_dmaengine_pcm_config atmel_dmaengine_pcm_config = {
+	.prepare_slave_config = atmel_pcm_configure_dma,
+	.pcm_hardware = &atmel_pcm_dma_hardware,
+	.prealloc_buffer_size = ATMEL_SSC_DMABUF_SIZE,
 };
 
 int atmel_pcm_dma_platform_register(struct device *dev)
 {
-	return snd_soc_register_platform(dev, &atmel_soc_platform);
+	return snd_dmaengine_pcm_register(dev, &atmel_dmaengine_pcm_config,
+			SND_DMAENGINE_PCM_FLAG_NO_RESIDUE);
 }
 EXPORT_SYMBOL(atmel_pcm_dma_platform_register);
 
 void atmel_pcm_dma_platform_unregister(struct device *dev)
 {
-	snd_soc_unregister_platform(dev);
+	snd_dmaengine_pcm_unregister(dev);
 }
 EXPORT_SYMBOL(atmel_pcm_dma_platform_unregister);
 

sound/soc/atmel/atmel_ssc_dai.c

@@ -73,6 +73,7 @@ static struct atmel_ssc_mask ssc_tx_mask = {
 	.ssc_disable	= SSC_BIT(CR_TXDIS),
 	.ssc_endx	= SSC_BIT(SR_ENDTX),
 	.ssc_endbuf	= SSC_BIT(SR_TXBUFE),
+	.ssc_error	= SSC_BIT(SR_OVRUN),
 	.pdc_enable	= ATMEL_PDC_TXTEN,
 	.pdc_disable	= ATMEL_PDC_TXTDIS,
 };
@@ -82,6 +83,7 @@ static struct atmel_ssc_mask ssc_rx_mask = {
 	.ssc_disable	= SSC_BIT(CR_RXDIS),
 	.ssc_endx	= SSC_BIT(SR_ENDRX),
 	.ssc_endbuf	= SSC_BIT(SR_RXBUFF),
+	.ssc_error	= SSC_BIT(SR_OVRUN),
 	.pdc_enable	= ATMEL_PDC_RXTEN,
 	.pdc_disable	= ATMEL_PDC_RXTDIS,
 };
@@ -196,15 +198,27 @@ static int atmel_ssc_startup(struct snd_pcm_substream *substream,
 			     struct snd_soc_dai *dai)
 {
 	struct atmel_ssc_info *ssc_p = &ssc_info[dai->id];
-	int dir_mask;
+	struct atmel_pcm_dma_params *dma_params;
+	int dir, dir_mask;
 
 	pr_debug("atmel_ssc_startup: SSC_SR=0x%u\n",
 		ssc_readl(ssc_p->ssc->regs, SR));
 
-	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		dir = 0;
 		dir_mask = SSC_DIR_MASK_PLAYBACK;
-	else
+	} else {
+		dir = 1;
 		dir_mask = SSC_DIR_MASK_CAPTURE;
+	}
+
+	dma_params = &ssc_dma_params[dai->id][dir];
+	dma_params->ssc = ssc_p->ssc;
+	dma_params->substream = substream;
+
+	ssc_p->dma_params[dir] = dma_params;
+
+	snd_soc_dai_set_dma_data(dai, substream, dma_params);
 
 	spin_lock_irq(&ssc_p->lock);
 	if (ssc_p->dir_mask & dir_mask) {
@@ -325,7 +339,6 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream,
 	struct snd_pcm_hw_params *params,
 	struct snd_soc_dai *dai)
 {
-	struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream);
 	int id = dai->id;
 	struct atmel_ssc_info *ssc_p = &ssc_info[id];
 	struct atmel_pcm_dma_params *dma_params;
@@ -344,19 +357,7 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream,
 	else
 		dir = 1;
 
-	dma_params = &ssc_dma_params[id][dir];
-	dma_params->ssc = ssc_p->ssc;
-	dma_params->substream = substream;
-
-	ssc_p->dma_params[dir] = dma_params;
-
-	/*
-	 * The snd_soc_pcm_stream->dma_data field is only used to communicate
-	 * the appropriate DMA parameters to the pcm driver hw_params()
-	 * function.  It should not be used for other purposes
-	 * as it is common to all substreams.
-	 */
-	snd_soc_dai_set_dma_data(rtd->cpu_dai, substream, dma_params);
+	dma_params = ssc_p->dma_params[dir];
 
 	channels = params_channels(params);
 
@@ -648,6 +649,7 @@ static int atmel_ssc_prepare(struct snd_pcm_substream *substream,
 	dma_params = ssc_p->dma_params[dir];
 
 	ssc_writel(ssc_p->ssc->regs, CR, dma_params->mask->ssc_enable);
+	ssc_writel(ssc_p->ssc->regs, IER, dma_params->mask->ssc_error);
 
 	pr_debug("%s enabled SSC_SR=0x%08x\n",
 			dir ? "receive" : "transmit",

sound/soc/atmel/atmel_wm8904.c

@@ -0,0 +1,254 @@
+/*
+ * atmel_wm8904 - Atmel ASoC driver for boards with WM8904 codec.
+ *
+ * Copyright (C) 2012 Atmel
+ *
+ * Author: Bo Shen <voice.shen@atmel.com>
+ *
+ * GPLv2 or later
+ */
+
+#include <linux/clk.h>
+#include <linux/module.h>
+#include <linux/of.h>
+#include <linux/of_device.h>
+#include <linux/pinctrl/consumer.h>
+
+#include <sound/soc.h>
+
+#include "../codecs/wm8904.h"
+#include "atmel_ssc_dai.h"
+
+#define MCLK_RATE 32768
+
+static struct clk *mclk;
+
+static const struct snd_soc_dapm_widget atmel_asoc_wm8904_dapm_widgets[] = {
+	SND_SOC_DAPM_HP("Headphone Jack", NULL),
+	SND_SOC_DAPM_MIC("Mic", NULL),
+	SND_SOC_DAPM_LINE("Line In Jack", NULL),
+};
+
+static int atmel_asoc_wm8904_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;
+	int ret;
+
+	ret = snd_soc_dai_set_pll(codec_dai, WM8904_FLL_MCLK, WM8904_FLL_MCLK,
+		32768, params_rate(params) * 256);
+	if (ret < 0) {
+		pr_err("%s - failed to set wm8904 codec PLL.", __func__);
+		return ret;
+	}
+
+	/*
+	 * As here wm8904 use FLL output as its system clock
+	 * so calling set_sysclk won't care freq parameter
+	 * then we pass 0
+	 */
+	ret = snd_soc_dai_set_sysclk(codec_dai, WM8904_CLK_FLL,
+			0, SND_SOC_CLOCK_IN);
+	if (ret < 0) {
+		pr_err("%s -failed to set wm8904 SYSCLK\n", __func__);
+		return ret;
+	}
+
+	return 0;
+}
+
+static struct snd_soc_ops atmel_asoc_wm8904_ops = {
+	.hw_params = atmel_asoc_wm8904_hw_params,
+};
+
+static int atmel_set_bias_level(struct snd_soc_card *card,
+		struct snd_soc_dapm_context *dapm,
+		enum snd_soc_bias_level level)
+{
+	if (dapm->bias_level == SND_SOC_BIAS_STANDBY) {
+		switch (level) {
+		case SND_SOC_BIAS_PREPARE:
+			clk_prepare_enable(mclk);
+			break;
+		case SND_SOC_BIAS_OFF:
+			clk_disable_unprepare(mclk);
+			break;
+		default:
+			break;
+		}
+	}
+
+	return 0;
+};
+
+static struct snd_soc_dai_link atmel_asoc_wm8904_dailink = {
+	.name = "WM8904",
+	.stream_name = "WM8904 PCM",
+	.codec_dai_name = "wm8904-hifi",
+	.dai_fmt = SND_SOC_DAIFMT_I2S
+		| SND_SOC_DAIFMT_NB_NF
+		| SND_SOC_DAIFMT_CBM_CFM,
+	.ops = &atmel_asoc_wm8904_ops,
+};
+
+static struct snd_soc_card atmel_asoc_wm8904_card = {
+	.name = "atmel_asoc_wm8904",
+	.owner = THIS_MODULE,
+	.set_bias_level = atmel_set_bias_level,
+	.dai_link = &atmel_asoc_wm8904_dailink,
+	.num_links = 1,
+	.dapm_widgets = atmel_asoc_wm8904_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(atmel_asoc_wm8904_dapm_widgets),
+	.fully_routed = true,
+};
+
+static int atmel_asoc_wm8904_dt_init(struct platform_device *pdev)
+{
+	struct device_node *np = pdev->dev.of_node;
+	struct device_node *codec_np, *cpu_np;
+	struct snd_soc_card *card = &atmel_asoc_wm8904_card;
+	struct snd_soc_dai_link *dailink = &atmel_asoc_wm8904_dailink;
+	int ret;
+
+	if (!np) {
+		dev_err(&pdev->dev, "only device tree supported\n");
+		return -EINVAL;
+	}
+
+	ret = snd_soc_of_parse_card_name(card, "atmel,model");
+	if (ret) {
+		dev_err(&pdev->dev, "failed to parse card name\n");
+		return ret;
+	}
+
+	ret = snd_soc_of_parse_audio_routing(card, "atmel,audio-routing");
+	if (ret) {
+		dev_err(&pdev->dev, "failed to parse audio routing\n");
+		return ret;
+	}
+
+	cpu_np = of_parse_phandle(np, "atmel,ssc-controller", 0);
+	if (!cpu_np) {
+		dev_err(&pdev->dev, "failed to get dai and pcm info\n");
+		ret = -EINVAL;
+		return ret;
+	}
+	dailink->cpu_of_node = cpu_np;
+	dailink->platform_of_node = cpu_np;
+	of_node_put(cpu_np);
+
+	codec_np = of_parse_phandle(np, "atmel,audio-codec", 0);
+	if (!codec_np) {
+		dev_err(&pdev->dev, "failed to get codec info\n");
+		ret = -EINVAL;
+		return ret;
+	}
+	dailink->codec_of_node = codec_np;
+	of_node_put(codec_np);
+
+	return 0;
+}
+
+static int atmel_asoc_wm8904_probe(struct platform_device *pdev)
+{
+	struct snd_soc_card *card = &atmel_asoc_wm8904_card;
+	struct snd_soc_dai_link *dailink = &atmel_asoc_wm8904_dailink;
+	struct clk *clk_src;
+	struct pinctrl *pinctrl;
+	int id, ret;
+
+	pinctrl = devm_pinctrl_get_select_default(&pdev->dev);
+	if (IS_ERR(pinctrl)) {
+		dev_err(&pdev->dev, "failed to request pinctrl\n");
+		return PTR_ERR(pinctrl);
+	}
+
+	card->dev = &pdev->dev;
+	ret = atmel_asoc_wm8904_dt_init(pdev);
+	if (ret) {
+		dev_err(&pdev->dev, "failed to init dt info\n");
+		return ret;
+	}
+
+	id = of_alias_get_id((struct device_node *)dailink->cpu_of_node, "ssc");
+	ret = atmel_ssc_set_audio(id);
+	if (ret != 0) {
+		dev_err(&pdev->dev, "failed to set SSC %d for audio\n", id);
+		return ret;
+	}
+
+	mclk = clk_get(NULL, "pck0");
+	if (IS_ERR(mclk)) {
+		dev_err(&pdev->dev, "failed to get pck0\n");
+		ret = PTR_ERR(mclk);
+		goto err_set_audio;
+	}
+
+	clk_src = clk_get(NULL, "clk32k");
+	if (IS_ERR(clk_src)) {
+		dev_err(&pdev->dev, "failed to get clk32k\n");
+		ret = PTR_ERR(clk_src);
+		goto err_set_audio;
+	}
+
+	ret = clk_set_parent(mclk, clk_src);
+	clk_put(clk_src);
+	if (ret != 0) {
+		dev_err(&pdev->dev, "failed to set MCLK parent\n");
+		goto err_set_audio;
+	}
+
+	dev_info(&pdev->dev, "setting pck0 to %dHz\n", MCLK_RATE);
+	clk_set_rate(mclk, MCLK_RATE);
+
+	ret = snd_soc_register_card(card);
+	if (ret) {
+		dev_err(&pdev->dev, "snd_soc_register_card failed\n");
+		goto err_set_audio;
+	}
+
+	return 0;
+
+err_set_audio:
+	atmel_ssc_put_audio(id);
+	return ret;
+}
+
+static int atmel_asoc_wm8904_remove(struct platform_device *pdev)
+{
+	struct snd_soc_card *card = platform_get_drvdata(pdev);
+	struct snd_soc_dai_link *dailink = &atmel_asoc_wm8904_dailink;
+	int id;
+
+	id = of_alias_get_id((struct device_node *)dailink->cpu_of_node, "ssc");
+
+	snd_soc_unregister_card(card);
+	atmel_ssc_put_audio(id);
+
+	return 0;
+}
+
+#ifdef CONFIG_OF
+static const struct of_device_id atmel_asoc_wm8904_dt_ids[] = {
+	{ .compatible = "atmel,asoc-wm8904", },
+	{ }
+};
+#endif
+
+static struct platform_driver atmel_asoc_wm8904_driver = {
+	.driver = {
+		.name = "atmel-wm8904-audio",
+		.owner = THIS_MODULE,
+		.of_match_table = of_match_ptr(atmel_asoc_wm8904_dt_ids),
+	},
+	.probe = atmel_asoc_wm8904_probe,
+	.remove = atmel_asoc_wm8904_remove,
+};
+
+module_platform_driver(atmel_asoc_wm8904_driver);
+
+/* Module information */
+MODULE_AUTHOR("Bo Shen <voice.shen@atmel.com>");
+MODULE_DESCRIPTION("ALSA SoC machine driver for Atmel EK with WM8904 codec");
+MODULE_LICENSE("GPL");

sound/soc/atmel/sam9x5_wm8731.c

@@ -0,0 +1,208 @@
+/*
+ * sam9x5_wm8731   --	SoC audio for AT91SAM9X5-based boards
+ *			that are using WM8731 as codec.
+ *
+ *  Copyright (C) 2011 Atmel,
+ *		  Nicolas Ferre <nicolas.ferre@atmel.com>
+ *
+ *  Copyright (C) 2013 Paratronic,
+ *		  Richard Genoud <richard.genoud@gmail.com>
+ *
+ * Based on sam9g20_wm8731.c by:
+ * Sedji Gaouaou <sedji.gaouaou@atmel.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/of.h>
+#include <linux/export.h>
+#include <linux/module.h>
+#include <linux/mod_devicetable.h>
+#include <linux/platform_device.h>
+#include <linux/device.h>
+
+#include <sound/soc.h>
+#include <sound/soc-dai.h>
+#include <sound/soc-dapm.h>
+
+#include "../codecs/wm8731.h"
+#include "atmel_ssc_dai.h"
+
+
+#define MCLK_RATE 12288000
+
+#define DRV_NAME "sam9x5-snd-wm8731"
+
+struct sam9x5_drvdata {
+	int ssc_id;
+};
+
+/*
+ * Logic for a wm8731 as connected on a at91sam9x5ek based board.
+ */
+static int sam9x5_wm8731_init(struct snd_soc_pcm_runtime *rtd)
+{
+	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct device *dev = rtd->dev;
+	int ret;
+
+	dev_dbg(dev, "ASoC: %s called\n", __func__);
+
+	/* set the codec system clock for DAC and ADC */
+	ret = snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK_XTAL,
+				     MCLK_RATE, SND_SOC_CLOCK_IN);
+	if (ret < 0) {
+		dev_err(dev, "ASoC: Failed to set WM8731 SYSCLK: %d\n", ret);
+		return ret;
+	}
+
+	return 0;
+}
+
+/*
+ * Audio paths on at91sam9x5ek board:
+ *
+ *  |A| ------------> |      | ---R----> Headphone Jack
+ *  |T| <----\        |  WM  | ---L--/
+ *  |9| ---> CLK <--> | 8731 | <--R----- Line In Jack
+ *  |1| <------------ |      | <--L--/
+ */
+static const struct snd_soc_dapm_widget sam9x5_dapm_widgets[] = {
+	SND_SOC_DAPM_HP("Headphone Jack", NULL),
+	SND_SOC_DAPM_LINE("Line In Jack", NULL),
+};
+
+static int sam9x5_wm8731_driver_probe(struct platform_device *pdev)
+{
+	struct device_node *np = pdev->dev.of_node;
+	struct device_node *codec_np, *cpu_np;
+	struct snd_soc_card *card;
+	struct snd_soc_dai_link *dai;
+	struct sam9x5_drvdata *priv;
+	int ret;
+
+	if (!np) {
+		dev_err(&pdev->dev, "No device node supplied\n");
+		return -EINVAL;
+	}
+
+	card = devm_kzalloc(&pdev->dev, sizeof(*card), GFP_KERNEL);
+	priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL);
+	dai = devm_kzalloc(&pdev->dev, sizeof(*dai), GFP_KERNEL);
+	if (!dai || !card || !priv) {
+		ret = -ENOMEM;
+		goto out;
+	}
+
+	card->dev = &pdev->dev;
+	card->owner = THIS_MODULE;
+	card->dai_link = dai;
+	card->num_links = 1;
+	card->dapm_widgets = sam9x5_dapm_widgets;
+	card->num_dapm_widgets = ARRAY_SIZE(sam9x5_dapm_widgets);
+	dai->name = "WM8731";
+	dai->stream_name = "WM8731 PCM";
+	dai->codec_dai_name = "wm8731-hifi";
+	dai->init = sam9x5_wm8731_init;
+	dai->dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
+		| SND_SOC_DAIFMT_CBM_CFM;
+
+	ret = snd_soc_of_parse_card_name(card, "atmel,model");
+	if (ret) {
+		dev_err(&pdev->dev, "atmel,model node missing\n");
+		goto out;
+	}
+
+	ret = snd_soc_of_parse_audio_routing(card, "atmel,audio-routing");
+	if (ret) {
+		dev_err(&pdev->dev, "atmel,audio-routing node missing\n");
+		goto out;
+	}
+
+	codec_np = of_parse_phandle(np, "atmel,audio-codec", 0);
+	if (!codec_np) {
+		dev_err(&pdev->dev, "atmel,audio-codec node missing\n");
+		ret = -EINVAL;
+		goto out;
+	}
+
+	dai->codec_of_node = codec_np;
+
+	cpu_np = of_parse_phandle(np, "atmel,ssc-controller", 0);
+	if (!cpu_np) {
+		dev_err(&pdev->dev, "atmel,ssc-controller node missing\n");
+		ret = -EINVAL;
+		goto out;
+	}
+	dai->cpu_of_node = cpu_np;
+	dai->platform_of_node = cpu_np;
+
+	priv->ssc_id = of_alias_get_id(cpu_np, "ssc");
+
+	ret = atmel_ssc_set_audio(priv->ssc_id);
+	if (ret != 0) {
+		dev_err(&pdev->dev,
+			"ASoC: Failed to set SSC %d for audio: %d\n",
+			ret, priv->ssc_id);
+		goto out;
+	}
+
+	of_node_put(codec_np);
+	of_node_put(cpu_np);
+
+	platform_set_drvdata(pdev, card);
+
+	ret = snd_soc_register_card(card);
+	if (ret) {
+		dev_err(&pdev->dev,
+			"ASoC: Platform device allocation failed\n");
+		goto out_put_audio;
+	}
+
+	dev_dbg(&pdev->dev, "ASoC: %s ok\n", __func__);
+
+	return ret;
+
+out_put_audio:
+	atmel_ssc_put_audio(priv->ssc_id);
+out:
+	return ret;
+}
+
+static int sam9x5_wm8731_driver_remove(struct platform_device *pdev)
+{
+	struct snd_soc_card *card = platform_get_drvdata(pdev);
+	struct sam9x5_drvdata *priv = card->drvdata;
+
+	snd_soc_unregister_card(card);
+	atmel_ssc_put_audio(priv->ssc_id);
+
+	return 0;
+}
+
+static const struct of_device_id sam9x5_wm8731_of_match[] = {
+	{ .compatible = "atmel,sam9x5-wm8731-audio", },
+	{},
+};
+MODULE_DEVICE_TABLE(of, sam9x5_wm8731_of_match);
+
+static struct platform_driver sam9x5_wm8731_driver = {
+	.driver = {
+		.name = DRV_NAME,
+		.owner = THIS_MODULE,
+		.of_match_table = of_match_ptr(sam9x5_wm8731_of_match),
+	},
+	.probe = sam9x5_wm8731_driver_probe,
+	.remove = sam9x5_wm8731_driver_remove,
+};
+module_platform_driver(sam9x5_wm8731_driver);
+
+/* Module information */
+MODULE_AUTHOR("Nicolas Ferre <nicolas.ferre@atmel.com>");
+MODULE_AUTHOR("Richard Genoud <richard.genoud@gmail.com>");
+MODULE_DESCRIPTION("ALSA SoC machine driver for AT91SAM9x5 - WM8731");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:" DRV_NAME);

sound/soc/au1x/db1200.c

@@ -73,12 +73,14 @@ static struct snd_soc_dai_link db1300_ac97_dai = {
 
 static struct snd_soc_card db1300_ac97_machine = {
 	.name		= "DB1300_AC97",
+	.owner		= THIS_MODULE,
 	.dai_link	= &db1300_ac97_dai,
 	.num_links	= 1,
 };
 
 static struct snd_soc_card db1550_ac97_machine = {
 	.name		= "DB1550_AC97",
+	.owner		= THIS_MODULE,
 	.dai_link	= &db1200_ac97_dai,
 	.num_links	= 1,
 };
@@ -145,6 +147,7 @@ static struct snd_soc_dai_link db1300_i2s_dai = {
 
 static struct snd_soc_card db1300_i2s_machine = {
 	.name		= "DB1300_I2S",
+	.owner		= THIS_MODULE,
 	.dai_link	= &db1300_i2s_dai,
 	.num_links	= 1,
 };
@@ -161,6 +164,7 @@ static struct snd_soc_dai_link db1550_i2s_dai = {
 
 static struct snd_soc_card db1550_i2s_machine = {
 	.name		= "DB1550_I2S",
+	.owner		= THIS_MODULE,
 	.dai_link	= &db1550_i2s_dai,
 	.num_links	= 1,
 };

sound/soc/au1x/psc-ac97.c

@@ -379,9 +379,6 @@ static int au1xpsc_ac97_drvprobe(struct platform_device *pdev)
 	mutex_init(&wd->lock);
 
 	iores = platform_get_resource(pdev, IORESOURCE_MEM, 0);
-	if (!iores)
-		return -ENODEV;
-
 	wd->mmio = devm_ioremap_resource(&pdev->dev, iores);
 	if (IS_ERR(wd->mmio))
 		return PTR_ERR(wd->mmio);

sound/soc/blackfin/bf5xx-ac97.h

@@ -9,7 +9,6 @@
 #ifndef _BF5XX_AC97_H
 #define _BF5XX_AC97_H
 
-extern struct snd_ac97 *ac97;
 /* Frame format in memory, only support stereo currently */
 struct ac97_frame {
 	u16 ac97_tag;		/* slot 0 */

sound/soc/cirrus/ep93xx-ac97.c

@@ -363,9 +363,6 @@ static int ep93xx_ac97_probe(struct platform_device *pdev)
 		return -ENOMEM;
 
 	res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
-	if (!res)
-		return -ENODEV;
-
 	info->regs = devm_ioremap_resource(&pdev->dev, res);
 	if (IS_ERR(info->regs))
 		return PTR_ERR(info->regs);

sound/soc/cirrus/ep93xx-i2s.c

@@ -376,9 +376,6 @@ static int ep93xx_i2s_probe(struct platform_device *pdev)
 		return -ENOMEM;
 
 	res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
-	if (!res)
-		return -ENODEV;
-
 	info->regs = devm_ioremap_resource(&pdev->dev, res);
 	if (IS_ERR(info->regs))
 		return PTR_ERR(info->regs);
@@ -411,7 +408,6 @@ 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);
@@ -426,7 +422,6 @@ static int ep93xx_i2s_remove(struct platform_device *pdev)
 	struct ep93xx_i2s_info *info = dev_get_drvdata(&pdev->dev);
 
 	snd_soc_unregister_component(&pdev->dev);
-	dev_set_drvdata(&pdev->dev, NULL);
 	clk_put(info->lrclk);
 	clk_put(info->sclk);
 	clk_put(info->mclk);

sound/soc/codecs/Kconfig

@@ -10,6 +10,7 @@ config SND_SOC_I2C_AND_SPI
 
 config SND_SOC_ALL_CODECS
 	tristate "Build all ASoC CODEC drivers"
+	depends on COMPILE_TEST
 	select SND_SOC_88PM860X if MFD_88PM860X
 	select SND_SOC_L3
 	select SND_SOC_AB8500_CODEC if ABX500_CORE
@@ -20,6 +21,7 @@ config SND_SOC_ALL_CODECS
 	select SND_SOC_AD73311
 	select SND_SOC_ADAU1373 if I2C
 	select SND_SOC_ADAV80X if SND_SOC_I2C_AND_SPI
+	select SND_SOC_ADAU1701 if I2C
 	select SND_SOC_ADS117X
 	select SND_SOC_AK4104 if SPI_MASTER
 	select SND_SOC_AK4535 if I2C
@@ -54,6 +56,8 @@ config SND_SOC_ALL_CODECS
 	select SND_SOC_MC13783 if MFD_MC13XXX
 	select SND_SOC_ML26124 if I2C
 	select SND_SOC_HDMI_CODEC
+	select SND_SOC_PCM1681 if I2C
+	select SND_SOC_PCM1792A if SPI_MASTER
 	select SND_SOC_PCM3008
 	select SND_SOC_RT5631 if I2C
 	select SND_SOC_RT5640 if I2C
@@ -122,6 +126,7 @@ config SND_SOC_ALL_CODECS
 	select SND_SOC_WM8994 if MFD_WM8994
 	select SND_SOC_WM8995 if SND_SOC_I2C_AND_SPI
 	select SND_SOC_WM8996 if I2C
+	select SND_SOC_WM8997 if MFD_WM8997
 	select SND_SOC_WM9081 if I2C
 	select SND_SOC_WM9090 if I2C
 	select SND_SOC_WM9705 if SND_SOC_AC97_BUS
@@ -145,8 +150,10 @@ config SND_SOC_ARIZONA
 	tristate
 	default y if SND_SOC_WM5102=y
 	default y if SND_SOC_WM5110=y
+	default y if SND_SOC_WM8997=y
 	default m if SND_SOC_WM5102=m
 	default m if SND_SOC_WM5110=m
+	default m if SND_SOC_WM8997=m
 
 config SND_SOC_WM_HUBS
 	tristate
@@ -198,6 +205,9 @@ config SND_SOC_AK4104
 config SND_SOC_AK4535
 	tristate
 
+config SND_SOC_AK4554
+	tristate
+
 config SND_SOC_AK4641
 	tristate
 
@@ -292,6 +302,12 @@ config SND_SOC_MAX9850
 config SND_SOC_HDMI_CODEC
        tristate
 
+config SND_SOC_PCM1681
+       tristate
+
+config SND_SOC_PCM1792A
+       tristate
+
 config SND_SOC_PCM3008
        tristate
 
@@ -500,6 +516,9 @@ config SND_SOC_WM8995
 config SND_SOC_WM8996
 	tristate
 
+config SND_SOC_WM8997
+	tristate
+
 config SND_SOC_WM9081
 	tristate
 

sound/soc/codecs/Makefile

@@ -11,6 +11,7 @@ snd-soc-adav80x-objs := adav80x.o
 snd-soc-ads117x-objs := ads117x.o
 snd-soc-ak4104-objs := ak4104.o
 snd-soc-ak4535-objs := ak4535.o
+snd-soc-ak4554-objs := ak4554.o
 snd-soc-ak4641-objs := ak4641.o
 snd-soc-ak4642-objs := ak4642.o
 snd-soc-ak4671-objs := ak4671.o
@@ -42,6 +43,8 @@ snd-soc-max9850-objs := max9850.o
 snd-soc-mc13783-objs := mc13783.o
 snd-soc-ml26124-objs := ml26124.o
 snd-soc-hdmi-codec-objs := hdmi.o
+snd-soc-pcm1681-objs := pcm1681.o
+snd-soc-pcm1792a-codec-objs := pcm1792a.o
 snd-soc-pcm3008-objs := pcm3008.o
 snd-soc-rt5631-objs := rt5631.o
 snd-soc-rt5640-objs := rt5640.o
@@ -114,6 +117,7 @@ snd-soc-wm8991-objs := wm8991.o
 snd-soc-wm8993-objs := wm8993.o
 snd-soc-wm8994-objs := wm8994.o wm8958-dsp2.o
 snd-soc-wm8995-objs := wm8995.o
+snd-soc-wm8997-objs := wm8997.o
 snd-soc-wm9081-objs := wm9081.o
 snd-soc-wm9090-objs := wm9090.o
 snd-soc-wm9705-objs := wm9705.o
@@ -138,6 +142,7 @@ obj-$(CONFIG_SND_SOC_ADAV80X)  += snd-soc-adav80x.o
 obj-$(CONFIG_SND_SOC_ADS117X)	+= snd-soc-ads117x.o
 obj-$(CONFIG_SND_SOC_AK4104)	+= snd-soc-ak4104.o
 obj-$(CONFIG_SND_SOC_AK4535)	+= snd-soc-ak4535.o
+obj-$(CONFIG_SND_SOC_AK4554)	+= snd-soc-ak4554.o
 obj-$(CONFIG_SND_SOC_AK4641)	+= snd-soc-ak4641.o
 obj-$(CONFIG_SND_SOC_AK4642)	+= snd-soc-ak4642.o
 obj-$(CONFIG_SND_SOC_AK4671)	+= snd-soc-ak4671.o
@@ -171,6 +176,8 @@ 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_HDMI_CODEC) += snd-soc-hdmi-codec.o
+obj-$(CONFIG_SND_SOC_PCM1681)	+= snd-soc-pcm1681.o
+obj-$(CONFIG_SND_SOC_PCM1792A)	+= snd-soc-pcm1792a-codec.o
 obj-$(CONFIG_SND_SOC_PCM3008)	+= snd-soc-pcm3008.o
 obj-$(CONFIG_SND_SOC_RT5631)	+= snd-soc-rt5631.o
 obj-$(CONFIG_SND_SOC_RT5640)	+= snd-soc-rt5640.o
@@ -239,6 +246,7 @@ obj-$(CONFIG_SND_SOC_WM8991)	+= snd-soc-wm8991.o
 obj-$(CONFIG_SND_SOC_WM8993)	+= snd-soc-wm8993.o
 obj-$(CONFIG_SND_SOC_WM8994)	+= snd-soc-wm8994.o
 obj-$(CONFIG_SND_SOC_WM8995)	+= snd-soc-wm8995.o
+obj-$(CONFIG_SND_SOC_WM8997)	+= snd-soc-wm8997.o
 obj-$(CONFIG_SND_SOC_WM9081)	+= snd-soc-wm9081.o
 obj-$(CONFIG_SND_SOC_WM9090)	+= snd-soc-wm9090.o
 obj-$(CONFIG_SND_SOC_WM9705)	+= snd-soc-wm9705.o

sound/soc/codecs/ac97.c

@@ -23,6 +23,16 @@
 #include <sound/initval.h>
 #include <sound/soc.h>
 
+static const struct snd_soc_dapm_widget ac97_widgets[] = {
+	SND_SOC_DAPM_INPUT("RX"),
+	SND_SOC_DAPM_OUTPUT("TX"),
+};
+
+static const struct snd_soc_dapm_route ac97_routes[] = {
+	{ "AC97 Capture", NULL, "RX" },
+	{ "TX", NULL, "AC97 Playback" },
+};
+
 static int ac97_prepare(struct snd_pcm_substream *substream,
 			struct snd_soc_dai *dai)
 {
@@ -117,6 +127,11 @@ static struct snd_soc_codec_driver soc_codec_dev_ac97 = {
 	.probe = 	ac97_soc_probe,
 	.suspend =	ac97_soc_suspend,
 	.resume =	ac97_soc_resume,
+
+	.dapm_widgets = ac97_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(ac97_widgets),
+	.dapm_routes = ac97_routes,
+	.num_dapm_routes = ARRAY_SIZE(ac97_routes),
 };
 
 static int ac97_probe(struct platform_device *pdev)

sound/soc/codecs/ad1980.c

@@ -96,6 +96,44 @@ SOC_ENUM("Capture Source", ad1980_cap_src),
 SOC_SINGLE("Mic Boost Switch", AC97_MIC, 6, 1, 0),
 };
 
+static const struct snd_soc_dapm_widget ad1980_dapm_widgets[] = {
+SND_SOC_DAPM_INPUT("MIC1"),
+SND_SOC_DAPM_INPUT("MIC2"),
+SND_SOC_DAPM_INPUT("CD_L"),
+SND_SOC_DAPM_INPUT("CD_R"),
+SND_SOC_DAPM_INPUT("AUX_L"),
+SND_SOC_DAPM_INPUT("AUX_R"),
+SND_SOC_DAPM_INPUT("LINE_IN_L"),
+SND_SOC_DAPM_INPUT("LINE_IN_R"),
+
+SND_SOC_DAPM_OUTPUT("LFE_OUT"),
+SND_SOC_DAPM_OUTPUT("CENTER_OUT"),
+SND_SOC_DAPM_OUTPUT("LINE_OUT_L"),
+SND_SOC_DAPM_OUTPUT("LINE_OUT_R"),
+SND_SOC_DAPM_OUTPUT("MONO_OUT"),
+SND_SOC_DAPM_OUTPUT("HP_OUT_L"),
+SND_SOC_DAPM_OUTPUT("HP_OUT_R"),
+};
+
+static const struct snd_soc_dapm_route ad1980_dapm_routes[] = {
+	{ "Capture", NULL, "MIC1" },
+	{ "Capture", NULL, "MIC2" },
+	{ "Capture", NULL, "CD_L" },
+	{ "Capture", NULL, "CD_R" },
+	{ "Capture", NULL, "AUX_L" },
+	{ "Capture", NULL, "AUX_R" },
+	{ "Capture", NULL, "LINE_IN_L" },
+	{ "Capture", NULL, "LINE_IN_R" },
+
+	{ "LFE_OUT", NULL, "Playback" },
+	{ "CENTER_OUT", NULL, "Playback" },
+	{ "LINE_OUT_L", NULL, "Playback" },
+	{ "LINE_OUT_R", NULL, "Playback" },
+	{ "MONO_OUT", NULL, "Playback" },
+	{ "HP_OUT_L", NULL, "Playback" },
+	{ "HP_OUT_R", NULL, "Playback" },
+};
+
 static unsigned int ac97_read(struct snd_soc_codec *codec,
 	unsigned int reg)
 {
@@ -253,6 +291,11 @@ static struct snd_soc_codec_driver soc_codec_dev_ad1980 = {
 	.reg_cache_step = 2,
 	.write = ac97_write,
 	.read = ac97_read,
+
+	.dapm_widgets = ad1980_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(ad1980_dapm_widgets),
+	.dapm_routes = ad1980_dapm_routes,
+	.num_dapm_routes = ARRAY_SIZE(ad1980_dapm_routes),
 };
 
 static int ad1980_probe(struct platform_device *pdev)

sound/soc/codecs/ad73311.c

@@ -23,6 +23,21 @@
 
 #include "ad73311.h"
 
+static const struct snd_soc_dapm_widget ad73311_dapm_widgets[] = {
+SND_SOC_DAPM_INPUT("VINP"),
+SND_SOC_DAPM_INPUT("VINN"),
+SND_SOC_DAPM_OUTPUT("VOUTN"),
+SND_SOC_DAPM_OUTPUT("VOUTP"),
+};
+
+static const struct snd_soc_dapm_route ad73311_dapm_routes[] = {
+	{ "Capture", NULL, "VINP" },
+	{ "Capture", NULL, "VINN" },
+
+	{ "VOUTN", NULL, "Playback" },
+	{ "VOUTP", NULL, "Playback" },
+};
+
 static struct snd_soc_dai_driver ad73311_dai = {
 	.name = "ad73311-hifi",
 	.playback = {
@@ -39,7 +54,12 @@ static struct snd_soc_dai_driver ad73311_dai = {
 		.formats = SNDRV_PCM_FMTBIT_S16_LE, },
 };
 
-static struct snd_soc_codec_driver soc_codec_dev_ad73311;
+static struct snd_soc_codec_driver soc_codec_dev_ad73311 = {
+	.dapm_widgets = ad73311_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(ad73311_dapm_widgets),
+	.dapm_routes = ad73311_dapm_routes,
+	.num_dapm_routes = ARRAY_SIZE(ad73311_dapm_routes),
+};
 
 static int ad73311_probe(struct platform_device *pdev)
 {

sound/soc/codecs/adau1701.c

@@ -91,7 +91,7 @@
 #define ADAU1701_OSCIPOW_OPD		0x04
 #define ADAU1701_DACSET_DACINIT		1
 
-#define ADAU1707_CLKDIV_UNSET		(-1UL)
+#define ADAU1707_CLKDIV_UNSET		(-1U)
 
 #define ADAU1701_FIRMWARE "adau1701.bin"
 
@@ -247,21 +247,21 @@ static int adau1701_reset(struct snd_soc_codec *codec, unsigned int clkdiv)
 	    gpio_is_valid(adau1701->gpio_pll_mode[1])) {
 		switch (clkdiv) {
 		case 64:
-			gpio_set_value(adau1701->gpio_pll_mode[0], 0);
-			gpio_set_value(adau1701->gpio_pll_mode[1], 0);
+			gpio_set_value_cansleep(adau1701->gpio_pll_mode[0], 0);
+			gpio_set_value_cansleep(adau1701->gpio_pll_mode[1], 0);
 			break;
 		case 256:
-			gpio_set_value(adau1701->gpio_pll_mode[0], 0);
-			gpio_set_value(adau1701->gpio_pll_mode[1], 1);
+			gpio_set_value_cansleep(adau1701->gpio_pll_mode[0], 0);
+			gpio_set_value_cansleep(adau1701->gpio_pll_mode[1], 1);
 			break;
 		case 384:
-			gpio_set_value(adau1701->gpio_pll_mode[0], 1);
-			gpio_set_value(adau1701->gpio_pll_mode[1], 0);
+			gpio_set_value_cansleep(adau1701->gpio_pll_mode[0], 1);
+			gpio_set_value_cansleep(adau1701->gpio_pll_mode[1], 0);
 			break;
 		case 0:	/* fallback */
 		case 512:
-			gpio_set_value(adau1701->gpio_pll_mode[0], 1);
-			gpio_set_value(adau1701->gpio_pll_mode[1], 1);
+			gpio_set_value_cansleep(adau1701->gpio_pll_mode[0], 1);
+			gpio_set_value_cansleep(adau1701->gpio_pll_mode[1], 1);
 			break;
 		}
 	}
@@ -269,10 +269,10 @@ static int adau1701_reset(struct snd_soc_codec *codec, unsigned int clkdiv)
 	adau1701->pll_clkdiv = clkdiv;
 
 	if (gpio_is_valid(adau1701->gpio_nreset)) {
-		gpio_set_value(adau1701->gpio_nreset, 0);
+		gpio_set_value_cansleep(adau1701->gpio_nreset, 0);
 		/* minimum reset time is 20ns */
 		udelay(1);
-		gpio_set_value(adau1701->gpio_nreset, 1);
+		gpio_set_value_cansleep(adau1701->gpio_nreset, 1);
 		/* power-up time may be as long as 85ms */
 		mdelay(85);
 	}
@@ -734,7 +734,10 @@ static int adau1701_i2c_remove(struct i2c_client *client)
 }
 
 static const struct i2c_device_id adau1701_i2c_id[] = {
+	{ "adau1401", 0 },
+	{ "adau1401a", 0 },
 	{ "adau1701", 0 },
+	{ "adau1702", 0 },
 	{ }
 };
 MODULE_DEVICE_TABLE(i2c, adau1701_i2c_id);

sound/soc/codecs/adav80x.c

@@ -868,6 +868,12 @@ static int adav80x_bus_remove(struct device *dev)
 }
 
 #if defined(CONFIG_SPI_MASTER)
+static const struct spi_device_id adav80x_spi_id[] = {
+	{ "adav801", 0 },
+	{ }
+};
+MODULE_DEVICE_TABLE(spi, adav80x_spi_id);
+
 static int adav80x_spi_probe(struct spi_device *spi)
 {
 	return adav80x_bus_probe(&spi->dev, SND_SOC_SPI);
@@ -885,15 +891,16 @@ static struct spi_driver adav80x_spi_driver = {
 	},
 	.probe		= adav80x_spi_probe,
 	.remove		= adav80x_spi_remove,
+	.id_table	= adav80x_spi_id,
 };
 #endif
 
 #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
-static const struct i2c_device_id adav80x_id[] = {
+static const struct i2c_device_id adav80x_i2c_id[] = {
 	{ "adav803", 0 },
 	{ }
 };
-MODULE_DEVICE_TABLE(i2c, adav80x_id);
+MODULE_DEVICE_TABLE(i2c, adav80x_i2c_id);
 
 static int adav80x_i2c_probe(struct i2c_client *client,
 			     const struct i2c_device_id *id)
@@ -913,7 +920,7 @@ static struct i2c_driver adav80x_i2c_driver = {
 	},
 	.probe = adav80x_i2c_probe,
 	.remove = adav80x_i2c_remove,
-	.id_table = adav80x_id,
+	.id_table = adav80x_i2c_id,
 };
 #endif
 

sound/soc/codecs/ads117x.c

@@ -23,6 +23,28 @@
 #define ADS117X_RATES (SNDRV_PCM_RATE_8000_48000)
 #define ADS117X_FORMATS (SNDRV_PCM_FMTBIT_S16_LE)
 
+static const struct snd_soc_dapm_widget ads117x_dapm_widgets[] = {
+SND_SOC_DAPM_INPUT("Input1"),
+SND_SOC_DAPM_INPUT("Input2"),
+SND_SOC_DAPM_INPUT("Input3"),
+SND_SOC_DAPM_INPUT("Input4"),
+SND_SOC_DAPM_INPUT("Input5"),
+SND_SOC_DAPM_INPUT("Input6"),
+SND_SOC_DAPM_INPUT("Input7"),
+SND_SOC_DAPM_INPUT("Input8"),
+};
+
+static const struct snd_soc_dapm_route ads117x_dapm_routes[] = {
+	{ "Capture", NULL, "Input1" },
+	{ "Capture", NULL, "Input2" },
+	{ "Capture", NULL, "Input3" },
+	{ "Capture", NULL, "Input4" },
+	{ "Capture", NULL, "Input5" },
+	{ "Capture", NULL, "Input6" },
+	{ "Capture", NULL, "Input7" },
+	{ "Capture", NULL, "Input8" },
+};
+
 static struct snd_soc_dai_driver ads117x_dai = {
 /* ADC */
 	.name = "ads117x-hifi",
@@ -34,7 +56,12 @@ static struct snd_soc_dai_driver ads117x_dai = {
 		.formats = ADS117X_FORMATS,},
 };
 
-static struct snd_soc_codec_driver soc_codec_dev_ads117x;
+static struct snd_soc_codec_driver soc_codec_dev_ads117x = {
+	.dapm_widgets = ads117x_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(ads117x_dapm_widgets),
+	.dapm_routes = ads117x_dapm_routes,
+	.num_dapm_routes = ARRAY_SIZE(ads117x_dapm_routes),
+};
 
 static int ads117x_probe(struct platform_device *pdev)
 {

sound/soc/codecs/ak4104.c

@@ -51,6 +51,17 @@ struct ak4104_private {
 	struct regmap *regmap;
 };
 
+static const struct snd_soc_dapm_widget ak4104_dapm_widgets[] = {
+SND_SOC_DAPM_PGA("TXE", AK4104_REG_TX, AK4104_TX_TXE, 0, NULL, 0),
+
+SND_SOC_DAPM_OUTPUT("TX"),
+};
+
+static const struct snd_soc_dapm_route ak4104_dapm_routes[] = {
+	{ "TXE", NULL, "Playback" },
+	{ "TX", NULL, "TXE" },
+};
+
 static int ak4104_set_dai_fmt(struct snd_soc_dai *codec_dai,
 			      unsigned int format)
 {
@@ -138,29 +149,11 @@ static int ak4104_hw_params(struct snd_pcm_substream *substream,
 	if (ret < 0)
 		return ret;
 
-	/* enable transmitter */
-	ret = regmap_update_bits(ak4104->regmap, AK4104_REG_TX,
-				 AK4104_TX_TXE, AK4104_TX_TXE);
-	if (ret < 0)
-		return ret;
-
 	return 0;
 }
 
-static int ak4104_hw_free(struct snd_pcm_substream *substream,
-			  struct snd_soc_dai *dai)
-{
-	struct snd_soc_codec *codec = dai->codec;
-	struct ak4104_private *ak4104 = snd_soc_codec_get_drvdata(codec);
-
-	/* disable transmitter */
-	return regmap_update_bits(ak4104->regmap, AK4104_REG_TX,
-				  AK4104_TX_TXE, 0);
-}
-
 static const struct snd_soc_dai_ops ak4101_dai_ops = {
 	.hw_params = ak4104_hw_params,
-	.hw_free = ak4104_hw_free,
 	.set_fmt = ak4104_set_dai_fmt,
 };
 
@@ -214,6 +207,11 @@ static int ak4104_remove(struct snd_soc_codec *codec)
 static struct snd_soc_codec_driver soc_codec_device_ak4104 = {
 	.probe =	ak4104_probe,
 	.remove =	ak4104_remove,
+
+	.dapm_widgets = ak4104_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(ak4104_dapm_widgets),
+	.dapm_routes = ak4104_dapm_routes,
+	.num_dapm_routes = ARRAY_SIZE(ak4104_dapm_routes),
 };
 
 static const struct regmap_config ak4104_regmap = {